diff options
Diffstat (limited to 'sound')
395 files changed, 22818 insertions, 5196 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 6833db9002ec..1140e9988fc5 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -96,6 +96,8 @@ source "sound/x86/Kconfig" source "sound/synth/Kconfig" +source "sound/xen/Kconfig" + endif # SND endif # !UML diff --git a/sound/Makefile b/sound/Makefile index 99d8c31262c8..797ecdcd35e2 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_DMASOUND) += oss/dmasound/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ + firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ xen/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..5fbd47a9177e 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -17,14 +17,9 @@ config SND_ARMAACI select SND_PCM select SND_AC97_CODEC -config SND_PXA2XX_PCM - tristate - select SND_PCM - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_PXA2XX_PCM select SND_AC97_CODEC select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 diff --git a/sound/arm/Makefile b/sound/arm/Makefile index e10d5b169565..34c769489877 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -6,9 +6,6 @@ obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o -obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o -snd-pxa2xx-pcm-objs := pxa2xx-pcm.o - obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 5950a9e218d9..8eafd3d3dff6 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/gpio.h> +#include <linux/of_gpio.h> #include <sound/pxa2xx-lib.h> @@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) dev_err(&dev->dev, "Invalid reset GPIO %d\n", pdata->reset_gpio); } + } else if (!pdata && dev->dev.of_node) { + pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node, + "reset-gpios", 0); + if (pdata->reset_gpio == -ENOENT) + pdata->reset_gpio = -1; + else if (pdata->reset_gpio < 0) + return pdata->reset_gpio; + reset_gpio = pdata->reset_gpio; } else { if (cpu_is_pxa27x()) reset_gpio = 113; diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..1f72672262d0 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,7 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> -#include <linux/dma/pxa-dma.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -27,8 +27,6 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -#include "pxa2xx-pcm.h" - static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97) { if (!pxa2xx_ac97_try_cold_reset()) @@ -63,61 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, -}; - static struct snd_pcm *pxa2xx_ac97_pcm; static struct snd_ac97 *pxa2xx_ac97_ac97; -static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; pxa2xx_audio_ops_t *platform_ops; - int r; + int ret, i; + + ret = pxa2xx_pcm_open(substream); + if (ret) + return ret; runtime->hw.channels_min = 2; runtime->hw.channels_max = 2; - r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - AC97_RATES_FRONT_DAC : AC97_RATES_ADC; - runtime->hw.rates = pxa2xx_ac97_ac97->rates[r]; + i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_RATES_FRONT_DAC : AC97_RATES_ADC; + runtime->hw.rates = pxa2xx_ac97_ac97->rates[i]; snd_pcm_limit_hw_rates(runtime); - platform_ops = substream->pcm->card->dev->platform_data; - if (platform_ops && platform_ops->startup) - return platform_ops->startup(substream, platform_ops->priv); - else - return 0; + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->startup) { + ret = platform_ops->startup(substream, platform_ops->priv); + if (ret < 0) + pxa2xx_pcm_close(substream); + } + + return ret; } -static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream) { pxa2xx_audio_ops_t *platform_ops; - platform_ops = substream->pcm->card->dev->platform_data; + platform_ops = substream->pcm->card->dev->platform_data; if (platform_ops && platform_ops->shutdown) platform_ops->shutdown(substream, platform_ops->priv); + + return 0; } static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) @@ -125,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; + int ret; + + ret = pxa2xx_pcm_prepare(substream); + if (ret < 0) + return ret; + return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); } -static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { - .playback_params = &pxa2xx_ac97_pcm_out, - .capture_params = &pxa2xx_ac97_pcm_in, - .startup = pxa2xx_ac97_pcm_startup, - .shutdown = pxa2xx_ac97_pcm_shutdown, - .prepare = pxa2xx_ac97_pcm_prepare, -}; - #ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) @@ -193,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif +static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = { + .open = pxa2xx_ac97_pcm_open, + .close = pxa2xx_ac97_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_ac97_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; + + +static int pxa2xx_ac97_pcm_new(struct snd_card *card) +{ + struct snd_pcm *pcm; + int stream, ret; + + ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm); + if (ret) + goto out; + + pcm->private_free = pxa2xx_pcm_free_dma_buffers; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_CAPTURE; + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + pxa2xx_ac97_pcm = pcm; + ret = 0; + + out: + return ret; +} + static int pxa2xx_ac97_probe(struct platform_device *dev) { struct snd_card *card; @@ -214,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); - ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); + ret = pxa2xx_ac97_pcm_new(card); if (ret) goto err; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..7931789d4a9f 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -16,8 +16,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "pxa2xx-pcm.h" - static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -25,8 +23,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, @@ -35,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .fifo_size = 32, }; -int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -64,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); +EXPORT_SYMBOL(pxa2xx_pcm_hw_params); -int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) +int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); +EXPORT_SYMBOL(pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -86,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(pxa2xx_pcm_pointer); -int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) +int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_prepare); +EXPORT_SYMBOL(pxa2xx_pcm_prepare); -int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) +int pxa2xx_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -125,17 +123,17 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } -EXPORT_SYMBOL(__pxa2xx_pcm_open); +EXPORT_SYMBOL(pxa2xx_pcm_open); -int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) +int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { return snd_dmaengine_pcm_close_release_chan(substream); } -EXPORT_SYMBOL(__pxa2xx_pcm_close); +EXPORT_SYMBOL(pxa2xx_pcm_close); int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -181,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers); +int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} +EXPORT_SYMBOL(pxa2xx_soc_pcm_new); + +const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; +EXPORT_SYMBOL(pxa2xx_pcm_ops); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx sound library"); MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c deleted file mode 100644 index 1c6f4b436de3..000000000000 --- a/sound/arm/pxa2xx-pcm.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/dma-mapping.h> -#include <linux/dmaengine.h> - -#include <mach/dma.h> - -#include <sound/core.h> -#include <sound/pxa2xx-lib.h> -#include <sound/dmaengine_pcm.h> - -#include "pxa2xx-pcm.h" - -static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - __pxa2xx_pcm_prepare(substream); - - return client->prepare(substream); -} - -static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; - int ret; - - ret = __pxa2xx_pcm_open(substream); - if (ret) - goto out; - - rtd = runtime->private_data; - - rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - client->playback_params : client->capture_params; - - ret = client->startup(substream); - if (!ret) - goto err2; - - return 0; - - err2: - __pxa2xx_pcm_close(substream); - out: - return ret; -} - -static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - client->shutdown(substream); - - return __pxa2xx_pcm_close(substream); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, - struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int play = client->playback_params ? 1 : 0; - int capt = client->capture_params ? 1 : 0; - int ret; - - ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm); - if (ret) - goto out; - - pcm->private_data = client; - pcm->private_free = pxa2xx_pcm_free_dma_buffers; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - goto out; - - if (play) { - int stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - if (capt) { - int stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - - if (rpcm) - *rpcm = pcm; - ret = 0; - - out: - return ret; -} - -EXPORT_SYMBOL(pxa2xx_pcm_new); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h deleted file mode 100644 index 8fa2b7c9e6b8..000000000000 --- a/sound/arm/pxa2xx-pcm.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -struct pxa2xx_runtime_data { - int dma_ch; - struct snd_dmaengine_dai_dma_data *params; -}; - -struct pxa2xx_pcm_client { - struct snd_dmaengine_dai_dma_data *playback_params; - struct snd_dmaengine_dai_dma_data *capture_params; - int (*startup)(struct snd_pcm_substream *); - void (*shutdown)(struct snd_pcm_substream *); - int (*prepare)(struct snd_pcm_substream *); -}; - -extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **); - diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6e937a8146a1..63b3ef9c83f5 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -48,7 +48,7 @@ config SND_MIXER_OSS depends on SND_OSSEMUL help To enable OSS mixer API emulation (/dev/mixer*), say Y here - and read <file:Documentation/sound/alsa/OSS-Emulation.txt>. + and read <file:Documentation/sound/designs/oss-emulation.rst>. Many programs still use the OSS API, so say Y. @@ -61,7 +61,7 @@ config SND_PCM_OSS select SND_PCM help To enable OSS digital audio (PCM) emulation (/dev/dsp*), say Y - here and read <file:Documentation/sound/alsa/OSS-Emulation.txt>. + here and read <file:Documentation/sound/designs/oss-emulation.rst>. Many programs still use the OSS API, so say Y. diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4563432badba..4b01a37c836e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1001,7 +1001,7 @@ static int snd_compress_proc_init(struct snd_compr *compr) compr->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; diff --git a/sound/core/device.c b/sound/core/device.c index cb0e46f66cc9..535102d564e3 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -240,6 +240,15 @@ void snd_device_free_all(struct snd_card *card) if (snd_BUG_ON(!card)) return; + list_for_each_entry_safe_reverse(dev, next, &card->devices, list) { + /* exception: free ctl and lowlevel stuff later */ + if (dev->type == SNDRV_DEV_CONTROL || + dev->type == SNDRV_DEV_LOWLEVEL) + continue; + __snd_device_free(dev); + } + + /* free all */ list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } diff --git a/sound/core/info.c b/sound/core/info.c index 4b36767af9e1..fe502bc5e6d2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -454,7 +454,7 @@ static struct snd_info_entry *create_subdir(struct module *mod, entry = snd_info_create_module_entry(mod, name, NULL); if (!entry) return NULL; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return NULL; @@ -470,7 +470,7 @@ int __init snd_info_init(void) snd_proc_root = snd_info_create_entry("asound", NULL); if (!snd_proc_root) return -ENOMEM; - snd_proc_root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + snd_proc_root->mode = S_IFDIR | 0555; snd_proc_root->p = proc_mkdir("asound", NULL); if (!snd_proc_root->p) goto error; @@ -716,7 +716,7 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent) kfree(entry); return NULL; } - entry->mode = S_IFREG | S_IRUGO; + entry->mode = S_IFREG | 0444; entry->content = SNDRV_INFO_CONTENT_TEXT; mutex_init(&entry->access); INIT_LIST_HEAD(&entry->children); diff --git a/sound/core/init.c b/sound/core/init.c index 79b4df1c1dc7..4849c611c0fe 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -703,7 +703,7 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return count; } -static DEVICE_ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); +static DEVICE_ATTR(id, 0644, card_id_show_attr, card_id_store_attr); static ssize_t card_number_show_attr(struct device *dev, @@ -713,7 +713,7 @@ card_number_show_attr(struct device *dev, return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } -static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); +static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL); static struct attribute *card_dev_attrs[] = { &dev_attr_id.attr, diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 379bf486ccc7..64d904bee8bb 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1247,7 +1247,7 @@ static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer) if (! entry) return; entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_mixer_oss_proc_read; entry->c.text.write = snd_mixer_oss_proc_write; entry->private_data = mixer; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1980f68246cb..905a53c1cde5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -3045,7 +3045,7 @@ static void snd_pcm_oss_proc_init(struct snd_pcm *pcm) continue; if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_pcm_oss_proc_read; entry->c.text.write = snd_pcm_oss_proc_write; entry->private_data = pstr; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 66ac89aad681..c352bfb973cc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -530,7 +530,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) pcm->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -552,7 +552,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) if (entry) { entry->c.text.read = snd_pcm_xrun_debug_read; entry->c.text.write = snd_pcm_xrun_debug_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = pstr; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -590,7 +590,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->pstr->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -647,7 +647,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->private_data = substream; entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; - entry->mode = S_IFREG | S_IWUSR; + entry->mode = S_IFREG | 0200; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -1087,7 +1087,7 @@ static ssize_t show_pcm_class(struct device *dev, return snprintf(buf, PAGE_SIZE, "%s\n", str); } -static DEVICE_ATTR(pcm_class, S_IRUGO, show_pcm_class, NULL); +static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 6491afbb5fd5..946ab080ac00 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -45,10 +45,7 @@ static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream, if (get_user(frames, src)) return -EFAULT; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = snd_pcm_playback_rewind(substream, frames); - else - err = snd_pcm_capture_rewind(substream, frames); + err = snd_pcm_rewind(substream, frames); if (put_user(err, src)) return -EFAULT; return err < 0 ? err : 0; @@ -62,10 +59,7 @@ static int snd_pcm_ioctl_forward_compat(struct snd_pcm_substream *substream, if (get_user(frames, src)) return -EFAULT; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = snd_pcm_playback_forward(substream, frames); - else - err = snd_pcm_capture_forward(substream, frames); + err = snd_pcm_forward(substream, frames); if (put_user(err, src)) return -EFAULT; return err < 0 ? err : 0; @@ -432,7 +426,7 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, get_user(frames, &data32->frames)) return -EFAULT; bufptr = compat_ptr(buf); - bufs = kmalloc(sizeof(void __user *) * ch, GFP_KERNEL); + bufs = kmalloc_array(ch, sizeof(void __user *), GFP_KERNEL); if (bufs == NULL) return -ENOMEM; for (i = 0; i < ch; i++) { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f4a19509cccf..44b5ae833082 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -191,10 +191,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, { snd_pcm_uframes_t avail; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); + avail = snd_pcm_avail(substream); if (avail > runtime->avail_max) runtime->avail_max = avail; if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { @@ -1856,10 +1853,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, * This check must happen after been added to the waitqueue * and having current state be INTERRUPTIBLE. */ - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); + avail = snd_pcm_avail(substream); if (avail >= runtime->twake) break; snd_pcm_stream_unlock_irq(substream); @@ -2175,10 +2169,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, runtime->twake = runtime->control->avail_min ? : 1; if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); + avail = snd_pcm_avail(substream); while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t cont; diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 16f254732b2a..7a499d02df6c 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -36,6 +36,24 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr); +static inline snd_pcm_uframes_t +snd_pcm_avail(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return snd_pcm_playback_avail(substream->runtime); + else + return snd_pcm_capture_avail(substream->runtime); +} + +static inline snd_pcm_uframes_t +snd_pcm_hw_avail(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return snd_pcm_playback_hw_avail(substream->runtime); + else + return snd_pcm_capture_hw_avail(substream->runtime); +} + #ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index ae33e456708c..4b5356a10315 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -201,7 +201,7 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_proc_read; entry->c.text.write = snd_pcm_lib_preallocate_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = substream; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0e875d5a9e86..cecc79772c94 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -99,6 +99,57 @@ static inline void down_write_nonblock(struct rw_semaphore *lock) cond_resched(); } +#define PCM_LOCK_DEFAULT 0 +#define PCM_LOCK_IRQ 1 +#define PCM_LOCK_IRQSAVE 2 + +static unsigned long __snd_pcm_stream_lock_mode(struct snd_pcm_substream *substream, + unsigned int mode) +{ + unsigned long flags = 0; + if (substream->pcm->nonatomic) { + down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); + mutex_lock(&substream->self_group.mutex); + } else { + switch (mode) { + case PCM_LOCK_DEFAULT: + read_lock(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQ: + read_lock_irq(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQSAVE: + read_lock_irqsave(&snd_pcm_link_rwlock, flags); + break; + } + spin_lock(&substream->self_group.lock); + } + return flags; +} + +static void __snd_pcm_stream_unlock_mode(struct snd_pcm_substream *substream, + unsigned int mode, unsigned long flags) +{ + if (substream->pcm->nonatomic) { + mutex_unlock(&substream->self_group.mutex); + up_read(&snd_pcm_link_rwsem); + } else { + spin_unlock(&substream->self_group.lock); + + switch (mode) { + case PCM_LOCK_DEFAULT: + read_unlock(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQ: + read_unlock_irq(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQSAVE: + read_unlock_irqrestore(&snd_pcm_link_rwlock, flags); + break; + } + } +} + /** * snd_pcm_stream_lock - Lock the PCM stream * @substream: PCM substream @@ -109,13 +160,7 @@ static inline void down_write_nonblock(struct rw_semaphore *lock) */ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { - if (substream->pcm->nonatomic) { - down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); - mutex_lock(&substream->self_group.mutex); - } else { - read_lock(&snd_pcm_link_rwlock); - spin_lock(&substream->self_group.lock); - } + __snd_pcm_stream_lock_mode(substream, PCM_LOCK_DEFAULT); } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); @@ -127,13 +172,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); */ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream) { - if (substream->pcm->nonatomic) { - mutex_unlock(&substream->self_group.mutex); - up_read(&snd_pcm_link_rwsem); - } else { - spin_unlock(&substream->self_group.lock); - read_unlock(&snd_pcm_link_rwlock); - } + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_DEFAULT, 0); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); @@ -147,9 +186,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); */ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) { - if (!substream->pcm->nonatomic) - local_irq_disable(); - snd_pcm_stream_lock(substream); + __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQ); } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); @@ -161,19 +198,13 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); */ void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream) { - snd_pcm_stream_unlock(substream); - if (!substream->pcm->nonatomic) - local_irq_enable(); + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQ, 0); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) { - unsigned long flags = 0; - if (!substream->pcm->nonatomic) - local_irq_save(flags); - snd_pcm_stream_lock(substream); - return flags; + return __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQSAVE); } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); @@ -187,9 +218,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags) { - snd_pcm_stream_unlock(substream); - if (!substream->pcm->nonatomic) - local_irq_restore(flags); + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQSAVE, flags); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); @@ -857,6 +886,18 @@ static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, return err; } +static inline snd_pcm_uframes_t +snd_pcm_calc_delay(struct snd_pcm_substream *substream) +{ + snd_pcm_uframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = snd_pcm_playback_hw_avail(substream->runtime); + else + delay = snd_pcm_capture_avail(substream->runtime); + return delay + substream->runtime->delay; +} + int snd_pcm_status(struct snd_pcm_substream *substream, struct snd_pcm_status *status) { @@ -908,21 +949,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream, _tstamp_end: status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - status->avail = snd_pcm_playback_avail(runtime); - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || - runtime->status->state == SNDRV_PCM_STATE_DRAINING) { - status->delay = runtime->buffer_size - status->avail; - status->delay += runtime->delay; - } else - status->delay = 0; - } else { - status->avail = snd_pcm_capture_avail(runtime); - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - status->delay = status->avail + runtime->delay; - else - status->delay = 0; - } + status->avail = snd_pcm_avail(substream); + status->delay = snd_pcm_running(substream) ? + snd_pcm_calc_delay(substream) : 0; status->avail_max = runtime->avail_max; status->overrange = runtime->overrange; runtime->avail_max = 0; @@ -2610,10 +2639,9 @@ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream, return ret < 0 ? 0 : frames; } -static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream, - snd_pcm_uframes_t frames) +static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream, + snd_pcm_uframes_t frames) { - struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t ret; if (frames == 0) @@ -2623,33 +2651,14 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst ret = do_pcm_hwsync(substream); if (!ret) ret = rewind_appl_ptr(substream, frames, - snd_pcm_playback_hw_avail(runtime)); + snd_pcm_hw_avail(substream)); snd_pcm_stream_unlock_irq(substream); return ret; } -static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substream, - snd_pcm_uframes_t frames) +static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream, + snd_pcm_uframes_t frames) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_sframes_t ret; - - if (frames == 0) - return 0; - - snd_pcm_stream_lock_irq(substream); - ret = do_pcm_hwsync(substream); - if (!ret) - ret = rewind_appl_ptr(substream, frames, - snd_pcm_capture_hw_avail(runtime)); - snd_pcm_stream_unlock_irq(substream); - return ret; -} - -static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *substream, - snd_pcm_uframes_t frames) -{ - struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t ret; if (frames == 0) @@ -2659,25 +2668,7 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs ret = do_pcm_hwsync(substream); if (!ret) ret = forward_appl_ptr(substream, frames, - snd_pcm_playback_avail(runtime)); - snd_pcm_stream_unlock_irq(substream); - return ret; -} - -static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *substream, - snd_pcm_uframes_t frames) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_sframes_t ret; - - if (frames == 0) - return 0; - - snd_pcm_stream_lock_irq(substream); - ret = do_pcm_hwsync(substream); - if (!ret) - ret = forward_appl_ptr(substream, frames, - snd_pcm_capture_avail(runtime)); + snd_pcm_avail(substream)); snd_pcm_stream_unlock_irq(substream); return ret; } @@ -2695,19 +2686,13 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) static int snd_pcm_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *delay) { - struct snd_pcm_runtime *runtime = substream->runtime; int err; snd_pcm_sframes_t n = 0; snd_pcm_stream_lock_irq(substream); err = do_pcm_hwsync(substream); - if (!err) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - n = snd_pcm_playback_hw_avail(runtime); - else - n = snd_pcm_capture_avail(runtime); - n += runtime->delay; - } + if (!err) + n = snd_pcm_calc_delay(substream); snd_pcm_stream_unlock_irq(substream); if (!err) *delay = n; @@ -2834,10 +2819,7 @@ static int snd_pcm_rewind_ioctl(struct snd_pcm_substream *substream, return -EFAULT; if (put_user(0, _frames)) return -EFAULT; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - result = snd_pcm_playback_rewind(substream, frames); - else - result = snd_pcm_capture_rewind(substream, frames); + result = snd_pcm_rewind(substream, frames); __put_user(result, _frames); return result < 0 ? result : 0; } @@ -2852,10 +2834,7 @@ static int snd_pcm_forward_ioctl(struct snd_pcm_substream *substream, return -EFAULT; if (put_user(0, _frames)) return -EFAULT; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - result = snd_pcm_playback_forward(substream, frames); - else - result = snd_pcm_capture_forward(substream, frames); + result = snd_pcm_forward(substream, frames); __put_user(result, _frames); return result < 0 ? result : 0; } @@ -2998,7 +2977,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, /* provided only for OSS; capture-only and no value returned */ if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) return -EINVAL; - result = snd_pcm_capture_forward(substream, *frames); + result = snd_pcm_forward(substream, *frames); return result < 0 ? result : 0; } case SNDRV_PCM_IOCTL_HW_PARAMS: @@ -3093,7 +3072,7 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) if (!frame_aligned(runtime, to->iov->iov_len)) return -EINVAL; frames = bytes_to_samples(runtime, to->iov->iov_len); - bufs = kmalloc(sizeof(void *) * to->nr_segs, GFP_KERNEL); + bufs = kmalloc_array(to->nr_segs, sizeof(void *), GFP_KERNEL); if (bufs == NULL) return -ENOMEM; for (i = 0; i < to->nr_segs; ++i) @@ -3128,7 +3107,7 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) !frame_aligned(runtime, from->iov->iov_len)) return -EINVAL; frames = bytes_to_samples(runtime, from->iov->iov_len); - bufs = kmalloc(sizeof(void *) * from->nr_segs, GFP_KERNEL); + bufs = kmalloc_array(from->nr_segs, sizeof(void *), GFP_KERNEL); if (bufs == NULL) return -ENOMEM; for (i = 0; i < from->nr_segs; ++i) @@ -3140,82 +3119,46 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) return result; } -static __poll_t snd_pcm_playback_poll(struct file *file, poll_table * wait) +static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; - __poll_t mask; + __poll_t mask, ok; snd_pcm_uframes_t avail; pcm_file = file->private_data; substream = pcm_file->substream; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ok = EPOLLOUT | EPOLLWRNORM; + else + ok = EPOLLIN | EPOLLRDNORM; if (PCM_RUNTIME_CHECK(substream)) - return EPOLLOUT | EPOLLWRNORM | EPOLLERR; - runtime = substream->runtime; - - poll_wait(file, &runtime->sleep, wait); + return ok | EPOLLERR; - snd_pcm_stream_lock_irq(substream); - avail = snd_pcm_playback_avail(runtime); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_RUNNING: - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_PAUSED: - if (avail >= runtime->control->avail_min) { - mask = EPOLLOUT | EPOLLWRNORM; - break; - } - /* Fall through */ - case SNDRV_PCM_STATE_DRAINING: - mask = 0; - break; - default: - mask = EPOLLOUT | EPOLLWRNORM | EPOLLERR; - break; - } - snd_pcm_stream_unlock_irq(substream); - return mask; -} - -static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait) -{ - struct snd_pcm_file *pcm_file; - struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; - __poll_t mask; - snd_pcm_uframes_t avail; - - pcm_file = file->private_data; - - substream = pcm_file->substream; - if (PCM_RUNTIME_CHECK(substream)) - return EPOLLIN | EPOLLRDNORM | EPOLLERR; runtime = substream->runtime; - poll_wait(file, &runtime->sleep, wait); + mask = 0; snd_pcm_stream_lock_irq(substream); - avail = snd_pcm_capture_avail(runtime); + avail = snd_pcm_avail(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_PAUSED: - if (avail >= runtime->control->avail_min) { - mask = EPOLLIN | EPOLLRDNORM; - break; - } - mask = 0; + if (avail >= runtime->control->avail_min) + mask = ok; break; case SNDRV_PCM_STATE_DRAINING: - if (avail > 0) { - mask = EPOLLIN | EPOLLRDNORM; - break; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + mask = ok; + if (!avail) + mask |= EPOLLERR; } - /* Fall through */ + break; default: - mask = EPOLLIN | EPOLLRDNORM | EPOLLERR; + mask = ok | EPOLLERR; break; } snd_pcm_stream_unlock_irq(substream); @@ -3707,7 +3650,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .open = snd_pcm_playback_open, .release = snd_pcm_release, .llseek = no_llseek, - .poll = snd_pcm_playback_poll, + .poll = snd_pcm_poll, .unlocked_ioctl = snd_pcm_ioctl, .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, @@ -3721,7 +3664,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .open = snd_pcm_capture_open, .release = snd_pcm_release, .llseek = no_llseek, - .poll = snd_pcm_capture_poll, + .poll = snd_pcm_poll, .unlocked_ioctl = snd_pcm_ioctl, .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 61a07fe34cd2..56ca78423040 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2004,7 +2004,8 @@ static int snd_seq_ioctl_query_next_client(struct snd_seq_client *client, struct snd_seq_client *cptr = NULL; /* search for next client */ - info->client++; + if (info->client < INT_MAX) + info->client++; if (info->client < 0) info->client = 0; for (; info->client < SNDRV_SEQ_MAX_CLIENTS; info->client++) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ab1112e90f88..a4c8543176b2 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -389,7 +389,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) if (snd_BUG_ON(!pool)) return -EINVAL; - cellptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); + cellptr = vmalloc(array_size(sizeof(struct snd_seq_event_cell), + pool->size)); if (!cellptr) return -ENOMEM; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 9e2912e3e80f..288f839a554b 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -657,7 +657,7 @@ static struct snd_midi_channel *snd_midi_channel_init_set(int n) struct snd_midi_channel *chan; int i; - chan = kmalloc(n * sizeof(struct snd_midi_channel), GFP_KERNEL); + chan = kmalloc_array(n, sizeof(struct snd_midi_channel), GFP_KERNEL); if (chan) { for (i = 0; i < n; i++) snd_midi_channel_init(chan+i, i); diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d21ece9f8d73..24d90abfc64d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -669,7 +669,7 @@ int snd_seq_event_port_attach(int client, /* Set up the port */ memset(&portinfo, 0, sizeof(portinfo)); portinfo.addr.client = client; - strlcpy(portinfo.name, portname ? portname : "Unamed port", + strlcpy(portinfo.name, portname ? portname : "Unnamed port", sizeof(portinfo.name)); portinfo.capability = cap; diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 23167578231f..f587d0e27476 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -371,9 +371,7 @@ static int initialize_timer(struct snd_seq_timer *tmr) tmr->ticks = 1; if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE)) { - unsigned long r = t->hw.resolution; - if (! r && t->hw.c_resolution) - r = t->hw.c_resolution(t); + unsigned long r = snd_timer_resolution(tmr->timeri); if (r) { tmr->ticks = (unsigned int)(1000000000uL / (r * freq)); if (! tmr->ticks) diff --git a/sound/core/timer.c b/sound/core/timer.c index 0ddcae495838..b6f076bbc72d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -427,25 +427,35 @@ int snd_timer_close(struct snd_timer_instance *timeri) } EXPORT_SYMBOL(snd_timer_close); +static unsigned long snd_timer_hw_resolution(struct snd_timer *timer) +{ + if (timer->hw.c_resolution) + return timer->hw.c_resolution(timer); + else + return timer->hw.resolution; +} + unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) { struct snd_timer * timer; + unsigned long ret = 0; + unsigned long flags; if (timeri == NULL) return 0; timer = timeri->timer; if (timer) { - if (timer->hw.c_resolution) - return timer->hw.c_resolution(timer); - return timer->hw.resolution; + spin_lock_irqsave(&timer->lock, flags); + ret = snd_timer_hw_resolution(timer); + spin_unlock_irqrestore(&timer->lock, flags); } - return 0; + return ret; } EXPORT_SYMBOL(snd_timer_resolution); static void snd_timer_notify1(struct snd_timer_instance *ti, int event) { - struct snd_timer *timer; + struct snd_timer *timer = ti->timer; unsigned long resolution = 0; struct snd_timer_instance *ts; struct timespec tstamp; @@ -457,14 +467,14 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_START || event > SNDRV_TIMER_EVENT_PAUSE)) return; - if (event == SNDRV_TIMER_EVENT_START || - event == SNDRV_TIMER_EVENT_CONTINUE) - resolution = snd_timer_resolution(ti); + if (timer && + (event == SNDRV_TIMER_EVENT_START || + event == SNDRV_TIMER_EVENT_CONTINUE)) + resolution = snd_timer_hw_resolution(timer); if (ti->ccallback) ti->ccallback(ti, event, &tstamp, resolution); if (ti->flags & SNDRV_TIMER_IFLG_SLAVE) return; - timer = ti->timer; if (timer == NULL) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) @@ -771,10 +781,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) spin_lock_irqsave(&timer->lock, flags); /* remember the current resolution */ - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; + resolution = snd_timer_hw_resolution(timer); /* loop for all active instances * Here we cannot use list_for_each_entry because the active_list of a @@ -1014,12 +1021,8 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstam spin_lock_irqsave(&timer->lock, flags); if (event == SNDRV_TIMER_EVENT_MSTART || event == SNDRV_TIMER_EVENT_MCONTINUE || - event == SNDRV_TIMER_EVENT_MRESUME) { - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; - } + event == SNDRV_TIMER_EVENT_MRESUME) + resolution = snd_timer_hw_resolution(timer); list_for_each_entry(ti, &timer->active_list_head, active_list) { if (ti->ccallback) ti->ccallback(ti, event, tstamp, resolution); @@ -1517,7 +1520,7 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) } else { if (id.subdevice < 0) id.subdevice = 0; - else + else if (id.subdevice < INT_MAX) id.subdevice++; } } @@ -1656,10 +1659,8 @@ static int snd_timer_user_gstatus(struct file *file, mutex_lock(®ister_mutex); t = snd_timer_find(&tid); if (t != NULL) { - if (t->hw.c_resolution) - gstatus.resolution = t->hw.c_resolution(t); - else - gstatus.resolution = t->hw.resolution; + spin_lock_irq(&t->lock); + gstatus.resolution = snd_timer_hw_resolution(t); if (t->hw.precise_resolution) { t->hw.precise_resolution(t, &gstatus.resolution_num, &gstatus.resolution_den); @@ -1667,6 +1668,7 @@ static int snd_timer_user_gstatus(struct file *file, gstatus.resolution_num = gstatus.resolution; gstatus.resolution_den = 1000000000uL; } + spin_unlock_irq(&t->lock); } else { err = -ENODEV; } diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 9e96186742d0..fd99d8abe2af 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -259,8 +259,8 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, struct link_master *master_link = snd_kcontrol_chip(master); struct link_slave *srec; - srec = kzalloc(sizeof(*srec) + - slave->count * sizeof(*slave->vd), GFP_KERNEL); + srec = kzalloc(struct_size(srec, slave.vd, slave->count), + GFP_KERNEL); if (!srec) return -ENOMEM; srec->kctl = slave; @@ -421,13 +421,15 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv && - (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE || - tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX || - tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) { - kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - memcpy(master->tlv, tlv, sizeof(master->tlv)); - kctl->tlv.p = master->tlv; + if (tlv) { + unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE]; + if (type == SNDRV_CTL_TLVT_DB_SCALE || + type == SNDRV_CTL_TLVT_DB_MINMAX || + type == SNDRV_CTL_TLVT_DB_MINMAX_MUTE) { + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + memcpy(master->tlv, tlv, sizeof(master->tlv)); + kctl->tlv.p = master->tlv; + } } return kctl; diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 7144cc36e8ae..648a12da44f9 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -153,7 +153,7 @@ config SND_SERIAL_U16550 select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here - and read <file:Documentation/sound/alsa/serial-u16550.txt>. + and read <file:Documentation/sound/cards/serial-u16550.rst>. This driver works with serial UARTs 16550 and better. This driver accesses the serial port hardware directly, so @@ -223,7 +223,7 @@ config SND_AC97_POWER_SAVE the device frequently. A value of 10 seconds would be a good choice for normal operations. - See Documentation/sound/alsa/powersave.txt for more details. + See Documentation/sound/designs/powersave.rst for more details. config SND_AC97_POWER_SAVE_DEFAULT int "Default time-out for AC97 power-save mode" diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index eab7f594ebe7..78a2fdc38531 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -768,20 +768,7 @@ static int loopback_close(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops loopback_playback_ops = { - .open = loopback_open, - .close = loopback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = loopback_hw_params, - .hw_free = loopback_hw_free, - .prepare = loopback_prepare, - .trigger = loopback_trigger, - .pointer = loopback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - -static const struct snd_pcm_ops loopback_capture_ops = { +static const struct snd_pcm_ops loopback_pcm_ops = { .open = loopback_open, .close = loopback_close, .ioctl = snd_pcm_lib_ioctl, @@ -804,8 +791,8 @@ static int loopback_pcm_new(struct loopback *loopback, substreams, substreams, &pcm); if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_pcm_ops); pcm->private_data = loopback; pcm->info_flags = 0; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 8fb9a54fe8ba..9af154db530a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1042,7 +1042,7 @@ static void dummy_proc_init(struct snd_dummy *chip) if (!snd_card_proc_new(chip->card, "dummy_pcm", &entry)) { snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = chip; } } diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index f32e81342247..b68e71ca7abd 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -41,11 +41,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Matthias Koenig <mk@phasorlab.de>"); diff --git a/sound/drivers/opl4/opl4_proc.c b/sound/drivers/opl4/opl4_proc.c index cd2c07fa2ef4..16b24091d799 100644 --- a/sound/drivers/opl4/opl4_proc.c +++ b/sound/drivers/opl4/opl4_proc.c @@ -104,7 +104,7 @@ int snd_opl4_create_proc(struct snd_opl4 *opl4) if (entry) { if (opl4->hardware < OPL3_HW_OPL4_ML) { /* OPL4 can access 4 MB external ROM/SRAM */ - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->size = 4 * 1024 * 1024; } else { /* OPL4-ML has 1 MB internal ROM */ diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index ec8a94325ef6..3cdf0a88d71b 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -60,11 +60,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Levent Guendogdu, Tobias Gehrig, Matthias Koenig"); diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index ec24f96794f5..8096891af913 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -183,7 +183,7 @@ void snd_bebob_proc_init(struct snd_bebob *bebob) bebob->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 55b4be9b0034..37062a233f6a 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,4 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ - dice-pcm.o dice-hwdep.o dice.o + dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ + dice-alesis.o dice-extension.o dice-mytek.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c new file mode 100644 index 000000000000..b2efb1c71a98 --- /dev/null +++ b/sound/firewire/dice/dice-alesis.c @@ -0,0 +1,52 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-alesis.c - a part of driver for DICE based devices + * + * Copyright (c) 2018 Takashi Sakamoto + */ + +#include "dice.h" + +static const unsigned int +alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { + {6, 6, 4}, /* Tx0 = Analog + S/PDIF. */ + {8, 4, 0}, /* Tx1 = ADAT1. */ +}; + +static const unsigned int +alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { + {10, 10, 8}, /* Tx0 = Analog + S/PDIF. */ + {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */ +}; + +int snd_dice_detect_alesis_formats(struct snd_dice *dice) +{ + __be32 reg; + u32 data; + int i; + int err; + + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, ®, + sizeof(reg)); + if (err < 0) + return err; + data = be32_to_cpu(reg); + + if (data == 4 || data == 6) { + memcpy(dice->tx_pcm_chs, alesis_io14_tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * + sizeof(unsigned int)); + } else { + memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * + sizeof(unsigned int)); + } + + for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i) + dice->rx_pcm_chs[0][i] = 8; + + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + + return 0; +} diff --git a/sound/firewire/dice/dice-extension.c b/sound/firewire/dice/dice-extension.c new file mode 100644 index 000000000000..a63fcbc875ad --- /dev/null +++ b/sound/firewire/dice/dice-extension.c @@ -0,0 +1,172 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-extension.c - a part of driver for DICE based devices + * + * Copyright (c) 2018 Takashi Sakamoto + */ + +#include "dice.h" + +/* For TCD2210/2220, TCAT defines extension of application protocol. */ + +#define DICE_EXT_APP_SPACE 0xffffe0200000uLL + +#define DICE_EXT_APP_CAPS_OFFSET 0x00 +#define DICE_EXT_APP_CAPS_SIZE 0x04 +#define DICE_EXT_APP_CMD_OFFSET 0x08 +#define DICE_EXT_APP_CMD_SIZE 0x0c +#define DICE_EXT_APP_MIXER_OFFSET 0x10 +#define DICE_EXT_APP_MIXER_SIZE 0x14 +#define DICE_EXT_APP_PEAK_OFFSET 0x18 +#define DICE_EXT_APP_PEAK_SIZE 0x1c +#define DICE_EXT_APP_ROUTER_OFFSET 0x20 +#define DICE_EXT_APP_ROUTER_SIZE 0x24 +#define DICE_EXT_APP_STREAM_OFFSET 0x28 +#define DICE_EXT_APP_STREAM_SIZE 0x2c +#define DICE_EXT_APP_CURRENT_OFFSET 0x30 +#define DICE_EXT_APP_CURRENT_SIZE 0x34 +#define DICE_EXT_APP_STANDALONE_OFFSET 0x38 +#define DICE_EXT_APP_STANDALONE_SIZE 0x3c +#define DICE_EXT_APP_APPLICATION_OFFSET 0x40 +#define DICE_EXT_APP_APPLICATION_SIZE 0x44 + +#define EXT_APP_STREAM_TX_NUMBER 0x0000 +#define EXT_APP_STREAM_RX_NUMBER 0x0004 +#define EXT_APP_STREAM_ENTRIES 0x0008 +#define EXT_APP_STREAM_ENTRY_SIZE 0x010c +#define EXT_APP_NUMBER_AUDIO 0x0000 +#define EXT_APP_NUMBER_MIDI 0x0004 +#define EXT_APP_NAMES 0x0008 +#define EXT_APP_NAMES_SIZE 256 +#define EXT_APP_AC3 0x0108 + +#define EXT_APP_CONFIG_LOW_ROUTER 0x0000 +#define EXT_APP_CONFIG_LOW_STREAM 0x1000 +#define EXT_APP_CONFIG_MIDDLE_ROUTER 0x2000 +#define EXT_APP_CONFIG_MIDDLE_STREAM 0x3000 +#define EXT_APP_CONFIG_HIGH_ROUTER 0x4000 +#define EXT_APP_CONFIG_HIGH_STREAM 0x5000 + +static inline int read_transaction(struct snd_dice *dice, u64 section_addr, + u32 offset, void *buf, size_t len) +{ + return snd_fw_transaction(dice->unit, + len == 4 ? TCODE_READ_QUADLET_REQUEST : + TCODE_READ_BLOCK_REQUEST, + section_addr + offset, buf, len, 0); +} + +static int read_stream_entries(struct snd_dice *dice, u64 section_addr, + u32 base_offset, unsigned int stream_count, + unsigned int mode, + unsigned int pcm_channels[MAX_STREAMS][3], + unsigned int midi_ports[MAX_STREAMS]) +{ + u32 entry_offset; + __be32 reg[2]; + int err; + int i; + + for (i = 0; i < stream_count; ++i) { + entry_offset = base_offset + i * EXT_APP_STREAM_ENTRY_SIZE; + err = read_transaction(dice, section_addr, + entry_offset + EXT_APP_NUMBER_AUDIO, + reg, sizeof(reg)); + if (err < 0) + return err; + pcm_channels[i][mode] = be32_to_cpu(reg[0]); + midi_ports[i] = max(midi_ports[i], be32_to_cpu(reg[1])); + } + + return 0; +} + +static int detect_stream_formats(struct snd_dice *dice, u64 section_addr) +{ + u32 base_offset; + __be32 reg[2]; + unsigned int stream_count; + int mode; + int err = 0; + + for (mode = 0; mode < SND_DICE_RATE_MODE_COUNT; ++mode) { + unsigned int cap; + + /* + * Some models report stream formats at highest mode, however + * they don't support the mode. Check clock capabilities. + */ + if (mode == 2) { + cap = CLOCK_CAP_RATE_176400 | CLOCK_CAP_RATE_192000; + } else if (mode == 1) { + cap = CLOCK_CAP_RATE_88200 | CLOCK_CAP_RATE_96000; + } else { + cap = CLOCK_CAP_RATE_32000 | CLOCK_CAP_RATE_44100 | + CLOCK_CAP_RATE_48000; + } + if (!(cap & dice->clock_caps)) + continue; + + base_offset = 0x2000 * mode + 0x1000; + + err = read_transaction(dice, section_addr, + base_offset + EXT_APP_STREAM_TX_NUMBER, + ®, sizeof(reg)); + if (err < 0) + break; + + base_offset += EXT_APP_STREAM_ENTRIES; + stream_count = be32_to_cpu(reg[0]); + err = read_stream_entries(dice, section_addr, base_offset, + stream_count, mode, + dice->tx_pcm_chs, + dice->tx_midi_ports); + if (err < 0) + break; + + base_offset += stream_count * EXT_APP_STREAM_ENTRY_SIZE; + stream_count = be32_to_cpu(reg[1]); + err = read_stream_entries(dice, section_addr, base_offset, + stream_count, + mode, dice->rx_pcm_chs, + dice->rx_midi_ports); + if (err < 0) + break; + } + + return err; +} + +int snd_dice_detect_extension_formats(struct snd_dice *dice) +{ + __be32 *pointers; + unsigned int i; + u64 section_addr; + int err; + + pointers = kmalloc_array(9, sizeof(__be32) * 2, GFP_KERNEL); + if (pointers == NULL) + return -ENOMEM; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_EXT_APP_SPACE, pointers, + 9 * sizeof(__be32) * 2, 0); + if (err < 0) + goto end; + + /* Check two of them for offset have the same value or not. */ + for (i = 0; i < 9; ++i) { + int j; + + for (j = i + 1; j < 9; ++j) { + if (pointers[i * 2] == pointers[j * 2]) + goto end; + } + } + + section_addr = DICE_EXT_APP_SPACE + be32_to_cpu(pointers[12]) * 4; + err = detect_stream_formats(dice, section_addr); +end: + kfree(pointers); + return err; +} diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 15a484b05298..9cad3d608229 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -175,13 +175,18 @@ #define GLOBAL_SAMPLE_RATE 0x05c /* + * Some old firmware versions do not have the following global registers. + * Windows drivers produced by TCAT lost backward compatibility in its + * early release because they can handle firmware only which supports the + * following registers. + */ + +/* * The version of the DICE driver specification that this device conforms to; * read-only. */ #define GLOBAL_VERSION 0x060 -/* Some old firmware versions do not have the following global registers: */ - /* * Supported sample rates and clock sources; read-only. */ diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c index 8ff6da3c51f7..84eca8a51a02 100644 --- a/sound/firewire/dice/dice-midi.c +++ b/sound/firewire/dice/dice-midi.c @@ -101,27 +101,18 @@ int snd_dice_create_midi(struct snd_dice *dice) .close = midi_close, .trigger = midi_playback_trigger, }; - __be32 reg; struct snd_rawmidi *rmidi; struct snd_rawmidi_str *str; unsigned int midi_in_ports, midi_out_ports; + int i; int err; - /* - * Use the number of MIDI conformant data channel at current sampling - * transfer frequency. - */ - err = snd_dice_transaction_read_tx(dice, TX_NUMBER_MIDI, - ®, sizeof(reg)); - if (err < 0) - return err; - midi_in_ports = be32_to_cpu(reg); - - err = snd_dice_transaction_read_rx(dice, RX_NUMBER_MIDI, - ®, sizeof(reg)); - if (err < 0) - return err; - midi_out_ports = be32_to_cpu(reg); + midi_in_ports = 0; + midi_out_ports = 0; + for (i = 0; i < MAX_STREAMS; ++i) { + midi_in_ports = max(midi_in_ports, dice->tx_midi_ports[i]); + midi_out_ports = max(midi_out_ports, dice->rx_midi_ports[i]); + } if (midi_in_ports + midi_out_ports == 0) return 0; diff --git a/sound/firewire/dice/dice-mytek.c b/sound/firewire/dice/dice-mytek.c new file mode 100644 index 000000000000..eb7d5492d10b --- /dev/null +++ b/sound/firewire/dice/dice-mytek.c @@ -0,0 +1,46 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-mytek.c - a part of driver for DICE based devices + * + * Copyright (c) 2018 Melvin Vermeeren + */ + +#include "dice.h" + +struct dice_mytek_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; +}; + +static const struct dice_mytek_spec stereo_192_dsd_dac = { + /* AES, TOSLINK, SPDIF, ADAT inputs on device */ + .tx_pcm_chs = {{8, 8, 8}, {0, 0, 0} }, + /* PCM 44.1-192, native DSD64/DSD128 to device */ + .rx_pcm_chs = {{4, 4, 4}, {0, 0, 0} } +}; + +/* + * Mytek has a few other firewire-capable devices, though newer models appear + * to lack the port more often than not. As I don't have access to any of them + * they are missing here. An example is the Mytek 8x192 ADDA, which is DICE. + */ + +int snd_dice_detect_mytek_formats(struct snd_dice *dice) +{ + int i; + const struct dice_mytek_spec *dev; + + dev = &stereo_192_dsd_dac; + + memcpy(dice->tx_pcm_chs, dev->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, dev->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + for (i = 0; i < MAX_STREAMS; ++i) { + dice->tx_midi_ports[i] = 0; + dice->rx_midi_ports[i] = 0; + } + + return 0; +} diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 7cb9e9713ac3..80351b29fe0d 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -9,43 +9,115 @@ #include "dice.h" +static int dice_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_substream *substream = rule->private; + struct snd_dice *dice = substream->private_data; + unsigned int index = substream->pcm->device; + + const struct snd_interval *c = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval rates = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int *pcm_channels; + enum snd_dice_rate_mode mode; + unsigned int i, rate; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + pcm_channels = dice->tx_pcm_chs[index]; + else + pcm_channels = dice->rx_pcm_chs[index]; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + + if (!snd_interval_test(c, pcm_channels[mode])) + continue; + + rates.min = min(rates.min, rate); + rates.max = max(rates.max, rate); + } + + return snd_interval_refine(r, &rates); +} + +static int dice_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_substream *substream = rule->private; + struct snd_dice *dice = substream->private_data; + unsigned int index = substream->pcm->device; + + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *c = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval channels = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int *pcm_channels; + enum snd_dice_rate_mode mode; + unsigned int i, rate; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + pcm_channels = dice->tx_pcm_chs[index]; + else + pcm_channels = dice->rx_pcm_chs[index]; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + + if (!snd_interval_test(r, rate)) + continue; + + channels.min = min(channels.min, pcm_channels[mode]); + channels.max = max(channels.max, pcm_channels[mode]); + } + + return snd_interval_refine(c, &channels); +} + static int limit_channels_and_rates(struct snd_dice *dice, struct snd_pcm_runtime *runtime, enum amdtp_stream_direction dir, - unsigned int index, unsigned int size) + unsigned int index) { struct snd_pcm_hardware *hw = &runtime->hw; - struct amdtp_stream *stream; - unsigned int rate; - __be32 reg; - int err; - - /* - * Retrieve current Multi Bit Linear Audio data channel and limit to - * it. - */ - if (dir == AMDTP_IN_STREAM) { - stream = &dice->tx_stream[index]; - err = snd_dice_transaction_read_tx(dice, - size * index + TX_NUMBER_AUDIO, - ®, sizeof(reg)); - } else { - stream = &dice->rx_stream[index]; - err = snd_dice_transaction_read_rx(dice, - size * index + RX_NUMBER_AUDIO, - ®, sizeof(reg)); + unsigned int *pcm_channels; + unsigned int i; + + if (dir == AMDTP_IN_STREAM) + pcm_channels = dice->tx_pcm_chs[index]; + else + pcm_channels = dice->rx_pcm_chs[index]; + + hw->channels_min = UINT_MAX; + hw->channels_max = 0; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + enum snd_dice_rate_mode mode; + unsigned int rate, channels; + + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + hw->rates |= snd_pcm_rate_to_rate_bit(rate); + + channels = pcm_channels[mode]; + if (channels == 0) + continue; + hw->channels_min = min(hw->channels_min, channels); + hw->channels_max = max(hw->channels_max, channels); } - if (err < 0) - return err; - hw->channels_min = hw->channels_max = be32_to_cpu(reg); - - /* Retrieve current sampling transfer frequency and limit to it. */ - err = snd_dice_transaction_get_rate(dice, &rate); - if (err < 0) - return err; - - hw->rates = snd_pcm_rate_to_rate_bit(rate); snd_pcm_limit_hw_rates(runtime); return 0; @@ -56,36 +128,34 @@ static int init_hw_info(struct snd_dice *dice, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + unsigned int index = substream->pcm->device; enum amdtp_stream_direction dir; struct amdtp_stream *stream; - __be32 reg[2]; - unsigned int count, size; int err; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = AM824_IN_PCM_FORMAT_BITS; dir = AMDTP_IN_STREAM; - stream = &dice->tx_stream[substream->pcm->device]; - err = snd_dice_transaction_read_tx(dice, TX_NUMBER, reg, - sizeof(reg)); + stream = &dice->tx_stream[index]; } else { hw->formats = AM824_OUT_PCM_FORMAT_BITS; dir = AMDTP_OUT_STREAM; - stream = &dice->rx_stream[substream->pcm->device]; - err = snd_dice_transaction_read_rx(dice, RX_NUMBER, reg, - sizeof(reg)); + stream = &dice->rx_stream[index]; } + err = limit_channels_and_rates(dice, substream->runtime, dir, + index); if (err < 0) return err; - count = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); - if (substream->pcm->device >= count) - return -ENXIO; - - size = be32_to_cpu(reg[1]) * 4; - err = limit_channels_and_rates(dice, substream->runtime, dir, - substream->pcm->device, size); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + dice_rate_constraint, substream, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + dice_channels_constraint, substream, + SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; @@ -95,6 +165,8 @@ static int init_hw_info(struct snd_dice *dice, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; + unsigned int source; + bool internal; int err; err = snd_dice_stream_lock_try(dice); @@ -105,6 +177,43 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; + err = snd_dice_transaction_get_clock_source(dice, &source); + if (err < 0) + goto err_locked; + switch (source) { + case CLOCK_SOURCE_AES1: + case CLOCK_SOURCE_AES2: + case CLOCK_SOURCE_AES3: + case CLOCK_SOURCE_AES4: + case CLOCK_SOURCE_AES_ANY: + case CLOCK_SOURCE_ADAT: + case CLOCK_SOURCE_TDIF: + case CLOCK_SOURCE_WC: + internal = false; + break; + default: + internal = true; + break; + } + + /* + * When source of clock is not internal or any PCM streams are running, + * available sampling rate is limited at current sampling rate. + */ + if (!internal || + amdtp_stream_pcm_running(&dice->tx_stream[0]) || + amdtp_stream_pcm_running(&dice->tx_stream[1]) || + amdtp_stream_pcm_running(&dice->rx_stream[0]) || + amdtp_stream_pcm_running(&dice->rx_stream[1])) { + unsigned int rate; + + err = snd_dice_transaction_get_rate(dice, &rate); + if (err < 0) + goto err_locked; + substream->runtime->hw.rate_min = rate; + substream->runtime->hw.rate_max = rate; + } + snd_pcm_set_sync(substream); end: return err; @@ -318,37 +427,19 @@ int snd_dice_create_pcm(struct snd_dice *dice) .page = snd_pcm_lib_get_vmalloc_page, .mmap = snd_pcm_lib_mmap_vmalloc, }; - __be32 reg; struct snd_pcm *pcm; - unsigned int i, max_capture, max_playback, capture, playback; + unsigned int capture, playback; + int i, j; int err; - /* Check whether PCM substreams are required. */ - if (dice->force_two_pcms) { - max_capture = max_playback = 2; - } else { - max_capture = max_playback = 0; - err = snd_dice_transaction_read_tx(dice, TX_NUMBER, ®, - sizeof(reg)); - if (err < 0) - return err; - max_capture = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS); - - err = snd_dice_transaction_read_rx(dice, RX_NUMBER, ®, - sizeof(reg)); - if (err < 0) - return err; - max_playback = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS); - } - for (i = 0; i < MAX_STREAMS; i++) { capture = playback = 0; - if (i < max_capture) - capture = 1; - if (i < max_playback) - playback = 1; - if (capture == 0 && playback == 0) - break; + for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j) { + if (dice->tx_pcm_chs[i][j] > 0) + capture = 1; + if (dice->rx_pcm_chs[i][j] > 0) + playback = 1; + } err = snd_pcm_new(dice->card, "DICE", i, playback, capture, &pcm); diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1bced59..bb870fc73f99 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry, >> CLOCK_RATE_SHIFT)); snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); if (quadlets >= 90) { + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); snd_iprintf(buffer, " clock caps:"); for (i = 0; i <= 6; ++i) if (buf.global.clock_caps & (1 << i)) @@ -243,10 +243,74 @@ static void dice_proc_read(struct snd_info_entry *entry, } } -void snd_dice_create_proc(struct snd_dice *dice) +static void dice_proc_read_formation(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + static const char *const rate_labels[] = { + [SND_DICE_RATE_MODE_LOW] = "low", + [SND_DICE_RATE_MODE_MIDDLE] = "middle", + [SND_DICE_RATE_MODE_HIGH] = "high", + }; + struct snd_dice *dice = entry->private_data; + int i, j; + + snd_iprintf(buffer, "Output stream from unit:\n"); + for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i) + snd_iprintf(buffer, "\t%s", rate_labels[i]); + snd_iprintf(buffer, "\tMIDI\n"); + for (i = 0; i < MAX_STREAMS; ++i) { + snd_iprintf(buffer, "Tx %u:", i); + for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j) + snd_iprintf(buffer, "\t%u", dice->tx_pcm_chs[i][j]); + snd_iprintf(buffer, "\t%u\n", dice->tx_midi_ports[i]); + } + + snd_iprintf(buffer, "Input stream to unit:\n"); + for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i) + snd_iprintf(buffer, "\t%s", rate_labels[i]); + snd_iprintf(buffer, "\n"); + for (i = 0; i < MAX_STREAMS; ++i) { + snd_iprintf(buffer, "Rx %u:", i); + for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j) + snd_iprintf(buffer, "\t%u", dice->rx_pcm_chs[i][j]); + snd_iprintf(buffer, "\t%u\n", dice->rx_midi_ports[i]); + } +} + +static void add_node(struct snd_dice *dice, struct snd_info_entry *root, + const char *name, + void (*op)(struct snd_info_entry *entry, + struct snd_info_buffer *buffer)) { struct snd_info_entry *entry; - if (!snd_card_proc_new(dice->card, "dice", &entry)) - snd_info_set_text_ops(entry, dice, dice_proc_read); + entry = snd_info_create_card_entry(dice->card, name, root); + if (!entry) + return; + + snd_info_set_text_ops(entry, dice, op); + if (snd_info_register(entry) < 0) + snd_info_free_entry(entry); +} + +void snd_dice_create_proc(struct snd_dice *dice) +{ + struct snd_info_entry *root; + + /* + * All nodes are automatically removed at snd_card_disconnect(), + * by following to link list. + */ + root = snd_info_create_card_entry(dice->card, "firewire", + dice->card->proc_root); + if (!root) + return; + root->mode = S_IFDIR | 0555; + if (snd_info_register(root) < 0) { + snd_info_free_entry(root); + return; + } + + add_node(dice, root, "dice", dice_proc_read); + add_node(dice, root, "formation", dice_proc_read_formation); } diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 928a255bfc35..c3c892c5c7ff 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -30,13 +30,43 @@ const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { [6] = 192000, }; +int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, + enum snd_dice_rate_mode *mode) +{ + /* Corresponding to each entry in snd_dice_rates. */ + static const enum snd_dice_rate_mode modes[] = { + [0] = SND_DICE_RATE_MODE_LOW, + [1] = SND_DICE_RATE_MODE_LOW, + [2] = SND_DICE_RATE_MODE_LOW, + [3] = SND_DICE_RATE_MODE_MIDDLE, + [4] = SND_DICE_RATE_MODE_MIDDLE, + [5] = SND_DICE_RATE_MODE_HIGH, + [6] = SND_DICE_RATE_MODE_HIGH, + }; + int i; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) { + if (!(dice->clock_caps & BIT(i))) + continue; + if (snd_dice_rates[i] != rate) + continue; + + *mode = modes[i]; + return 0; + } + + return -EINVAL; +} + /* * This operation has an effect to synchronize GLOBAL_STATUS/GLOBAL_SAMPLE_RATE * to GLOBAL_STATUS. Especially, just after powering on, these are different. */ -static int ensure_phase_lock(struct snd_dice *dice) +static int ensure_phase_lock(struct snd_dice *dice, unsigned int rate) { __be32 reg, nominal; + u32 data; + int i; int err; err = snd_dice_transaction_read_global(dice, GLOBAL_CLOCK_SELECT, @@ -44,9 +74,21 @@ static int ensure_phase_lock(struct snd_dice *dice) if (err < 0) return err; + data = be32_to_cpu(reg); + + data &= ~CLOCK_RATE_MASK; + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + if (snd_dice_rates[i] == rate) + break; + } + if (i == ARRAY_SIZE(snd_dice_rates)) + return -EINVAL; + data |= i << CLOCK_RATE_SHIFT; + if (completion_done(&dice->clock_accepted)) reinit_completion(&dice->clock_accepted); + reg = cpu_to_be32(data); err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT, ®, sizeof(reg)); if (err < 0) @@ -192,6 +234,7 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, unsigned int rate, struct reg_params *params) { __be32 reg[2]; + enum snd_dice_rate_mode mode; unsigned int i, pcm_chs, midi_ports; struct amdtp_stream *streams; struct fw_iso_resources *resources; @@ -206,12 +249,23 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, resources = dice->rx_resources; } + err = snd_dice_stream_get_rate_mode(dice, rate, &mode); + if (err < 0) + return err; + for (i = 0; i < params->count; i++) { + unsigned int pcm_cache; + unsigned int midi_cache; + if (dir == AMDTP_IN_STREAM) { + pcm_cache = dice->tx_pcm_chs[i][mode]; + midi_cache = dice->tx_midi_ports[i]; err = snd_dice_transaction_read_tx(dice, params->size * i + TX_NUMBER_AUDIO, reg, sizeof(reg)); } else { + pcm_cache = dice->rx_pcm_chs[i][mode]; + midi_cache = dice->rx_midi_ports[i]; err = snd_dice_transaction_read_rx(dice, params->size * i + RX_NUMBER_AUDIO, reg, sizeof(reg)); @@ -221,6 +275,14 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, pcm_chs = be32_to_cpu(reg[0]); midi_ports = be32_to_cpu(reg[1]); + /* These are important for developer of this driver. */ + if (pcm_chs != pcm_cache || midi_ports != midi_cache) { + dev_info(&dice->unit->device, + "cache mismatch: pcm: %u:%u, midi: %u:%u\n", + pcm_chs, pcm_cache, midi_ports, midi_cache); + return -EPROTO; + } + err = keep_resources(dice, dir, i, rate, pcm_chs, midi_ports); if (err < 0) return err; @@ -256,6 +318,68 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, return err; } +static int start_duplex_streams(struct snd_dice *dice, unsigned int rate) +{ + struct reg_params tx_params, rx_params; + int i; + int err; + + err = get_register_params(dice, &tx_params, &rx_params); + if (err < 0) + return err; + + /* Stop transmission. */ + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + snd_dice_transaction_clear_enable(dice); + release_resources(dice); + + err = ensure_phase_lock(dice, rate); + if (err < 0) { + dev_err(&dice->unit->device, "fail to ensure phase lock\n"); + return err; + } + + /* Likely to have changed stream formats. */ + err = get_register_params(dice, &tx_params, &rx_params); + if (err < 0) + return err; + + /* Start both streams. */ + err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params); + if (err < 0) + goto error; + err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params); + if (err < 0) + goto error; + + err = snd_dice_transaction_set_enable(dice); + if (err < 0) { + dev_err(&dice->unit->device, "fail to enable interface\n"); + goto error; + } + + for (i = 0; i < MAX_STREAMS; i++) { + if ((i < tx_params.count && + !amdtp_stream_wait_callback(&dice->tx_stream[i], + CALLBACK_TIMEOUT)) || + (i < rx_params.count && + !amdtp_stream_wait_callback(&dice->rx_stream[i], + CALLBACK_TIMEOUT))) { + err = -ETIMEDOUT; + goto error; + } + } + + return 0; +error: + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + snd_dice_transaction_clear_enable(dice); + release_resources(dice); + return err; +} + /* * MEMO: After this function, there're two states of streams: * - None streams are running. @@ -265,17 +389,13 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { unsigned int curr_rate; unsigned int i; - struct reg_params tx_params, rx_params; - bool need_to_start; + enum snd_dice_rate_mode mode; int err; if (dice->substreams_counter == 0) return -EIO; - err = get_register_params(dice, &tx_params, &rx_params); - if (err < 0) - return err; - + /* Check sampling transmission frequency. */ err = snd_dice_transaction_get_rate(dice, &curr_rate); if (err < 0) { dev_err(&dice->unit->device, @@ -285,72 +405,36 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) - return -EINVAL; + goto restart; - /* Judge to need to restart streams. */ - for (i = 0; i < MAX_STREAMS; i++) { - if (i < tx_params.count) { - if (amdtp_streaming_error(&dice->tx_stream[i]) || - !amdtp_stream_running(&dice->tx_stream[i])) - break; - } - if (i < rx_params.count) { - if (amdtp_streaming_error(&dice->rx_stream[i]) || - !amdtp_stream_running(&dice->rx_stream[i])) - break; - } + /* Check error of packet streaming. */ + for (i = 0; i < MAX_STREAMS; ++i) { + if (amdtp_streaming_error(&dice->tx_stream[i])) + break; + if (amdtp_streaming_error(&dice->rx_stream[i])) + break; } - need_to_start = (i < MAX_STREAMS); - - if (need_to_start) { - /* Stop transmission. */ - snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); - release_resources(dice); - - err = ensure_phase_lock(dice); - if (err < 0) { - dev_err(&dice->unit->device, - "fail to ensure phase lock\n"); - return err; - } + if (i < MAX_STREAMS) + goto restart; - /* Start both streams. */ - err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params); - if (err < 0) - goto error; - err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params); - if (err < 0) - goto error; - - err = snd_dice_transaction_set_enable(dice); - if (err < 0) { - dev_err(&dice->unit->device, - "fail to enable interface\n"); - goto error; - } - - for (i = 0; i < MAX_STREAMS; i++) { - if ((i < tx_params.count && - !amdtp_stream_wait_callback(&dice->tx_stream[i], - CALLBACK_TIMEOUT)) || - (i < rx_params.count && - !amdtp_stream_wait_callback(&dice->rx_stream[i], - CALLBACK_TIMEOUT))) { - err = -ETIMEDOUT; - goto error; - } - } + /* Check required streams are running or not. */ + err = snd_dice_stream_get_rate_mode(dice, rate, &mode); + if (err < 0) + return err; + for (i = 0; i < MAX_STREAMS; ++i) { + if (dice->tx_pcm_chs[i][mode] > 0 && + !amdtp_stream_running(&dice->tx_stream[i])) + break; + if (dice->rx_pcm_chs[i][mode] > 0 && + !amdtp_stream_running(&dice->rx_stream[i])) + break; } + if (i < MAX_STREAMS) + goto restart; - return err; -error: - snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); - release_resources(dice); - return err; + return 0; +restart: + return start_duplex_streams(dice, rate); } /* @@ -484,6 +568,69 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice) } } +int snd_dice_stream_detect_current_formats(struct snd_dice *dice) +{ + unsigned int rate; + enum snd_dice_rate_mode mode; + __be32 reg[2]; + struct reg_params tx_params, rx_params; + int i; + int err; + + /* If extended protocol is available, detect detail spec. */ + err = snd_dice_detect_extension_formats(dice); + if (err >= 0) + return err; + + /* + * Available stream format is restricted at current mode of sampling + * clock. + */ + err = snd_dice_transaction_get_rate(dice, &rate); + if (err < 0) + return err; + + err = snd_dice_stream_get_rate_mode(dice, rate, &mode); + if (err < 0) + return err; + + /* + * Just after owning the unit (GLOBAL_OWNER), the unit can return + * invalid stream formats. Selecting clock parameters have an effect + * for the unit to refine it. + */ + err = ensure_phase_lock(dice, rate); + if (err < 0) + return err; + + err = get_register_params(dice, &tx_params, &rx_params); + if (err < 0) + return err; + + for (i = 0; i < tx_params.count; ++i) { + err = snd_dice_transaction_read_tx(dice, + tx_params.size * i + TX_NUMBER_AUDIO, + reg, sizeof(reg)); + if (err < 0) + return err; + dice->tx_pcm_chs[i][mode] = be32_to_cpu(reg[0]); + dice->tx_midi_ports[i] = max_t(unsigned int, + be32_to_cpu(reg[1]), dice->tx_midi_ports[i]); + } + for (i = 0; i < rx_params.count; ++i) { + err = snd_dice_transaction_read_rx(dice, + rx_params.size * i + RX_NUMBER_AUDIO, + reg, sizeof(reg)); + if (err < 0) + return err; + dice->rx_pcm_chs[i][mode] = be32_to_cpu(reg[0]); + dice->rx_midi_ports[i] = max_t(unsigned int, + be32_to_cpu(reg[1]), dice->rx_midi_ports[i]); + } + + return 0; +} + static void dice_lock_changed(struct snd_dice *dice) { dice->dev_lock_changed = true; diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c new file mode 100644 index 000000000000..a8875d24ba2a --- /dev/null +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-tc_electronic.c - a part of driver for DICE based devices + * + * Copyright (c) 2018 Takashi Sakamoto + */ + +#include "dice.h" + +struct dice_tc_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +static const struct dice_tc_spec desktop_konnekt6 = { + .tx_pcm_chs = {{6, 6, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{6, 6, 4}, {0, 0, 0} }, + .has_midi = false, +}; + +static const struct dice_tc_spec impact_twin = { + .tx_pcm_chs = {{14, 10, 6}, {0, 0, 0} }, + .rx_pcm_chs = {{14, 10, 6}, {0, 0, 0} }, + .has_midi = true, +}; + +static const struct dice_tc_spec konnekt_8 = { + .tx_pcm_chs = {{4, 4, 3}, {0, 0, 0} }, + .rx_pcm_chs = {{4, 4, 3}, {0, 0, 0} }, + .has_midi = true, +}; + +static const struct dice_tc_spec konnekt_24d = { + .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, + .has_midi = true, +}; + +static const struct dice_tc_spec konnekt_live = { + .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, + .has_midi = true, +}; + +static const struct dice_tc_spec studio_konnekt_48 = { + .tx_pcm_chs = {{16, 16, 8}, {16, 16, 7} }, + .rx_pcm_chs = {{16, 16, 8}, {14, 14, 7} }, + .has_midi = true, +}; + +static const struct dice_tc_spec digital_konnekt_x32 = { + .tx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .has_midi = false, +}; + +int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_tc_spec *spec; + } *entry, entries[] = { + {0x00000020, &konnekt_24d}, + {0x00000021, &konnekt_8}, + {0x00000022, &studio_konnekt_48}, + {0x00000023, &konnekt_live}, + {0x00000024, &desktop_konnekt6}, + {0x00000027, &impact_twin}, + {0x00000030, &digital_konnekt_x32}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 0f0350320ae8..b7e138b5abcf 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice) static int get_subaddrs(struct snd_dice *dice) { static const int min_values[10] = { - 10, 0x64 / 4, + 10, 0x60 / 4, 10, 0x18 / 4, 10, 0x18 / 4, 0, 0, @@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice) } } - /* - * Check that the implemented DICE driver specification major version - * number matches. - */ - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, sizeof(version), 0); - if (err < 0) - goto end; + if (be32_to_cpu(pointers[1]) > 0x18) { + /* + * Check that the implemented DICE driver specification major + * version number matches. + */ + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, sizeof(version), 0); + if (err < 0) + goto end; - if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { - dev_err(&dice->unit->device, - "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - err = -ENODEV; - goto end; + if ((version & cpu_to_be32(0xff000000)) != + cpu_to_be32(0x01000000)) { + dev_err(&dice->unit->device, + "unknown DICE version: 0x%08x\n", + be32_to_cpu(version)); + err = -ENODEV; + goto end; + } + + /* Set up later. */ + dice->clock_caps = 1; } dice->global_offset = be32_to_cpu(pointers[0]) * 4; dice->tx_offset = be32_to_cpu(pointers[2]) * 4; dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - dice->sync_offset = be32_to_cpu(pointers[6]) * 4; - dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; - /* Set up later. */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) - dice->clock_caps = 1; + /* Old firmware doesn't support these fields. */ + if (pointers[7]) + dice->sync_offset = be32_to_cpu(pointers[6]) * 4; + if (pointers[9]) + dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; end: kfree(pointers); return err; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96bb01b6b751..774eb2205668 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -15,40 +15,15 @@ MODULE_LICENSE("GPL v2"); #define OUI_LOUD 0x000ff2 #define OUI_FOCUSRITE 0x00130e #define OUI_TCELECTRONIC 0x000166 +#define OUI_ALESIS 0x000595 +#define OUI_MAUDIO 0x000d6c +#define OUI_MYTEK 0x001ee8 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 #define LOUD_CATEGORY_ID 0x10 -/* - * Some models support several isochronous channels, while these streams are not - * always available. In this case, add the model name to this list. - */ -static bool force_two_pcm_support(struct fw_unit *unit) -{ - static const char *const models[] = { - /* TC Electronic models. */ - "StudioKonnekt48", - /* Focusrite models. */ - "SAFFIRE_PRO_40", - "LIQUID_SAFFIRE_56", - "SAFFIRE_PRO_40_1", - }; - char model[32]; - unsigned int i; - int err; - - err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); - if (err < 0) - return false; - - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return i < ARRAY_SIZE(models); -} +#define MODEL_ALESIS_IO_BOTH 0x000001 static int check_dice_category(struct fw_unit *unit) { @@ -75,11 +50,6 @@ static int check_dice_category(struct fw_unit *unit) } } - if (vendor == OUI_FOCUSRITE || vendor == OUI_TCELECTRONIC) { - if (force_two_pcm_support(unit)) - return 0; - } - if (vendor == OUI_WEISS) category = WEISS_CATEGORY_ID; else if (vendor == OUI_LOUD) @@ -186,9 +156,6 @@ static void do_registration(struct work_struct *work) if (err < 0) return; - if (force_two_pcm_support(dice->unit)) - dice->force_two_pcms = true; - err = snd_dice_transaction_init(dice); if (err < 0) goto error; @@ -199,6 +166,10 @@ static void do_registration(struct work_struct *work) dice_card_strings(dice); + err = dice->detect_formats(dice); + if (err < 0) + goto error; + err = snd_dice_stream_init_duplex(dice); if (err < 0) goto error; @@ -239,14 +210,17 @@ error: "Sound card registration failed: %d\n", err); } -static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) +static int dice_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) { struct snd_dice *dice; int err; - err = check_dice_category(unit); - if (err < 0) - return -ENODEV; + if (!entry->driver_data) { + err = check_dice_category(unit); + if (err < 0) + return -ENODEV; + } /* Allocate this independent of sound card instance. */ dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL); @@ -256,6 +230,13 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, dice); + if (!entry->driver_data) { + dice->detect_formats = snd_dice_stream_detect_current_formats; + } else { + dice->detect_formats = + (snd_dice_detect_formats_t)entry->driver_data; + } + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); init_completion(&dice->clock_accepted); @@ -313,16 +294,97 @@ static void dice_bus_reset(struct fw_unit *unit) #define DICE_INTERFACE 0x000001 static const struct ieee1394_device_id dice_id_table[] = { + /* M-Audio Profire 2626 has a different value in version field. */ { - .match_flags = IEEE1394_MATCH_VERSION, - .version = DICE_INTERFACE, + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_MAUDIO, + .model_id = 0x000010, + .driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats, }, - /* M-Audio Profire 610/2626 has a different value in version field. */ + /* M-Audio Profire 610 has a different value in version field. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, - .vendor_id = 0x000d6c, - .specifier_id = 0x000d6c, + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_MAUDIO, + .model_id = 0x000011, + .driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats, + }, + /* TC Electronic Konnekt 24D. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000020, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Konnekt 8. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000021, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Studio Konnekt 48. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000022, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Konnekt Live. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000023, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Desktop Konnekt 6. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000024, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Impact Twin. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000027, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* TC Electronic Digital Konnekt x32. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000030, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, + /* Alesis iO14/iO26. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_ALESIS, + .model_id = MODEL_ALESIS_IO_BOTH, + .driver_data = (kernel_ulong_t)snd_dice_detect_alesis_formats, + }, + /* Mytek Stereo 192 DSD-DAC. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_MYTEK, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_mytek_formats, + }, + { + .match_flags = IEEE1394_MATCH_VERSION, + .version = DICE_INTERFACE, }, { } }; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index da00e75e09d4..83353a3559e8 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -63,6 +63,16 @@ */ #define MAX_STREAMS 2 +enum snd_dice_rate_mode { + SND_DICE_RATE_MODE_LOW = 0, + SND_DICE_RATE_MODE_MIDDLE, + SND_DICE_RATE_MODE_HIGH, + SND_DICE_RATE_MODE_COUNT, +}; + +struct snd_dice; +typedef int (*snd_dice_detect_formats_t)(struct snd_dice *dice); + struct snd_dice { struct snd_card *card; struct fw_unit *unit; @@ -80,6 +90,11 @@ struct snd_dice { unsigned int rsrv_offset; unsigned int clock_caps; + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int tx_midi_ports[MAX_STREAMS]; + unsigned int rx_midi_ports[MAX_STREAMS]; + snd_dice_detect_formats_t detect_formats; struct fw_address_handler notification_handler; int owner_generation; @@ -98,8 +113,6 @@ struct snd_dice { bool global_enabled; struct completion clock_accepted; unsigned int substreams_counter; - - bool force_two_pcms; }; enum snd_dice_addr_type { @@ -190,11 +203,14 @@ void snd_dice_transaction_destroy(struct snd_dice *dice); #define SND_DICE_RATES_COUNT 7 extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; +int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, + enum snd_dice_rate_mode *mode); int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate); void snd_dice_stream_stop_duplex(struct snd_dice *dice); int snd_dice_stream_init_duplex(struct snd_dice *dice); void snd_dice_stream_destroy_duplex(struct snd_dice *dice); void snd_dice_stream_update_duplex(struct snd_dice *dice); +int snd_dice_stream_detect_current_formats(struct snd_dice *dice); int snd_dice_stream_lock_try(struct snd_dice *dice); void snd_dice_stream_lock_release(struct snd_dice *dice); @@ -207,4 +223,9 @@ void snd_dice_create_proc(struct snd_dice *dice); int snd_dice_create_midi(struct snd_dice *dice); +int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice); +int snd_dice_detect_alesis_formats(struct snd_dice *dice); +int snd_dice_detect_extension_formats(struct snd_dice *dice); +int snd_dice_detect_mytek_formats(struct snd_dice *dice); + #endif diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c index a1d601f31165..6996d5a6ff5f 100644 --- a/sound/firewire/digi00x/digi00x-proc.c +++ b/sound/firewire/digi00x/digi00x-proc.c @@ -79,7 +79,7 @@ void snd_dg00x_proc_init(struct snd_dg00x *dg00x) if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireface/ff-proc.c b/sound/firewire/fireface/ff-proc.c index 69441d121f71..40ccbfd8ef89 100644 --- a/sound/firewire/fireface/ff-proc.c +++ b/sound/firewire/fireface/ff-proc.c @@ -52,7 +52,7 @@ void snd_ff_proc_init(struct snd_ff *ff) ff->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 12aa15df435d..ad7a0a32557d 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -147,7 +147,7 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) __le32 *reg; int i; - reg = kzalloc(sizeof(__le32) * 18, GFP_KERNEL); + reg = kcalloc(18, sizeof(__le32), GFP_KERNEL); if (reg == NULL) return -ENOMEM; diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index 9c21f31b8b21..779ecec5af62 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -219,7 +219,7 @@ void snd_efw_proc_init(struct snd_efw *efw) efw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 46092fa3ff9b..3919e186a30b 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -569,18 +569,20 @@ static int isight_create_mixer(struct isight *isight) return err; isight->gain_max = be32_to_cpu(value); - isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX; - isight->gain_tlv[1] = 2 * sizeof(unsigned int); + isight->gain_tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_MINMAX; + isight->gain_tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int); err = reg_read(isight, REG_GAIN_DB_START, &value); if (err < 0) return err; - isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100; + isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MIN] = + (s32)be32_to_cpu(value) * 100; err = reg_read(isight, REG_GAIN_DB_END, &value); if (err < 0) return err; - isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100; + isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX] = + (s32)be32_to_cpu(value) * 100; ctl = snd_ctl_new1(&gain_control, isight); if (ctl) diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index 4edc064999ed..ab6830a6d242 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -107,7 +107,7 @@ void snd_motu_proc_init(struct snd_motu *motu) motu->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 8ba4f9f262b8..27dac071bc73 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -103,7 +103,7 @@ void snd_oxfw_proc_init(struct snd_oxfw *oxfw) oxfw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 413ab6313bb6..1e5b2c802635 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -49,7 +49,6 @@ static bool detect_loud_models(struct fw_unit *unit) "Tapco LINK.firewire 4x6", "U.420"}; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, @@ -57,12 +56,7 @@ static bool detect_loud_models(struct fw_unit *unit) if (err < 0) return false; - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return (i < ARRAY_SIZE(models)); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int name_card(struct snd_oxfw *oxfw) diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index ea1506679c66..1ebf00c83409 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -27,7 +27,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit, void *p; int err; - b->packets = kmalloc(count * sizeof(*b->packets), GFP_KERNEL); + b->packets = kmalloc_array(count, sizeof(*b->packets), GFP_KERNEL); if (!b->packets) { err = -ENOMEM; goto error; diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c index bfd4a4c06914..fee3bf32a0da 100644 --- a/sound/firewire/tascam/tascam-proc.c +++ b/sound/firewire/tascam/tascam-proc.c @@ -78,7 +78,7 @@ void snd_tscm_proc_init(struct snd_tscm *tscm) tscm->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3129546398d0..2d90e11b3eaa 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -5,11 +5,12 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool +config SND_HDA_COMPONENT + bool + config SND_HDA_I915 bool - default y - depends on DRM_I915 - depends on SND_HDA_CORE + select SND_HDA_COMPONENT config SND_HDA_EXT_CORE tristate diff --git a/sound/hda/Makefile b/sound/hda/Makefile index e4e726f2ce98..2160202e2dc1 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -6,6 +6,7 @@ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) # for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_COMPONENT) += hdac_component.o snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 0daf31383084..9c37d9af3023 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -87,9 +87,10 @@ static const struct hdac_io_ops hdac_ext_default_io = { * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, +int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops) + const struct hdac_io_ops *io_ops, + const struct hdac_ext_bus_ops *ext_ops) { int ret; static int idx; @@ -98,15 +99,16 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, if (io_ops == NULL) io_ops = &hdac_ext_default_io; - ret = snd_hdac_bus_init(&ebus->bus, dev, ops, io_ops); + ret = snd_hdac_bus_init(bus, dev, ops, io_ops); if (ret < 0) return ret; - INIT_LIST_HEAD(&ebus->hlink_list); - ebus->idx = idx++; + bus->ext_ops = ext_ops; + INIT_LIST_HEAD(&bus->hlink_list); + bus->idx = idx++; - mutex_init(&ebus->lock); - ebus->cmd_dma_state = true; + mutex_init(&bus->lock); + bus->cmd_dma_state = true; return 0; } @@ -116,10 +118,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus * @ebus: the pointer to extended bus object */ -void snd_hdac_ext_bus_exit(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_exit(struct hdac_bus *bus) { - snd_hdac_bus_exit(&ebus->bus); - WARN_ON(!list_empty(&ebus->hlink_list)); + snd_hdac_bus_exit(bus); + WARN_ON(!list_empty(&bus->hlink_list)); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit); @@ -135,21 +137,15 @@ static void default_release(struct device *dev) * * Returns zero for success or a negative error code. */ -int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, + struct hdac_device *hdev) { - struct hdac_ext_device *edev; - struct hdac_device *hdev = NULL; - struct hdac_bus *bus = ebus_to_hbus(ebus); char name[15]; int ret; - edev = kzalloc(sizeof(*edev), GFP_KERNEL); - if (!edev) - return -ENOMEM; - hdev = &edev->hdev; - edev->ebus = ebus; + hdev->bus = bus; - snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); + snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr); ret = snd_hdac_device_init(hdev, bus, name, addr); if (ret < 0) { @@ -176,10 +172,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init); */ void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); - snd_hdac_device_exit(hdev); - kfree(edev); + kfree(hdev); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); @@ -188,14 +182,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); * * @ebus: HD-audio extended bus */ -void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus) { struct hdac_device *codec, *__codec; /* * we need to remove all the codec devices objects created in the * snd_hdac_ext_bus_device_init */ - list_for_each_entry_safe(codec, __codec, &ebus->bus.codec_list, list) { + list_for_each_entry_safe(codec, __codec, &bus->codec_list, list) { snd_hdac_device_unregister(codec); put_device(&codec->dev); } @@ -204,35 +198,31 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_remove); #define dev_to_hdac(dev) (container_of((dev), \ struct hdac_device, dev)) -static inline struct hdac_ext_driver *get_edrv(struct device *dev) +static inline struct hdac_driver *get_hdrv(struct device *dev) { struct hdac_driver *hdrv = drv_to_hdac_driver(dev->driver); - struct hdac_ext_driver *edrv = to_ehdac_driver(hdrv); - - return edrv; + return hdrv; } -static inline struct hdac_ext_device *get_edev(struct device *dev) +static inline struct hdac_device *get_hdev(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); - struct hdac_ext_device *edev = to_ehdac_device(hdev); - - return edev; + return hdev; } static int hda_ext_drv_probe(struct device *dev) { - return (get_edrv(dev))->probe(get_edev(dev)); + return (get_hdrv(dev))->probe(get_hdev(dev)); } static int hdac_ext_drv_remove(struct device *dev) { - return (get_edrv(dev))->remove(get_edev(dev)); + return (get_hdrv(dev))->remove(get_hdev(dev)); } static void hdac_ext_drv_shutdown(struct device *dev) { - return (get_edrv(dev))->shutdown(get_edev(dev)); + return (get_hdrv(dev))->shutdown(get_hdev(dev)); } /** @@ -240,20 +230,20 @@ static void hdac_ext_drv_shutdown(struct device *dev) * * @drv: ext hda driver structure */ -int snd_hda_ext_driver_register(struct hdac_ext_driver *drv) +int snd_hda_ext_driver_register(struct hdac_driver *drv) { - drv->hdac.type = HDA_DEV_ASOC; - drv->hdac.driver.bus = &snd_hda_bus_type; + drv->type = HDA_DEV_ASOC; + drv->driver.bus = &snd_hda_bus_type; /* we use default match */ if (drv->probe) - drv->hdac.driver.probe = hda_ext_drv_probe; + drv->driver.probe = hda_ext_drv_probe; if (drv->remove) - drv->hdac.driver.remove = hdac_ext_drv_remove; + drv->driver.remove = hdac_ext_drv_remove; if (drv->shutdown) - drv->hdac.driver.shutdown = hdac_ext_drv_shutdown; + drv->driver.shutdown = hdac_ext_drv_shutdown; - return driver_register(&drv->hdac.driver); + return driver_register(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); @@ -262,8 +252,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); * * @drv: ext hda driver structure */ -void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv) +void snd_hda_ext_driver_unregister(struct hdac_driver *drv) { - driver_unregister(&drv->hdac.driver); + driver_unregister(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_unregister); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 84f3b8168716..5bc4a1d587d4 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -39,9 +39,8 @@ * @ebus: HD-audio extended core bus * @enable: flag to turn on/off the capability */ -void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL"); @@ -60,9 +59,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); * @ebus: HD-audio extended core bus * @enable: flag to enable/disable interrupt */ -void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL\n"); @@ -89,12 +87,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); * in hlink_list of extended hdac bus * Note: this will be freed on bus exit by driver */ -int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus) { int idx; u32 link_count; struct hdac_ext_link *hlink; - struct hdac_bus *bus = &ebus->bus; link_count = readl(bus->mlcap + AZX_REG_ML_MLCD) + 1; @@ -114,7 +111,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) /* since link in On, update the ref */ hlink->ref_count = 1; - list_add_tail(&hlink->list, &ebus->hlink_list); + list_add_tail(&hlink->list, &bus->hlink_list); } return 0; @@ -127,12 +124,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_get_ml_capabilities); * @ebus: HD-audio ext core bus */ -void snd_hdac_link_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_link_free_all(struct hdac_bus *bus) { struct hdac_ext_link *l; - while (!list_empty(&ebus->hlink_list)) { - l = list_first_entry(&ebus->hlink_list, struct hdac_ext_link, list); + while (!list_empty(&bus->hlink_list)) { + l = list_first_entry(&bus->hlink_list, struct hdac_ext_link, list); list_del(&l->list); kfree(l); } @@ -144,7 +141,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); * @ebus: HD-audio extended core bus * @codec_name: codec name */ -struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, +struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, const char *codec_name) { int i; @@ -153,10 +150,10 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, if (sscanf(codec_name, "ehdaudio%dD%d", &bus_idx, &addr) != 2) return NULL; - if (ebus->idx != bus_idx) + if (bus->idx != bus_idx) return NULL; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { if (hlink->lsdiid & (0x1 << addr)) return hlink; @@ -219,12 +216,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down); * snd_hdac_ext_bus_link_power_up_all -power up all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); @@ -240,12 +237,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up_all); * snd_hdac_ext_bus_link_power_down_all -power down all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) @@ -256,39 +253,48 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); -int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 0 to 1, count will be 1 so power up this link * as well, also check the dma status and trigger that */ if (++link->ref_count == 1) { - if (!ebus->cmd_dma_state) { - snd_hdac_bus_init_cmd_io(&ebus->bus); - ebus->cmd_dma_state = true; + if (!bus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(bus); + bus->cmd_dma_state = true; } ret = snd_hdac_ext_bus_link_power_up(link); + + /* + * wait for 521usec for codec to report status + * HDA spec section 4.3 - Codec Discovery + */ + udelay(521); + bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); + snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); -int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; struct hdac_ext_link *hlink; bool link_up = false; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 1 to 0, count will be 0 @@ -301,7 +307,7 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, * now check if all links are off, if so turn off * cmd dma as well */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) { link_up = true; break; @@ -309,12 +315,12 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, } if (!link_up) { - snd_hdac_bus_stop_cmd_io(&ebus->bus); - ebus->cmd_dma_state = false; + snd_hdac_bus_stop_cmd_io(bus); + bus->cmd_dma_state = false; } } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c96d7a7a36af..1bd27576db98 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -25,7 +25,7 @@ /** * snd_hdac_ext_stream_init - initialize each stream (aka device) - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @idx: stream index number * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE) @@ -34,18 +34,16 @@ * initialize the stream, if ppcap is enabled then init those and then * invoke hdac stream initialization routine */ -void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_init(struct hdac_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag) { - struct hdac_bus *bus = &ebus->bus; - if (bus->ppcap) { stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + AZX_PPHC_INTERVAL * idx; stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE + - AZX_PPLC_MULTI * ebus->num_streams + + AZX_PPLC_MULTI * bus->num_streams + AZX_PPLC_INTERVAL * idx; } @@ -71,12 +69,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** * snd_hdac_ext_stream_init_all - create and initialize the stream objects * for an extended hda bus - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @start_idx: start index for streams * @num_stream: number of streams to initialize * @dir: direction of streams */ -int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, +int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir) { int stream_tag = 0; @@ -88,7 +86,7 @@ int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, if (!stream) return -ENOMEM; tag = ++stream_tag; - snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag); + snd_hdac_ext_stream_init(bus, stream, idx, dir, tag); idx++; } @@ -100,17 +98,16 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); /** * snd_hdac_stream_free_all - free hdac extended stream objects * - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_stream_free_all(struct hdac_bus *bus) { struct hdac_stream *s, *_s; struct hdac_ext_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); list_for_each_entry_safe(s, _s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); list_del(&s->list); kfree(stream); } @@ -119,15 +116,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); /** * snd_hdac_ext_stream_decouple - decouple the hdac stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @decouple: flag to decouple */ -void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *stream, bool decouple) { struct hdac_stream *hstream = &stream->hstream; - struct hdac_bus *bus = &ebus->bus; u32 val; int mask = AZX_PPCTL_PROCEN(hstream->index); @@ -251,19 +247,18 @@ void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); static struct hdac_ext_stream * -hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_link_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -277,34 +272,33 @@ hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, } if (!hstream->link_locked) { - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; } static struct hdac_ext_stream * -hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_host_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -313,17 +307,17 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, if (!stream->opened) { if (!hstream->decoupled) - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->hstream.opened = 1; res->hstream.running = 0; res->hstream.substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; @@ -331,7 +325,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, /** * snd_hdac_ext_stream_assign - assign a stream for the PCM - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @substream: PCM substream to assign * @type: type of stream (coupled, host or link stream) * @@ -346,27 +340,26 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, * the same stream object when it's used beforehand. when a stream is * decoupled, it becomes a host stream and link stream. */ -struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *ebus, +struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type) { struct hdac_ext_stream *hstream = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: - stream = snd_hdac_stream_assign(hbus, substream); + stream = snd_hdac_stream_assign(bus, substream); if (stream) hstream = container_of(stream, struct hdac_ext_stream, hstream); return hstream; case HDAC_EXT_STREAM_TYPE_HOST: - return hdac_ext_host_stream_assign(ebus, substream); + return hdac_ext_host_stream_assign(bus, substream); case HDAC_EXT_STREAM_TYPE_LINK: - return hdac_ext_link_stream_assign(ebus, substream); + return hdac_ext_link_stream_assign(bus, substream); default: return NULL; @@ -384,7 +377,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_assign); void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) { struct hdac_bus *bus = stream->hstream.bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: @@ -393,13 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) case HDAC_EXT_STREAM_TYPE_HOST: if (stream->decoupled && !stream->link_locked) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); snd_hdac_stream_release(&stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: if (stream->decoupled && !stream->hstream.opened) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); spin_lock_irq(&bus->reg_lock); stream->link_locked = 0; stream->link_substream = NULL; @@ -415,16 +407,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_release); /** * snd_hdac_ext_stream_spbcap_enable - enable SPIB for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable SPIB * @index: stream index for which SPIB need to be enabled */ -void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -446,14 +437,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable); /** * snd_hdac_ext_stream_set_spib - sets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: spib value to set */ -int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -468,15 +458,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_spib); /** * snd_hdac_ext_stream_get_spbmaxfifo - gets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * * Return maxfifo for the stream */ -int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_ext_stream *stream) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -490,11 +479,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo); /** * snd_hdac_ext_stop_streams - stop all stream if running - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_ext_stop_streams(struct hdac_ext_bus *ebus) +void snd_hdac_ext_stop_streams(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); struct hdac_stream *stream; if (bus->chip_init) { @@ -507,16 +495,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stop_streams); /** * snd_hdac_ext_stream_drsm_enable - enable DMA resume for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable DRSM * @index: stream index for which DRSM need to be enabled */ -void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -538,14 +525,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable); /** * snd_hdac_ext_stream_set_dpibr - sets the dpibr value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: dpib value to set */ -int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -560,7 +546,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr); /** * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: lpib value to set */ diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c new file mode 100644 index 000000000000..6e46a9c73aed --- /dev/null +++ b/sound/hda/hdac_component.c @@ -0,0 +1,335 @@ +// SPDX-License-Identifier: GPL-2.0 +// hdac_component.c - routines for sync between HD-A core and DRM driver + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_component.h> +#include <sound/hda_register.h> + +static void hdac_acomp_release(struct device *dev, void *res) +{ +} + +static struct drm_audio_component *hdac_get_acomp(struct device *dev) +{ + return devres_find(dev, hdac_acomp_release, NULL, NULL); +} + +/** + * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup + * @bus: HDA core bus + * @enable: enable or disable the wakeup + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function should be called during the chip reset, also called at + * resume for updating STATESTS register read. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) + return 0; + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); + +/** + * snd_hdac_display_power - Power up / down the power refcount + * @bus: HDA core bus + * @enable: power up or down + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function manages a refcount and calls the get_power() and + * put_power() ops accordingly, toggling the codec wakeup, too. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->drm_power_refcount++) { + if (acomp->ops->get_power) + acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } + } else { + WARN_ON(!bus->drm_power_refcount); + if (!--bus->drm_power_refcount) + if (acomp->ops->put_power) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_display_power); + +/** + * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @rate: the sample rate to set + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets N/CTS value based on the given sample rate. + * Returns zero for success, or a negative error code. + */ +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) + return -ENODEV; + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); +} +EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); + +/** + * snd_hdac_acomp_get_eld - Get the audio state and ELD via component + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @audio_enabled: the pointer to store the current audio state + * @buffer: the buffer pointer to store ELD bytes + * @max_bytes: the max bytes to be stored on @buffer + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function queries the current state of the audio on the given + * digital port and fetches the ELD bytes onto the given buffer. + * It returns the number of bytes for the total ELD data, zero for + * invalid ELD, or a negative error code. + * + * The return size is the total bytes required for the whole ELD bytes, + * thus it may be over @max_bytes. If it's over @max_bytes, it implies + * that only a part of ELD bytes have been fetched. + */ +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, + bool *audio_enabled, char *buffer, int max_bytes) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->get_eld) + return -ENODEV; + + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, + buffer, max_bytes); +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); + +static int hdac_component_master_bind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + int ret; + + if (WARN_ON(!acomp)) + return -EINVAL; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops))) { + ret = -EINVAL; + goto out_unbind; + } + + /* pin the module to avoid dynamic unbinding, but only if given */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + if (acomp->audio_ops && acomp->audio_ops->master_bind) { + ret = acomp->audio_ops->master_bind(dev, acomp); + if (ret < 0) + goto module_put; + } + + return 0; + + module_put: + module_put(acomp->ops->owner); +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hdac_component_master_unbind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + + if (acomp->audio_ops && acomp->audio_ops->master_unbind) + acomp->audio_ops->master_unbind(dev, acomp); + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hdac_component_master_ops = { + .bind = hdac_component_master_bind, + .unbind = hdac_component_master_unbind, +}; + +/** + * snd_hdac_acomp_register_notifier - Register audio component ops + * @bus: HDA core bus + * @aops: audio component ops + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets the given ops to be called by the graphics driver. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops) +{ + if (!bus->audio_component) + return -ENODEV; + + bus->audio_component->audio_ops = aops; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_register_notifier); + +/** + * snd_hdac_acomp_init - Initialize audio component + * @bus: HDA core bus + * @match_master: match function for finding components + * @extra_size: Extra bytes to allocate + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function initializes and sets up the audio component to communicate + * with graphics driver. + * + * Unlike snd_hdac_i915_init(), this function doesn't synchronize with the + * binding with the DRM component. Each caller needs to sync via master_bind + * audio_ops. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct drm_audio_component *acomp; + int ret; + + if (WARN_ON(hdac_get_acomp(dev))) + return -EBUSY; + + acomp = devres_alloc(hdac_acomp_release, sizeof(*acomp) + extra_size, + GFP_KERNEL); + if (!acomp) + return -ENOMEM; + acomp->audio_ops = aops; + bus->audio_component = acomp; + devres_add(dev, acomp); + + component_match_add(dev, &match, match_master, bus); + ret = component_master_add_with_match(dev, &hdac_component_master_ops, + match); + if (ret < 0) + goto out_err; + + return 0; + +out_err: + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + dev_info(dev, "failed to add audio component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_init); + +/** + * snd_hdac_acomp_exit - Finalize audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function releases the audio component that has been used. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp) + return 0; + + WARN_ON(bus->drm_power_refcount); + if (bus->drm_power_refcount > 0 && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hdac_component_master_ops); + + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_exit); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index cbe818eda336..8f2aa8bc1185 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -15,89 +15,11 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_register.h> -static struct i915_audio_component *hdac_acomp; - -/** - * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup - * @bus: HDA core bus - * @enable: enable or disable the wakeup - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function should be called during the chip reset, also called at - * resume for updating STATESTS register read. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) -{ - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(bus->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(bus->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); - -/** - * snd_hdac_display_power - Power up / down the power refcount - * @bus: HDA core bus - * @enable: power up or down - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function manages a refcount and calls the i915 get_power() and - * put_power() ops accordingly, toggling the codec wakeup, too. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_display_power(struct hdac_bus *bus, bool enable) -{ - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - dev_dbg(bus->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!bus->i915_power_refcount++) { - acomp->ops->get_power(acomp->dev); - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); - } - } else { - WARN_ON(!bus->i915_power_refcount); - if (!--bus->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_display_power); - #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -119,7 +41,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_display_power); */ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; struct pci_dev *pci = to_pci_dev(bus->dev); int cdclk_freq; unsigned int bclk_m, bclk_n; @@ -158,181 +80,11 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/* There is a fixed mapping between audio pin node and display port. - * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: - * Pin Widget 5 - PORT B (port = 1 in i915 driver) - * Pin Widget 6 - PORT C (port = 2 in i915 driver) - * Pin Widget 7 - PORT D (port = 3 in i915 driver) - * - * on VLV, ILK: - * Pin Widget 4 - PORT B (port = 1 in i915 driver) - * Pin Widget 5 - PORT C (port = 2 in i915 driver) - * Pin Widget 6 - PORT D (port = 3 in i915 driver) - */ -static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) +static int i915_component_master_match(struct device *dev, void *data) { - int base_nid; - - switch (codec->vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - base_nid = 3; - break; - default: - base_nid = 4; - break; - } - - if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) - return -1; - return pin_nid - base_nid; -} - -/** - * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @rate: the sample rate to set - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets N/CTS value based on the given sample rate. - * Returns zero for success, or a negative error code. - */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, - int dev_id, int rate) -{ - struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) - return -ENODEV; - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - pipe = dev_id; - return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); -} -EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); - -/** - * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @audio_enabled: the pointer to store the current audio state - * @buffer: the buffer pointer to store ELD bytes - * @max_bytes: the max bytes to be stored on @buffer - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function queries the current state of the audio on the given - * digital port and fetches the ELD bytes onto the given buffer. - * It returns the number of bytes for the total ELD data, zero for - * invalid ELD, or a negative error code. - * - * The return size is the total bytes required for the whole ELD bytes, - * thus it may be over @max_bytes. If it's over @max_bytes, it implies - * that only a part of ELD bytes have been fetched. - */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, - bool *audio_enabled, char *buffer, int max_bytes) -{ - struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->get_eld) - return -ENODEV; - - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - - pipe = dev_id; - return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, - buffer, max_bytes); -} -EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); - -static int hdac_component_master_bind(struct device *dev) -{ - struct i915_audio_component *acomp = hdac_acomp; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hdac_component_master_unbind(struct device *dev) -{ - struct i915_audio_component *acomp = hdac_acomp; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hdac_component_master_ops = { - .bind = hdac_component_master_bind, - .unbind = hdac_component_master_unbind, -}; - -static int hdac_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ return !strcmp(dev->driver->name, "i915"); } -/** - * snd_hdac_i915_register_notifier - Register i915 audio component ops - * @aops: i915 audio component ops - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets the given ops to be called by the i915 graphics driver. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) -{ - if (!hdac_acomp) - return -ENODEV; - - hdac_acomp->audio_ops = aops; - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); - /* check whether intel graphics is present */ static bool i915_gfx_present(void) { @@ -359,83 +111,26 @@ static bool i915_gfx_present(void) */ int snd_hdac_i915_init(struct hdac_bus *bus) { - struct component_match *match = NULL; - struct device *dev = bus->dev; - struct i915_audio_component *acomp; - int ret; - - if (WARN_ON(hdac_acomp)) - return -EBUSY; + struct drm_audio_component *acomp; + int err; if (!i915_gfx_present()) return -ENODEV; - acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); + err = snd_hdac_acomp_init(bus, NULL, + i915_component_master_match, + sizeof(struct i915_audio_component) - sizeof(*acomp)); + if (err < 0) + return err; + acomp = bus->audio_component; if (!acomp) - return -ENOMEM; - bus->audio_component = acomp; - hdac_acomp = acomp; - - component_match_add(dev, &match, hdac_component_master_match, bus); - ret = component_master_add_with_match(dev, &hdac_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - + return -ENODEV; + if (!acomp->ops) + request_module("i915"); if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; + snd_hdac_acomp_exit(bus); + return -ENODEV; } - dev_dbg(dev, "bound to i915 component master\n"); - return 0; -out_master_del: - component_master_del(dev, &hdac_component_master_ops); -out_err: - kfree(acomp); - bus->audio_component = NULL; - hdac_acomp = NULL; - dev_info(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; } EXPORT_SYMBOL_GPL(snd_hdac_i915_init); - -/** - * snd_hdac_i915_exit - Finalize i915 audio component - * @bus: HDA core bus - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function releases the i915 audio component that has been used. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_exit(struct hdac_bus *bus) -{ - struct device *dev = bus->dev; - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp) - return 0; - - WARN_ON(bus->i915_power_refcount); - if (bus->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hdac_component_master_ops); - - kfree(acomp); - bus->audio_component = NULL; - hdac_acomp = NULL; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_exit); diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 47a358fab132..419e285e0226 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -65,10 +65,10 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) { struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); + const unsigned int *v; int i; - for (i = 0; i < codec->vendor_verbs.used; i++) { - unsigned int *v = snd_array_elem(&codec->vendor_verbs, i); + snd_array_for_each(&codec->vendor_verbs, i, v) { if (verb == *v) return true; } diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index cb54d9c0a77f..43b35a873d78 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -20,7 +20,8 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" - depends on ISA && ISA_DMA_API + depends on ISA || COMPILE_TEST + depends on ISA_DMA_API default y help Support for sound devices connected via the ISA bus. diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index d09e456107ad..de6ef1b1cf0e 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -192,7 +192,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip) } /* find index of an item in "-1"-ended array */ -int array_find(int array[], int item) +static int array_find(int array[], int item) { int i; @@ -203,7 +203,7 @@ int array_find(int array[], int item) return -1; } /* the same for long */ -int array_find_l(long array[], long item) +static int array_find_l(long array[], long item) { int i; diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 45e561c425bf..6c584d9b6c42 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -757,9 +757,9 @@ static int snd_msnd_pinnacle_cfg_reset(int cfg) static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -801,22 +801,22 @@ MODULE_LICENSE("GPL"); MODULE_FIRMWARE(INITCODEFILE); MODULE_FIRMWARE(PERMCODEFILE); -module_param_hw_array(io, long, ioport, NULL, S_IRUGO); +module_param_hw_array(io, long, ioport, NULL, 0444); MODULE_PARM_DESC(io, "IO port #"); -module_param_hw_array(irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(mem, long, iomem, NULL, S_IRUGO); -module_param_array(write_ndelay, int, NULL, S_IRUGO); -module_param(calibrate_signal, int, S_IRUGO); +module_param_hw_array(irq, int, irq, NULL, 0444); +module_param_hw_array(mem, long, iomem, NULL, 0444); +module_param_array(write_ndelay, int, NULL, 0444); +module_param(calibrate_signal, int, 0444); #ifndef MSND_CLASSIC -module_param_array(digital, int, NULL, S_IRUGO); -module_param_hw_array(cfg, long, ioport, NULL, S_IRUGO); -module_param_array(reset, int, 0, S_IRUGO); -module_param_hw_array(mpu_io, long, ioport, NULL, S_IRUGO); -module_param_hw_array(mpu_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(ide_io0, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_io1, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(joystick_io, long, ioport, NULL, S_IRUGO); +module_param_array(digital, int, NULL, 0444); +module_param_hw_array(cfg, long, ioport, NULL, 0444); +module_param_array(reset, int, 0, 0444); +module_param_hw_array(mpu_io, long, ioport, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); +module_param_hw_array(ide_io0, long, ioport, NULL, 0444); +module_param_hw_array(ide_io1, long, ioport, NULL, 0444); +module_param_hw_array(ide_irq, int, irq, NULL, 0444); +module_param_hw_array(joystick_io, long, ioport, NULL, 0444); #endif diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c09d9b914efe..a985e9183be9 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -592,7 +592,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) *vport = devm_ioport_map(devptr, port[dev], 0x10); if (*vport == NULL) { snd_printk(KERN_ERR PFX - "I/O port cannot be iomaped.\n"); + "I/O port cannot be iomapped.\n"); err = -EBUSY; goto err_unmap1; } @@ -607,7 +607,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); if (!vmss_port) { snd_printk(KERN_ERR PFX - "MSS port I/O cannot be iomaped.\n"); + "MSS port I/O cannot be iomapped.\n"); err = -EBUSY; goto err_unmap2; } diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 8c0f8a9ee0ba..fc9bcd47d6a4 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -420,7 +420,7 @@ static int sq_allocate_buffers(struct sound_queue *sq, int num, int size) return 0; sq->numBufs = num; sq->bufSize = size; - sq->buffers = kmalloc (num * sizeof(char *), GFP_KERNEL); + sq->buffers = kmalloc_array (num, sizeof(char *), GFP_KERNEL); if (!sq->buffers) return -ENOMEM; for (i = 0; i < num; i++) { diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index d9f3fdb777e4..4105d9f653d9 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -175,7 +175,7 @@ config SND_BT87X help If you want to record audio from TV cards based on Brooktree Bt878/Bt879 chips, say Y here and read - <file:Documentation/sound/alsa/Bt87x.txt>. + <file:Documentation/sound/cards/bt87x.rst>. To compile this driver as a module, choose M here: the module will be called snd-bt87x. @@ -210,7 +210,7 @@ config SND_CMIPCI help If you want to use soundcards based on C-Media CMI8338, CMI8738, CMI8768 or CMI8770 chips, say Y here and read - <file:Documentation/sound/alsa/CMIPCI.txt>. + <file:Documentation/sound/cards/cmipci.rst>. To compile this driver as a module, choose M here: the module will be called snd-cmipci. @@ -472,8 +472,8 @@ config SND_EMU10K1 Audigy and E-mu APS (partially supported) soundcards. The confusing multitude of mixer controls is documented in - <file:Documentation/sound/alsa/SB-Live-mixer.txt> and - <file:Documentation/sound/alsa/Audigy-mixer.txt>. + <file:Documentation/sound/cards/sb-live-mixer.rst> and + <file:Documentation/sound/cards/audigy-mixer.rst>. To compile this driver as a module, choose M here: the module will be called snd-emu10k1. @@ -735,7 +735,7 @@ config SND_MIXART select SND_PCM help If you want to use Digigram miXart soundcards, say Y here and - read <file:Documentation/sound/alsa/MIXART.txt>. + read <file:Documentation/sound/cards/mixart.rst>. To compile this driver as a module, choose M here: the module will be called snd-mixart. diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 6320bf084e47..e120a11c69e8 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -448,7 +448,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { @@ -474,7 +474,7 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) sprintf(name, "codec97#%d", bus->num); if ((entry = snd_info_create_card_entry(bus->card, name, bus->card->proc_root)) != NULL) { - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04eeada..d9c54c08e2db 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -258,7 +258,7 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip) while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) && --retry) - mdelay(1); + usleep_range(1000, 2000); if (!retry) { dev_err(chip->card->dev, "[%s] Link is not ready.\n", __func__); @@ -872,7 +872,7 @@ snd_ad1889_init(struct snd_ad1889 *chip) ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */ ad1889_readw(chip, AD_DS_CCS); /* flush posted write */ - mdelay(10); + usleep_range(10000, 11000); /* enable Master and Target abort interrupts */ ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE); diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 720361455c60..64e0961f93ba 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -69,27 +69,27 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static bool enable_hpi_hwdep = 1; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ALSA ID string for AudioScience soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "ALSA enable AudioScience soundcard."); -module_param(enable_hpi_hwdep, bool, S_IRUGO|S_IWUSR); +module_param(enable_hpi_hwdep, bool, 0644); MODULE_PARM_DESC(enable_hpi_hwdep, "ALSA enable HPI hwdep for AudioScience soundcard "); /* identify driver */ #ifdef KERNEL_ALSA_BUILD static char *build_info = "Built using headers from kernel source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built using headers from kernel source"); #else static char *build_info = "Built within ALSA source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built within ALSA source"); #endif diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..7d049569012c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -46,14 +46,14 @@ MODULE_FIRMWARE("asihpi/dsp8900.bin"); #endif static int prealloc_stream_buf; -module_param(prealloc_stream_buf, int, S_IRUGO); +module_param(prealloc_stream_buf, int, 0444); MODULE_PARM_DESC(prealloc_stream_buf, "Preallocate size for per-adapter stream buffer"); /* Allow the debug level to be changed after module load. E.g. echo 2 > /sys/module/asihpi/parameters/hpiDebugLevel */ -module_param(hpi_debug_level, int, S_IRUGO | S_IWUSR); +module_param(hpi_debug_level, int, 0644); MODULE_PARM_DESC(hpi_debug_level, "debug verbosity 0..5"); /* List of adapters found */ diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 9b2b8b38122f..a2c85cc37972 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,7 +431,7 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); @@ -440,12 +440,12 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { entry->c.text.write = snd_ca0106_proc_i2c_write; entry->private_data = emu; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 26a657870664..452cc79b44af 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1139,7 +1139,7 @@ static int save_mixer_state(struct cmipci *cm) struct snd_ctl_elem_value *val; unsigned int i; - val = kmalloc(sizeof(*val), GFP_ATOMIC); + val = kmalloc(sizeof(*val), GFP_KERNEL); if (!val) return -ENOMEM; for (i = 0; i < CM_SAVED_MIXERS; i++) { diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 655fbea1692c..4910d3f46d4b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -58,7 +58,7 @@ MODULE_PARM_DESC(id, "ID string for the CS46xx soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable CS46xx soundcard."); module_param_array(external_amp, bool, NULL, 0444); -MODULE_PARM_DESC(external_amp, "Force to enable external amplifer."); +MODULE_PARM_DESC(external_amp, "Force to enable external amplifier."); module_param_array(thinkpad, bool, NULL, 0444); MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..146e1a3498c7 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -460,7 +460,7 @@ static int load_firmware(struct snd_cs46xx *chip, entry->size = le32_to_cpu(fwdat[fwlen++]); if (fwlen + entry->size > fwsize) goto error_inval; - entry->data = kmalloc(entry->size * 4, GFP_KERNEL); + entry->data = kmalloc_array(entry->size, 4, GFP_KERNEL); if (!entry->data) goto error; memcpy_le32(entry->data, &fwdat[fwlen], entry->size * 4); @@ -2849,7 +2849,7 @@ static int snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46xx *chip) entry->private_data = chip; entry->c.ops = &snd_cs46xx_proc_io_ops; entry->size = region->size; - entry->mode = S_IFREG | S_IRUSR; + entry->mode = S_IFREG | 0400; } } #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -4036,8 +4036,9 @@ int snd_cs46xx_create(struct snd_card *card, snd_cs46xx_proc_init(card, chip); #ifdef CONFIG_PM_SLEEP - chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * - ARRAY_SIZE(saved_regs), GFP_KERNEL); + chip->saved_regs = kmalloc_array(ARRAY_SIZE(saved_regs), + sizeof(*chip->saved_regs), + GFP_KERNEL); if (!chip->saved_regs) { snd_cs46xx_free(chip); return -ENOMEM; diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index aa61615288ff..598d140bb7cb 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -240,10 +240,13 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) return NULL; /* better to use vmalloc for this big table */ - ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) * - DSP_MAX_SYMBOLS); + ins->symbol_table.symbols = + vmalloc(array_size(DSP_MAX_SYMBOLS, + sizeof(struct dsp_symbol_entry))); ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL); - ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL); + ins->modules = kmalloc_array(DSP_MAX_MODULES, + sizeof(struct dsp_module_desc), + GFP_KERNEL); if (!ins->symbol_table.symbols || !ins->code.data || !ins->modules) { cs46xx_dsp_spos_destroy(chip); goto error; @@ -798,7 +801,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -814,7 +817,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_symbols", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -826,7 +829,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -838,7 +841,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "parameter", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -850,7 +853,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "sample", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -862,7 +865,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "task_tree", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -874,7 +877,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "scb_info", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..abb01ce66983 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -271,7 +271,7 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = scb_info; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 908658a00377..2ada8444abd9 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -275,7 +275,7 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) /* Get AMIXER resource */ n_amixer = (n_amixer < 2) ? 2 : n_amixer; - apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + apcm->amixers = kcalloc(n_amixer, sizeof(void *), GFP_KERNEL); if (!apcm->amixers) { err = -ENOMEM; goto error1; @@ -543,18 +543,18 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) } if (n_srcc) { - apcm->srccs = kzalloc(sizeof(void *)*n_srcc, GFP_KERNEL); + apcm->srccs = kcalloc(n_srcc, sizeof(void *), GFP_KERNEL); if (!apcm->srccs) return -ENOMEM; } if (n_amixer) { - apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + apcm->amixers = kcalloc(n_amixer, sizeof(void *), GFP_KERNEL); if (!apcm->amixers) { err = -ENOMEM; goto error1; } } - apcm->srcimps = kzalloc(sizeof(void *)*n_srcimp, GFP_KERNEL); + apcm->srcimps = kcalloc(n_srcimp, sizeof(void *), GFP_KERNEL); if (!apcm->srcimps) { err = -ENOMEM; goto error1; @@ -819,7 +819,7 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, /* Get AMIXER resource */ n_amixer = (n_amixer < 2) ? 2 : n_amixer; - apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + apcm->amixers = kcalloc(n_amixer, sizeof(void *), GFP_KERNEL); if (!apcm->amixers) { err = -ENOMEM; goto error1; @@ -1378,19 +1378,19 @@ static int atc_get_resources(struct ct_atc *atc) num_daios = ((atc->model == CTSB1270) ? 8 : 7); num_srcs = ((atc->model == CTSB1270) ? 6 : 4); - atc->daios = kzalloc(sizeof(void *)*num_daios, GFP_KERNEL); + atc->daios = kcalloc(num_daios, sizeof(void *), GFP_KERNEL); if (!atc->daios) return -ENOMEM; - atc->srcs = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL); + atc->srcs = kcalloc(num_srcs, sizeof(void *), GFP_KERNEL); if (!atc->srcs) return -ENOMEM; - atc->srcimps = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL); + atc->srcimps = kcalloc(num_srcs, sizeof(void *), GFP_KERNEL); if (!atc->srcimps) return -ENOMEM; - atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); + atc->pcm = kcalloc(2 * 4, sizeof(void *), GFP_KERNEL); if (!atc->pcm) return -ENOMEM; diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 7f089cb433e1..f35a7341e446 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -398,7 +398,8 @@ static int dao_rsc_init(struct dao *dao, if (err) return err; - dao->imappers = kzalloc(sizeof(void *)*desc->msr*2, GFP_KERNEL); + dao->imappers = kzalloc(array3_size(sizeof(void *), desc->msr, 2), + GFP_KERNEL); if (!dao->imappers) { err = -ENOMEM; goto error1; diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 4f4a2a5dedb8..db710d0a609f 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -910,13 +910,14 @@ static int ct_mixer_get_mem(struct ct_mixer **rmixer) if (!mixer) return -ENOMEM; - mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM), + mixer->amixers = kcalloc(NUM_CT_AMIXERS * CHN_NUM, sizeof(void *), GFP_KERNEL); if (!mixer->amixers) { err = -ENOMEM; goto error1; } - mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL); + mixer->sums = kcalloc(NUM_CT_SUMS * CHN_NUM, sizeof(void *), + GFP_KERNEL); if (!mixer->sums) { err = -ENOMEM; goto error2; diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index bb4c9c3c89ae..a4fc10723fc6 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -679,7 +679,7 @@ static int srcimp_rsc_init(struct srcimp *srcimp, return err; /* Reserve memory for imapper nodes */ - srcimp->imappers = kzalloc(sizeof(struct imapper)*desc->msr, + srcimp->imappers = kcalloc(desc->msr, sizeof(struct imapper), GFP_KERNEL); if (!srcimp->imappers) { err = -ENOMEM; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 08e874e9a7f6..2099e9ce441a 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -17,7 +17,7 @@ static bool use_system_timer; MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); -module_param(use_system_timer, bool, S_IRUGO); +module_param(use_system_timer, bool, 0444); struct ct_timer_ops { void (*init)(struct ct_timer_instance *); diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index f2f32779de98..b2874220be1b 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -26,9 +26,9 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; static unsigned int multiple = 2; MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)"); -module_param(reference_rate, uint, S_IRUGO); +module_param(reference_rate, uint, 0444); MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)"); -module_param(multiple, uint, S_IRUGO); +module_param(multiple, uint, 0444); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0935a5c8741f..358ef7dcf410 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -59,7 +59,7 @@ static int get_firmware(const struct firmware **fw_entry, dev_dbg(chip->card->dev, "firmware requested: %s\n", card_fw[fw_index].data); snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); - err = request_firmware(fw_entry, name, pci_device(chip)); + err = request_firmware(fw_entry, name, &chip->pci->dev); if (err < 0) dev_err(chip->card->dev, "get_firmware(): Firmware not available (%d)\n", err); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 152ce158583c..44b390a667d5 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -559,10 +559,4 @@ static inline int monitor_index(const struct echoaudio *chip, int out, int in) return out * num_busses_in(chip) + in; } - -#ifndef pci_device -#define pci_device(chip) (&chip->pci->dev) -#endif - - #endif /* _ECHOAUDIO_H_ */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 18267de3a269..61f85ff91cd9 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1941,9 +1941,10 @@ int snd_emu10k1_create(struct snd_card *card, (unsigned long)emu->ptb_pages.addr, (unsigned long)(emu->ptb_pages.addr + emu->ptb_pages.bytes)); - emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); - emu->page_addr_table = vmalloc(emu->max_cache_pages * - sizeof(unsigned long)); + emu->page_ptr_table = vmalloc(array_size(sizeof(void *), + emu->max_cache_pages)); + emu->page_addr_table = vmalloc(array_size(sizeof(unsigned long), + emu->max_cache_pages)); if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) { err = -ENOMEM; goto error; @@ -2099,7 +2100,7 @@ static int alloc_pm_buffer(struct snd_emu10k1 *emu) size = ARRAY_SIZE(saved_regs); if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); - emu->saved_ptr = vmalloc(4 * NUM_G * size); + emu->saved_ptr = vmalloc(array3_size(4, NUM_G, size)); if (!emu->saved_ptr) return -ENOMEM; if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2c2b12a06177..611589cbdad6 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1070,7 +1070,7 @@ static int snd_emu10k1x_proc_init(struct emu10k1x *emu) if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = emu; } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index a2b56b188be4..de2ecbe95d6c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -170,7 +170,7 @@ static char *audigy_outs[32] = { /* 0x0f */ "Rear Right", /* 0x10 */ "AC97 Front Left", /* 0x11 */ "AC97 Front Right", - /* 0x12 */ "ADC Caputre Left", + /* 0x12 */ "ADC Capture Left", /* 0x13 */ "ADC Capture Right", /* 0x14 */ NULL, /* 0x15 */ NULL, @@ -421,14 +421,10 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu, snd_fx8010_irq_handler_t *handler, unsigned char gpr_running, void *private_data, - struct snd_emu10k1_fx8010_irq **r_irq) + struct snd_emu10k1_fx8010_irq *irq) { - struct snd_emu10k1_fx8010_irq *irq; unsigned long flags; - irq = kmalloc(sizeof(*irq), GFP_ATOMIC); - if (irq == NULL) - return -ENOMEM; irq->handler = handler; irq->gpr_running = gpr_running; irq->private_data = private_data; @@ -443,8 +439,6 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu, emu->fx8010.irq_handlers = irq; } spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags); - if (r_irq) - *r_irq = irq; return 0; } @@ -468,7 +462,6 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu, tmp->next = tmp->next->next; } spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags); - kfree(irq); return 0; } @@ -2690,16 +2683,16 @@ int snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu) int len; len = emu->audigy ? 0x200 : 0x100; - emu->saved_gpr = kmalloc(len * 4, GFP_KERNEL); + emu->saved_gpr = kmalloc_array(len, 4, GFP_KERNEL); if (! emu->saved_gpr) return -ENOMEM; len = emu->audigy ? 0x100 : 0xa0; - emu->tram_val_saved = kmalloc(len * 4, GFP_KERNEL); - emu->tram_addr_saved = kmalloc(len * 4, GFP_KERNEL); + emu->tram_val_saved = kmalloc_array(len, 4, GFP_KERNEL); + emu->tram_addr_saved = kmalloc_array(len, 4, GFP_KERNEL); if (! emu->tram_val_saved || ! emu->tram_addr_saved) return -ENOMEM; len = emu->audigy ? 2 * 1024 : 2 * 512; - emu->saved_icode = vmalloc(len * 4); + emu->saved_icode = vmalloc(array_size(len, 4)); if (! emu->saved_icode) return -ENOMEM; return 0; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cefe613ef7b7..69f9b100bd24 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1724,7 +1724,7 @@ static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substre case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - snd_emu10k1_fx8010_unregister_irq_handler(emu, pcm->irq); pcm->irq = NULL; + snd_emu10k1_fx8010_unregister_irq_handler(emu, &pcm->irq); snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0); pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size); pcm->tram_shift = 0; @@ -1858,7 +1858,9 @@ int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device) if (!kctl) return -ENOMEM; kctl->id.device = device; - snd_ctl_add(emu->card, kctl); + err = snd_ctl_add(emu->card, kctl); + if (err < 0) + return err; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024); diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bde0d1954f56..b57008031792 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -135,7 +135,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, /* 15 */ "Rear Right", /* 16 */ "AC97 Front Left", /* 17 */ "AC97 Front Right", - /* 18 */ "ADC Caputre Left", + /* 18 */ "ADC Capture Left", /* 19 */ "ADC Capture Right", /* 20 */ "???", /* 21 */ "???", @@ -574,32 +574,32 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } #endif @@ -621,35 +621,35 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 5865f3b90b34..dbc7d8d0e1c4 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,13 +248,13 @@ __found_pages: static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) { if (addr & ~emu->dma_mask) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "max memory size is 0x%lx (addr = 0x%lx)!!\n", emu->dma_mask, (unsigned long)addr); return 0; } if (addr & (EMUPAGESIZE-1)) { - dev_err(emu->card->dev, "page is not aligned\n"); + dev_err_ratelimited(emu->card->dev, "page is not aligned\n"); return 0; } return 1; @@ -345,7 +345,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst else addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); return NULL; diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index a30da78a95b7..4948b95f6665 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -874,7 +874,7 @@ int snd_p16v_mixer(struct snd_emu10k1 *emu) int snd_p16v_alloc_pm_buffer(struct snd_emu10k1 *emu) { - emu->p16v_saved = vmalloc(NUM_CHS * 4 * 0x80); + emu->p16v_saved = vmalloc(array_size(NUM_CHS * 4, 0x80)); if (! emu->p16v_saved) return -ENOMEM; return 0; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 73a67bc3586b..e3fb9c61017c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1068,11 +1068,19 @@ static int snd_fm801_mixer(struct fm801 *chip) if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97_sec)) < 0) return err; } - for (i = 0; i < FM801_CONTROLS; i++) - snd_ctl_add(chip->card, snd_ctl_new1(&snd_fm801_controls[i], chip)); + for (i = 0; i < FM801_CONTROLS; i++) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_fm801_controls[i], chip)); + if (err < 0) + return err; + } if (chip->multichannel) { - for (i = 0; i < FM801_CONTROLS_MULTI; i++) - snd_ctl_add(chip->card, snd_ctl_new1(&snd_fm801_controls_multi[i], chip)); + for (i = 0; i < FM801_CONTROLS_MULTI; i++) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_fm801_controls_multi[i], chip)); + if (err < 0) + return err; + } } return 0; } diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f7a492c382d9..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -127,11 +127,15 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" + select SND_DYNAMIC_MINORS help Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. + Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS + to assure the multiple streams for DP-MST support. + comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index d3ea73171a3d..b9a6b66aeb0e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -793,11 +793,11 @@ EXPORT_SYMBOL_GPL(snd_hda_add_verbs); */ void snd_hda_apply_verbs(struct hda_codec *codec) { + const struct hda_verb **v; int i; - for (i = 0; i < codec->verbs.used; i++) { - struct hda_verb **v = snd_array_elem(&codec->verbs, i); + + snd_array_for_each(&codec->verbs, i, v) snd_hda_sequence_write(codec, *v); - } } EXPORT_SYMBOL_GPL(snd_hda_apply_verbs); @@ -890,10 +890,10 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; u32 cfg = pin->cfg; const struct hda_pintbl *t_pins; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bc3a7468e17..3fd0c16fa602 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -158,7 +158,7 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); if (len == -ENOSPC) { len = snd_hda_get_num_raw_conns(codec, nid); - result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); + result = kmalloc_array(len, sizeof(hda_nid_t), GFP_KERNEL); if (!result) return -ENOMEM; len = snd_hda_get_raw_connections(codec, nid, result, len); @@ -438,7 +438,7 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) int i; hda_nid_t nid; - codec->wcaps = kmalloc(codec->core.num_nodes * 4, GFP_KERNEL); + codec->wcaps = kmalloc_array(codec->core.num_nodes, 4, GFP_KERNEL); if (!codec->wcaps) return -ENOMEM; nid = codec->core.start_nid; @@ -481,9 +481,10 @@ static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, struct snd_array *array, hda_nid_t nid) { + struct hda_pincfg *pin; int i; - for (i = 0; i < array->used; i++) { - struct hda_pincfg *pin = snd_array_elem(array, i); + + snd_array_for_each(array, i, pin) { if (pin->nid == nid) return pin; } @@ -618,14 +619,15 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target); */ void snd_hda_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + /* don't shut up pins when unloading the driver; otherwise it breaks * the default pin setup at the next load of the driver */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ snd_hda_codec_read(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); @@ -638,13 +640,14 @@ EXPORT_SYMBOL_GPL(snd_hda_shutup_pins); /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + if (!codec->pins_shutup) return; if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { snd_hda_codec_write(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin->ctrl); @@ -697,8 +700,7 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->nid == nid) return p; } @@ -856,6 +858,39 @@ static void snd_hda_codec_dev_release(struct device *dev) kfree(codec); } +#define DEV_NAME_LEN 31 + +static int snd_hda_codec_device_init(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) +{ + char name[DEV_NAME_LEN]; + struct hda_codec *codec; + int err; + + dev_dbg(card->dev, "%s: entry\n", __func__); + + if (snd_BUG_ON(!bus)) + return -EINVAL; + if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) + return -EINVAL; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + sprintf(name, "hdaudioC%dD%d", card->number, codec_addr); + err = snd_hdac_device_init(&codec->core, &bus->core, name, codec_addr); + if (err < 0) { + kfree(codec); + return err; + } + + codec->core.type = HDA_DEV_LEGACY; + *codecp = codec; + + return err; +} + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -867,7 +902,19 @@ static void snd_hda_codec_dev_release(struct device *dev) int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp) { - struct hda_codec *codec; + int ret; + + ret = snd_hda_codec_device_init(bus, card, codec_addr, codecp); + if (ret < 0) + return ret; + + return snd_hda_codec_device_new(bus, card, codec_addr, *codecp); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_new); + +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec) +{ char component[31]; hda_nid_t fg; int err; @@ -877,25 +924,14 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, .dev_free = snd_hda_codec_dev_free, }; + dev_dbg(card->dev, "%s: entry\n", __func__); + if (snd_BUG_ON(!bus)) return -EINVAL; if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - sprintf(component, "hdaudioC%dD%d", card->number, codec_addr); - err = snd_hdac_device_init(&codec->core, &bus->core, component, - codec_addr); - if (err < 0) { - kfree(codec); - return err; - } - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.type = HDA_DEV_LEGACY; codec->core.exec_verb = codec_exec_verb; codec->bus = bus; @@ -955,15 +991,13 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (err < 0) goto error; - if (codecp) - *codecp = codec; return 0; error: put_device(hda_codec_dev(codec)); return err; } -EXPORT_SYMBOL_GPL(snd_hda_codec_new); +EXPORT_SYMBOL_GPL(snd_hda_codec_device_new); /** * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults @@ -1076,8 +1110,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, /* make other inactive cvts with the same stream-tag dirty */ type = get_wcaps_type(get_wcaps(codec, nid)); list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (!p->active && p->stream_tag == stream_tag && get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; @@ -1140,12 +1173,11 @@ static void really_cleanup_stream(struct hda_codec *codec, static void purify_inactive_streams(struct hda_codec *codec) { struct hda_codec *c; + struct hda_cvt_setup *p; int i; list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - struct hda_cvt_setup *p; - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (p->dirty) really_cleanup_stream(c, p); } @@ -1156,10 +1188,10 @@ static void purify_inactive_streams(struct hda_codec *codec) /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { + struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->stream_tag) really_cleanup_stream(codec, p); } @@ -1493,10 +1525,10 @@ static void get_ctl_amp_tlv(struct snd_kcontrol *kcontrol, unsigned int *tlv) val1 = ((int)val1) * ((int)val2); if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; - tlv[0] = SNDRV_CTL_TLVT_DB_SCALE; - tlv[1] = 2 * sizeof(unsigned int); - tlv[2] = val1; - tlv[3] = val2; + tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE; + tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int); + tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = val1; + tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = val2; } /** @@ -1544,10 +1576,10 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; step = (step + 1) * 25; - tlv[0] = SNDRV_CTL_TLVT_DB_SCALE; - tlv[1] = 2 * sizeof(unsigned int); - tlv[2] = -nums * step; - tlv[3] = step; + tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE; + tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int); + tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = -nums * step; + tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = step; } EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv); @@ -1845,10 +1877,10 @@ static int init_slave_0dB(struct snd_kcontrol *slave, } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) tlv = kctl->tlv.p; - if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_SCALE) + if (!tlv || tlv[SNDRV_CTL_TLVO_TYPE] != SNDRV_CTL_TLVT_DB_SCALE) return 0; - step = tlv[3]; + step = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP]; step &= ~TLV_DB_SCALE_MUTE; if (!step) return 0; @@ -1860,7 +1892,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave, } arg->step = step; - val = -tlv[2] / step; + val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step; if (val > 0) { put_kctl_with_value(slave, val); return val; @@ -2175,6 +2207,8 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.iec958.status[0] = spdif->status & 0xff; @@ -2282,6 +2316,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2308,6 +2344,8 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE; @@ -2336,6 +2374,8 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2461,10 +2501,10 @@ EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls); struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, hda_nid_t nid) { + struct hda_spdif_out *spdif; int i; - for (i = 0; i < codec->spdif_out.used; i++) { - struct hda_spdif_out *spdif = - snd_array_elem(&codec->spdif_out, i); + + snd_array_for_each(&codec->spdif_out, i, spdif) { if (spdif->nid == nid) return spdif; } @@ -2483,6 +2523,8 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); spdif->nid = (u16)-1; @@ -2503,6 +2545,8 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) struct hda_spdif_out *spdif; unsigned short val; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); if (spdif->nid != nid) { @@ -2887,8 +2931,9 @@ static int hda_codec_runtime_suspend(struct device *dev) list_for_each_entry(pcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); - if (codec_has_clkstop(codec) && codec_has_epss(codec) && - (state & AC_PWRST_CLK_STOP_OK)) + if (codec->link_down_at_suspend || + (codec_has_clkstop(codec) && codec_has_epss(codec) && + (state & AC_PWRST_CLK_STOP_OK))) snd_hdac_codec_link_down(&codec->core); snd_hdac_link_power(&codec->core, false); return 0; @@ -2979,6 +3024,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) sync_power_up_states(codec); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_build_controls); /* * PCM stuff @@ -3184,6 +3230,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_parse_pcms); /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 681c360f29f9..e03b5c1ccc5c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -258,6 +258,7 @@ struct hda_codec { unsigned int power_save_node:1; /* advanced PM for each widget */ unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ unsigned int force_pin_prefix:1; /* Add location prefix */ + unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; @@ -307,6 +308,8 @@ struct hda_codec { */ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5cc65093d941..db773e219aaa 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -264,10 +264,10 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, int anchor_nid) { struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->depth <= 0) continue; if ((!from_nid || path->path[0] == from_nid) && @@ -325,10 +325,10 @@ EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx); static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->path[0] == nid) return true; } @@ -351,11 +351,11 @@ static bool is_reachable_path(struct hda_codec *codec, static bool is_ctl_used(struct hda_codec *codec, unsigned int val, int type) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; val &= AMP_VAL_COMPARE_MASK; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if ((path->ctls[type] & AMP_VAL_COMPARE_MASK) == val) return true; } @@ -638,13 +638,13 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, { struct hda_gen_spec *spec = codec->spec; int type = get_wcaps_type(get_wcaps(codec, nid)); + const struct nid_path *path; int i, n; if (nid == codec->core.afg) return true; - for (n = 0; n < spec->paths.used; n++) { - struct nid_path *path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->active) continue; if (codec->power_save_node) { @@ -2065,7 +2065,7 @@ static int parse_output_paths(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, spec->vmaster_tlv); if (spec->dac_min_mute) - spec->vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; + spec->vmaster_tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] |= TLV_DB_SCALE_MUTE; } } @@ -2696,10 +2696,10 @@ static const struct snd_kcontrol_new out_jack_mode_enum = { static bool find_kctl_name(struct hda_codec *codec, const char *name, int idx) { struct hda_gen_spec *spec = codec->spec; + const struct snd_kcontrol_new *kctl; int i; - for (i = 0; i < spec->kctls.used; i++) { - struct snd_kcontrol_new *kctl = snd_array_elem(&spec->kctls, i); + snd_array_for_each(&spec->kctls, i, kctl) { if (!strcmp(kctl->name, name) && kctl->index == idx) return true; } @@ -4021,8 +4021,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path; int n; - for (n = 0; n < spec->paths.used; n++) { - path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->depth) continue; if (path->path[0] == nid || @@ -5831,10 +5830,10 @@ static void init_digital(struct hda_codec *codec) */ static void clear_unsol_on_unused_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a0c93b9c9a28..1ae1850b3bfd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2209,7 +2209,18 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ + SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ + SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ + SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 033aa84365b9..c6b778b2580c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -825,8 +825,9 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_caps & AC_WCAP_CONN_LIST) { conn_len = snd_hda_get_num_raw_conns(codec, nid); if (conn_len > 0) { - conn = kmalloc(sizeof(hda_nid_t) * conn_len, - GFP_KERNEL); + conn = kmalloc_array(conn_len, + sizeof(hda_nid_t), + GFP_KERNEL); if (!conn) return; if (snd_hda_get_raw_connections(codec, nid, conn, diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9b7efece4484..6ec79c58d48d 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -80,10 +80,10 @@ static ssize_t pin_configs_show(struct hda_codec *codec, struct snd_array *list, char *buf) { + const struct hda_pincfg *pin; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < list->used; i++) { - struct hda_pincfg *pin = snd_array_elem(list, i); + snd_array_for_each(list, i, pin) { len += sprintf(buf + len, "0x%02x 0x%08x\n", pin->nid, pin->cfg); } @@ -217,10 +217,10 @@ static ssize_t init_verbs_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_verb *v; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->init_verbs.used; i++) { - struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + snd_array_for_each(&codec->init_verbs, i, v) { len += snprintf(buf + len, PAGE_SIZE - len, "0x%02x 0x%03x 0x%04x\n", v->nid, v->verb, v->param); @@ -267,10 +267,10 @@ static ssize_t hints_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_hint *hint; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { len += snprintf(buf + len, PAGE_SIZE - len, "%s = %s\n", hint->key, hint->val); } @@ -280,10 +280,10 @@ static ssize_t hints_show(struct device *dev, static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) { + struct hda_hint *hint; int i; - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { if (!strcmp(hint->key, key)) return hint; } @@ -783,13 +783,13 @@ void snd_hda_sysfs_init(struct hda_codec *codec) void snd_hda_sysfs_clear(struct hda_codec *codec) { #ifdef CONFIG_SND_HDA_RECONFIG + struct hda_hint *hint; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { kfree(hint->key); /* we don't need to free hint->val */ } snd_array_free(&codec->hints); diff --git a/sound/pci/hda/hp_x360_helper.c b/sound/pci/hda/hp_x360_helper.c new file mode 100644 index 000000000000..969542c57358 --- /dev/null +++ b/sound/pci/hda/hp_x360_helper.c @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for HP X360 laptops with top B&O speakers + * to be included from codec driver + */ + +static void alc295_fixup_hp_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170110 }, + { } + }; + static const struct coef_fw alc295_hp_speakers_coefs[] = { + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0600), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0xc0c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0008), WRITE_COEF(0x28, 0xb000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x002e), WRITE_COEF(0x28, 0x0800), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x00c1), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0320), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0039), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003b), WRITE_COEF(0x28, 0xffff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003c), WRITE_COEF(0x28, 0xffd0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0080), WRITE_COEF(0x28, 0x0880), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x0dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0018), WRITE_COEF(0x28, 0x0219), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x005d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x9142), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c0), WRITE_COEF(0x28, 0x01ce), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c1), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c2), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c3), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c4), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c5), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c6), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c7), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c8), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c9), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ca), WRITE_COEF(0x28, 0x01c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cb), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cc), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cd), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ce), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cf), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d0), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d1), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d2), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d3), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0062), WRITE_COEF(0x28, 0x8000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0063), WRITE_COEF(0x28, 0x5f5f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0064), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0065), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0066), WRITE_COEF(0x28, 0x4004), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0067), WRITE_COEF(0x28, 0x0802), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0068), WRITE_COEF(0x28, 0x890f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0069), WRITE_COEF(0x28, 0xe021), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0070), WRITE_COEF(0x28, 0x8012), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0071), WRITE_COEF(0x28, 0x3450), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0072), WRITE_COEF(0x28, 0x0123), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0073), WRITE_COEF(0x28, 0x4543), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0074), WRITE_COEF(0x28, 0x2100), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0075), WRITE_COEF(0x28, 0x4321), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0076), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x8200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0051), WRITE_COEF(0x28, 0x0707), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0052), WRITE_COEF(0x28, 0x4090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x721f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0012), WRITE_COEF(0x28, 0xebeb), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x009e), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0060), WRITE_COEF(0x28, 0x2213), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x3000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0500), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0040), WRITE_COEF(0x28, 0x800c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0046), WRITE_COEF(0x28, 0xc22e), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x004b), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x82ec), WRITE_COEF(0x29, 0xb024), + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + alc295_fixup_disable_dac3(codec, fix, action); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, alc295_hp_speakers_coefs); + break; + } +} diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h deleted file mode 100644 index 3b8b7d78f9e0..000000000000 --- a/sound/pci/hda/local.h +++ /dev/null @@ -1,40 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - */ - -#ifndef __HDAC_LOCAL_H -#define __HDAC_LOCAL_H - -int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); - -#define get_wcaps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) -/* get the widget type from widget capability bits */ -static inline int get_wcaps_type(unsigned int wcaps) -{ - if (!wcaps) - return -1; /* invalid type */ - return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -} - -#define get_pin_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) - -static inline -unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) -{ - unsigned int val; - - if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) - return -1; - return val; -} - -#define get_amp_caps(codec, nid, dir) \ - hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) - -#define get_power_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) - -#endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea8651993..321e95c409c1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,9 @@ #include <linux/module.h> #include <linux/firmware.h> #include <linux/kernel.h> +#include <linux/types.h> +#include <linux/io.h> +#include <linux/pci.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -39,9 +42,15 @@ /* Enable this to see controls for tuning purpose. */ /*#define ENABLE_TUNING_CONTROLS*/ +#ifdef ENABLE_TUNING_CONTROLS +#include <sound/tlv.h> +#endif + #define FLOAT_ZERO 0x00000000 #define FLOAT_ONE 0x3f800000 #define FLOAT_TWO 0x40000000 +#define FLOAT_THREE 0x40400000 +#define FLOAT_EIGHT 0x41000000 #define FLOAT_MINUS_5 0xc0a00000 #define UNSOL_TAG_DSP 0x16 @@ -72,16 +81,22 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *const dirstr[2] = { "Playback", "Capture" }; +#define NUM_OF_OUTPUTS 3 enum { SPEAKER_OUT, - HEADPHONE_OUT + HEADPHONE_OUT, + SURROUND_OUT }; enum { @@ -89,6 +104,15 @@ enum { LINE_MIC_IN }; +/* Strings for Input Source Enum Control */ +static const char *const in_src_str[3] = {"Rear Mic", "Line", "Front Mic" }; +#define IN_SRC_NUM_OF_INPUTS 3 +enum { + REAR_MIC, + REAR_LINE_IN, + FRONT_MIC, +}; + enum { #define VNODE_START_NID 0x80 VNID_SPK = VNODE_START_NID, /* Speaker vnid */ @@ -122,13 +146,28 @@ enum { VOICEFX = IN_EFFECT_END_NID, PLAY_ENHANCEMENT, CRYSTAL_VOICE, - EFFECT_END_NID + EFFECT_END_NID, + OUTPUT_SOURCE_ENUM, + INPUT_SOURCE_ENUM, + XBASS_XOVER, + EQ_PRESET_ENUM, + SMART_VOLUME_ENUM, + MIC_BOOST_ENUM #define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID) }; /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* + * Default values for the effect slider controls, they are in order of their + * effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then + * X-bass. + */ +static const unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50}; +/* Amount of effect level sliders for ca0132_alt controls. */ +#define EFFECT_LEVEL_SLIDERS 5 + /* Latency introduced by DSP blocks in milliseconds. */ #define DSP_CAPTURE_INIT_LATENCY 0 #define DSP_CRYSTAL_VOICE_LATENCY 124 @@ -150,7 +189,7 @@ struct ct_effect { #define EFX_DIR_OUT 0 #define EFX_DIR_IN 1 -static struct ct_effect ca0132_effects[EFFECTS_COUNT] = { +static const struct ct_effect ca0132_effects[EFFECTS_COUNT] = { { .name = "Surround", .nid = SURROUND, .mid = 0x96, @@ -277,7 +316,7 @@ struct ct_tuning_ctl { unsigned int def_val;/*effect default values*/ }; -static struct ct_tuning_ctl ca0132_tuning_ctls[] = { +static const struct ct_tuning_ctl ca0132_tuning_ctls[] = { { .name = "Wedge Angle", .parent_nid = VOICE_FOCUS, .nid = WEDGE_ANGLE, @@ -392,14 +431,14 @@ struct ct_voicefx_preset { unsigned int vals[VOICEFX_MAX_PARAM_COUNT]; }; -static struct ct_voicefx ca0132_voicefx = { +static const struct ct_voicefx ca0132_voicefx = { .name = "VoiceFX Capture Switch", .nid = VOICEFX, .mid = 0x95, .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18} }; -static struct ct_voicefx_preset ca0132_voicefx_presets[] = { +static const struct ct_voicefx_preset ca0132_voicefx_presets[] = { { .name = "Neutral", .vals = { 0x00000000, 0x43C80000, 0x44AF0000, 0x44FA0000, 0x3F800000, 0x3F800000, @@ -472,6 +511,161 @@ static struct ct_voicefx_preset ca0132_voicefx_presets[] = { } }; +/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */ + +#define EQ_PRESET_MAX_PARAM_COUNT 11 + +struct ct_eq { + char *name; + hda_nid_t nid; + int mid; + int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/ +}; + +struct ct_eq_preset { + char *name; /*preset name*/ + unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT]; +}; + +static const struct ct_eq ca0132_alt_eq_enum = { + .name = "FX: Equalizer Preset Switch", + .nid = EQ_PRESET_ENUM, + .mid = 0x96, + .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20} +}; + + +static const struct ct_eq_preset ca0132_alt_eq_presets[] = { + { .name = "Flat", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + }, + { .name = "Acoustic", + .vals = { 0x00000000, 0x00000000, 0x3F8CCCCD, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Classical", + .vals = { 0x00000000, 0x00000000, 0x40C00000, + 0x40C00000, 0x40466666, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x40466666, 0x40466666 } + }, + { .name = "Country", + .vals = { 0x00000000, 0xBF99999A, 0x00000000, + 0x3FA66666, 0x3FA66666, 0x3F8CCCCD, + 0x00000000, 0x00000000, 0x40000000, + 0x40466666, 0x40800000 } + }, + { .name = "Dance", + .vals = { 0x00000000, 0xBF99999A, 0x40000000, + 0x40466666, 0x40866666, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Jazz", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x3F8CCCCD, 0x40800000, 0x40800000, + 0x40800000, 0x00000000, 0x3F8CCCCD, + 0x40466666, 0x40466666 } + }, + { .name = "New Age", + .vals = { 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x3F8CCCCD, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Pop", + .vals = { 0x00000000, 0xBFCCCCCD, 0x00000000, + 0x40000000, 0x40000000, 0x00000000, + 0xBF99999A, 0xBF99999A, 0x00000000, + 0x40466666, 0x40C00000 } + }, + { .name = "Rock", + .vals = { 0x00000000, 0xBF99999A, 0xBF99999A, + 0x3F8CCCCD, 0x40000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Vocal", + .vals = { 0x00000000, 0xC0000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x40466666, + 0x40800000, 0x40466666, 0x00000000, + 0x00000000, 0x3F8CCCCD } + } +}; + +/* DSP command sequences for ca0132_alt_select_out */ +#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */ +struct ca0132_alt_out_set { + char *name; /*preset name*/ + unsigned char commands; + unsigned int mids[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int vals[ALT_OUT_SET_MAX_COMMANDS]; +}; + +static const struct ca0132_alt_out_set alt_out_presets[] = { + { .name = "Line Out", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Headphone", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Surround", + .commands = 8, + .mids = { 0x96, 0x8F, 0x96, 0x96, + 0x96, 0x96, 0x96, 0x96 }, + .reqs = { 0x18, 0x01, 0x1F, 0x15, + 0x3A, 0x1A, 0x1B, 0x1C }, + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + } +}; + +/* + * DSP volume setting structs. Req 1 is left volume, req 2 is right volume, + * and I don't know what the third req is, but it's always zero. I assume it's + * some sort of update or set command to tell the DSP there's new volume info. + */ +#define DSP_VOL_OUT 0 +#define DSP_VOL_IN 1 + +struct ct_dsp_volume_ctl { + hda_nid_t vnid; + int mid; /* module ID*/ + unsigned int reqs[3]; /* scp req ID */ +}; + +static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = { + { .vnid = VNID_SPK, + .mid = 0x32, + .reqs = {3, 4, 2} + }, + { .vnid = VNID_MIC, + .mid = 0x37, + .reqs = {2, 3, 1} + } +}; + enum hda_cmd_vendor_io { /* for DspIO node */ VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000, @@ -698,11 +892,12 @@ enum dsp_download_state { */ struct ca0132_spec { - struct snd_kcontrol_new *mixers[5]; + const struct snd_kcontrol_new *mixers[5]; unsigned int num_mixers; const struct hda_verb *base_init_verbs; const struct hda_verb *base_exit_verbs; const struct hda_verb *chip_init_verbs; + const struct hda_verb *sbz_init_verbs; struct hda_verb *spec_init_verbs; struct auto_pin_cfg autocfg; @@ -719,6 +914,7 @@ struct ca0132_spec { hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; hda_nid_t unsol_tag_hp; + hda_nid_t unsol_tag_front_hp; /* for desktop ca0132 codecs */ hda_nid_t unsol_tag_amic1; /* chip access */ @@ -734,6 +930,9 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; + bool startup_check_entered; + bool dsp_reload; /* mixer and effects related */ unsigned char dmic_ctl; @@ -746,6 +945,17 @@ struct ca0132_spec { long effects_switch[EFFECTS_COUNT]; long voicefx_val; long cur_mic_boost; + /* ca0132_alt control related values */ + unsigned char in_enum_val; + unsigned char out_enum_val; + unsigned char mic_boost_enum_val; + unsigned char smart_volume_setting; + long fx_ctl_val[EFFECT_LEVEL_SLIDERS]; + long xbass_xover_freq; + long eq_preset_val; + unsigned int tlv[4]; + struct hda_vmaster_mute_hook vmaster_mute; + struct hda_codec *codec; struct delayed_work unsol_hp_work; @@ -754,6 +964,25 @@ struct ca0132_spec { #ifdef ENABLE_TUNING_CONTROLS long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif + /* + * Sound Blaster Z PCI region 2 iomem, used for input and output + * switching, and other unknown commands. + */ + void __iomem *mem_base; + + /* + * Whether or not to use the alt functions like alt_select_out, + * alt_select_in, etc. Only used on desktop codecs for now, because of + * surround sound support. + */ + bool use_alt_functions; + + /* + * Whether or not to use alt controls: volume effect sliders, EQ + * presets, smart volume presets, and new control names with FX prefix. + * Renames PlayEnhancement and CrystalVoice too. + */ + bool use_alt_controls; }; /* @@ -762,6 +991,9 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_ALIENWARE_M17XR4, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -778,10 +1010,46 @@ static const struct hda_pintbl alienware_pincfgs[] = { {} }; +/* Sound Blaster Z pin configs taken from Windows Driver */ +static const struct hda_pintbl sbz_pincfgs[] = { + { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c510f0 }, /* SPDIF In */ + { 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + +/* Recon3D integrated pin configs taken from Windows Driver */ +static const struct hda_pintbl r3di_pincfgs[] = { + { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x41c520f0 }, /* SPDIF In */ + { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x500000f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { + SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4), SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), {} }; @@ -965,6 +1233,29 @@ exit: } /* + * Write given value to the given address through the chip I/O widget. + * not protected by the Mutex + */ +static int chipio_write_no_mutex(struct hda_codec *codec, + unsigned int chip_addx, const unsigned int data) +{ + int err; + + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_write_data(codec, data); + if (err < 0) + goto exit; + +exit: + return err; +} + +/* * Write multiple values to the given address through the chip I/O widget. * protected by the Mutex */ @@ -1058,6 +1349,81 @@ static void chipio_set_control_param(struct hda_codec *codec, } /* + * Set chip parameters through the chip I/O widget. NO MUTEX. + */ +static void chipio_set_control_param_no_mutex(struct hda_codec *codec, + enum control_param_id param_id, int param_val) +{ + int val; + + if ((param_id < 32) && (param_val < 8)) { + val = (param_val << 5) | (param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_SET, val); + } else { + if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) { + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, + param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, + param_val); + } + } +} +/* + * Connect stream to a source point, and then connect + * that source point to a destination point. + */ +static void chipio_set_stream_source_dest(struct hda_codec *codec, + int streamid, int source_point, int dest_point) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point); +} + +/* + * Set number of channels in the selected stream. + */ +static void chipio_set_stream_channels(struct hda_codec *codec, + int streamid, unsigned int channels) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAMS_CHANNELS, channels); +} + +/* + * Enable/Disable audio stream. + */ +static void chipio_set_stream_control(struct hda_codec *codec, + int streamid, int enable) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_CONTROL, enable); +} + + +/* + * Set sampling rate of the connection point. NO MUTEX. + */ +static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec, + int connid, enum ca0132_sample_rate rate) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_ID, connid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate); +} + +/* * Set sampling rate of the connection point. */ static void chipio_set_conn_rate(struct hda_codec *codec, @@ -1420,8 +1786,8 @@ static int dspio_send_scp_message(struct hda_codec *codec, * Returns zero or a negative error code. */ static int dspio_scp(struct hda_codec *codec, - int mod_id, int req, int dir, void *data, unsigned int len, - void *reply, unsigned int *reply_len) + int mod_id, int src_id, int req, int dir, const void *data, + unsigned int len, void *reply, unsigned int *reply_len) { int status = 0; struct scp_msg scp_send, scp_reply; @@ -1445,7 +1811,7 @@ static int dspio_scp(struct hda_codec *codec, return -EINVAL; } - scp_send.hdr = make_scp_header(mod_id, 0x20, (dir == SCP_GET), req, + scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req, 0, 0, 0, len/sizeof(unsigned int)); if (data != NULL && len > 0) { len = min((unsigned int)(sizeof(scp_send.data)), len); @@ -1502,15 +1868,24 @@ static int dspio_scp(struct hda_codec *codec, * Set DSP parameters */ static int dspio_set_param(struct hda_codec *codec, int mod_id, - int req, void *data, unsigned int len) + int src_id, int req, const void *data, unsigned int len) { - return dspio_scp(codec, mod_id, req, SCP_SET, data, len, NULL, NULL); + return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, NULL, + NULL); } static int dspio_set_uint_param(struct hda_codec *codec, int mod_id, - int req, unsigned int data) + int req, const unsigned int data) { - return dspio_set_param(codec, mod_id, req, &data, sizeof(unsigned int)); + return dspio_set_param(codec, mod_id, 0x20, req, &data, + sizeof(unsigned int)); +} + +static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id, + int req, const unsigned int data) +{ + return dspio_set_param(codec, mod_id, 0x00, req, &data, + sizeof(unsigned int)); } /* @@ -1522,8 +1897,9 @@ static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan) unsigned int size = sizeof(dma_chan); codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n"); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_GET, NULL, 0, dma_chan, &size); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0, + dma_chan, &size); if (status < 0) { codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n"); @@ -1552,8 +1928,9 @@ static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan) codec_dbg(codec, " dspio_free_dma_chan() -- begin\n"); codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_SET, &dma_chan, sizeof(dma_chan), NULL, &dummy); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan, + sizeof(dma_chan), NULL, &dummy); if (status < 0) { codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n"); @@ -2575,14 +2952,16 @@ exit: */ static void dspload_post_setup(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); + if (!spec->use_alt_functions) { + /*set DSP speaker to 2.0 configuration*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - /*set DSP speaker to 2.0 configuration*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - - /*update write pointer*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + /*update write pointer*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + } } /** @@ -2690,6 +3069,170 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* + * Setup GPIO for the other variants of Core3D. + */ + +/* + * Sets up the GPIO pins so that they are discoverable. If this isn't done, + * the card shows as having no GPIO pins. + */ +static void ca0132_gpio_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); + snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); + break; + } + +} + +/* Sets the GPIO for audio output. */ +static void ca0132_gpio_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x04); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x06); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x1E); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x1F); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x0C); + break; + } +} + +/* + * GPIO control functions for the Recon3D integrated. + */ + +enum r3di_gpio_bit { + /* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */ + R3DI_MIC_SELECT_BIT = 1, + /* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */ + R3DI_OUT_SELECT_BIT = 2, + /* + * I dunno what this actually does, but it stays on until the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADING = 3, + /* + * Same as above, no clue what it does, but it comes on after the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADED = 4 +}; + +enum r3di_mic_select { + /* Set GPIO bit 1 to 0 for rear mic */ + R3DI_REAR_MIC = 0, + /* Set GPIO bit 1 to 1 for front microphone*/ + R3DI_FRONT_MIC = 1 +}; + +enum r3di_out_select { + /* Set GPIO bit 2 to 0 for headphone */ + R3DI_HEADPHONE_OUT = 0, + /* Set GPIO bit 2 to 1 for speaker */ + R3DI_LINE_OUT = 1 +}; +enum r3di_dsp_status { + /* Set GPIO bit 3 to 1 until DSP is downloaded */ + R3DI_DSP_DOWNLOADING = 0, + /* Set GPIO bit 4 to 1 once DSP is downloaded */ + R3DI_DSP_DOWNLOADED = 1 +}; + + +static void r3di_gpio_mic_set(struct hda_codec *codec, + enum r3di_mic_select cur_mic) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_mic) { + case R3DI_REAR_MIC: + cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT); + break; + case R3DI_FRONT_MIC: + cur_gpio |= (1 << R3DI_MIC_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +static void r3di_gpio_out_set(struct hda_codec *codec, + enum r3di_out_select cur_out) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_out) { + case R3DI_HEADPHONE_OUT: + cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT); + break; + case R3DI_LINE_OUT: + cur_gpio |= (1 << R3DI_OUT_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +static void r3di_gpio_dsp_status_set(struct hda_codec *codec, + enum r3di_dsp_status dsp_status) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (dsp_status) { + case R3DI_DSP_DOWNLOADING: + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + break; + case R3DI_DSP_DOWNLOADED: + /* Set DOWNLOADING bit to 0. */ + cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING); + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED); + break; + } + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2852,6 +3395,24 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, .tlv = { .c = ca0132_volume_tlv }, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) } +/* + * Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the + * volume put, which is used for setting the DSP volume. This was done because + * the ca0132 functions were taking too much time and causing lag. + */ +#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = snd_hda_mixer_amp_volume_info, \ + .get = snd_hda_mixer_amp_volume_get, \ + .put = ca0132_alt_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) } + #define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -2864,9 +3425,88 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, /* stereo */ #define CA0132_CODEC_VOL(xname, nid, dir) \ CA0132_CODEC_VOL_MONO(xname, nid, 3, dir) +#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \ + CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir) #define CA0132_CODEC_MUTE(xname, nid, dir) \ CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir) +/* lookup tables */ +/* + * Lookup table with decibel values for the DSP. When volume is changed in + * Windows, the DSP is also sent the dB value in floating point. In Windows, + * these values have decimal points, probably because the Windows driver + * actually uses floating point. We can't here, so I made a lookup table of + * values -90 to 9. -90 is the lowest decibel value for both the ADC's and the + * DAC's, and 9 is the maximum. + */ +static const unsigned int float_vol_db_lookup[] = { +0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000, +0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000, +0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000, +0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000, +0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000, +0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000, +0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000, +0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000, +0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000, +0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000, +0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000, +0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000, +0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000, +0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000, +0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000, +0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000, +0x40C00000, 0x40E00000, 0x41000000, 0x41100000 +}; + +/* + * This table counts from float 0 to 1 in increments of .01, which is + * useful for a few different sliders. + */ +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD, +0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE, +0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B, +0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F, +0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1, +0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333, +0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85, +0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7, +0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14, +0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D, +0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666, +0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F, +0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8, +0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1, +0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A, +0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333, +0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000 +}; + +/* + * This table counts from float 10 to 1000, which is the range of the x-bass + * crossover slider in Windows. + */ +static const unsigned int float_xbass_xover_lookup[] = { +0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000, +0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000, +0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000, +0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000, +0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000, +0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000, +0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000, +0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000, +0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000, +0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000, +0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000, +0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000, +0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000, +0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000, +0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000, +0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000, +0x44728000, 0x44750000, 0x44778000, 0x447A0000 +}; + /* The following are for tuning of products */ #ifdef ENABLE_TUNING_CONTROLS @@ -2942,7 +3582,7 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, break; snd_hda_power_up(codec); - dspio_set_param(codec, ca0132_tuning_ctls[i].mid, + dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, &(lookup[idx]), sizeof(unsigned int)); snd_hda_power_down(codec); @@ -3068,8 +3708,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); -static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0); static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, @@ -3207,7 +3847,7 @@ static int ca0132_select_out(struct hda_codec *codec) pin_ctl & ~PIN_HP); /* enable speaker node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); snd_hda_set_pin_ctl(codec, spec->out_pins[0], pin_ctl | PIN_OUT); } else { @@ -3251,13 +3891,209 @@ exit: return err < 0 ? err : 0; } +/* + * This function behaves similarly to the ca0132_select_out funciton above, + * except with a few differences. It adds the ability to select the current + * output with an enumerated control "output source" if the auto detect + * mute switch is set to off. If the auto detect mute switch is enabled, it + * will detect either headphone or lineout(SPEAKER_OUT) from jack detection. + * It also adds the ability to auto-detect the front headphone port. The only + * way to select surround is to disable auto detect, and set Surround with the + * enumerated control. + */ +static int ca0132_alt_select_out(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int pin_ctl; + int jack_present; + int auto_jack; + unsigned int i; + unsigned int tmp; + int err; + /* Default Headphone is rear headphone */ + hda_nid_t headphone_nid = spec->out_pins[1]; + + codec_dbg(codec, "%s\n", __func__); + + snd_hda_power_up_pm(codec); + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + /* + * If headphone rear or front is plugged in, set to headphone. + * If neither is plugged in, set to rear line out. Only if + * hp/speaker auto detect is enabled. + */ + if (auto_jack) { + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) || + snd_hda_jack_detect(codec, spec->unsol_tag_front_hp); + + if (jack_present) + spec->cur_out_type = HEADPHONE_OUT; + else + spec->cur_out_type = SPEAKER_OUT; + } else + spec->cur_out_type = spec->out_enum_val; + + /* Begin DSP output switch */ + tmp = FLOAT_ONE; + err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp); + if (err < 0) + goto exit; + + switch (spec->cur_out_type) { + case SPEAKER_OUT: + codec_dbg(codec, "%s speaker\n", __func__); + /*speaker out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + + /* disable headphone node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* enable line-out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Enable EAPD */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + + /* If PlayEnhancement is enabled, set different source */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + break; + case HEADPHONE_OUT: + codec_dbg(codec, "%s hp\n", __func__); + /* Headphone out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0107, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x12); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x21); + r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); + break; + } + + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + /* disable speaker*/ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl & ~PIN_HP); + + /* enable headphone, either front or rear */ + + if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp)) + headphone_nid = spec->out_pins[2]; + else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp)) + headphone_nid = spec->out_pins[1]; + + pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, headphone_nid, + pin_ctl | PIN_HP); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO); + break; + case SURROUND_OUT: + codec_dbg(codec, "%s surround\n", __func__); + /* Surround out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + /* enable line out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Disable headphone out */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* Enable EAPD on line out */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + /* enable center/lfe out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[2], + pin_ctl | PIN_OUT); + /* Now set rear surround node as out. */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[3], + pin_ctl | PIN_OUT); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + break; + } + + /* run through the output dsp commands for line-out */ + for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) { + err = dspio_set_uint_param(codec, + alt_out_presets[spec->cur_out_type].mids[i], + alt_out_presets[spec->cur_out_type].reqs[i], + alt_out_presets[spec->cur_out_type].vals[i]); + + if (err < 0) + goto exit; + } + +exit: + snd_hda_power_down_pm(codec); + + return err < 0 ? err : 0; +} + static void ca0132_unsol_hp_delayed(struct work_struct *work) { struct ca0132_spec *spec = container_of( to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; - ca0132_select_out(spec->codec); + if (spec->use_alt_functions) + ca0132_alt_select_out(spec->codec); + else + ca0132_select_out(spec->codec); + jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp); if (jack) { jack->block_report = 0; @@ -3268,6 +4104,10 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) static void ca0132_set_dmic(struct hda_codec *codec, int enable); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); +static void resume_mic1(struct hda_codec *codec, unsigned int oldval); +static int stop_mic1(struct hda_codec *codec); +static int ca0132_cvoice_switch_set(struct hda_codec *codec); +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val); /* * Select the active VIP source @@ -3310,6 +4150,71 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val) return 1; } +static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + codec_dbg(codec, "%s\n", __func__); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* if CrystalVoice is off, vipsource should be 0 */ + if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] || + (val == 0) || spec->in_enum_val == REAR_LINE_IN) { + codec_dbg(codec, "%s: off.", __func__); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + + if (spec->in_enum_val == REAR_LINE_IN) + tmp = FLOAT_ZERO; + else { + if (spec->quirk == QUIRK_SBZ) + tmp = FLOAT_THREE; + else + tmp = FLOAT_ONE; + } + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + } else { + codec_dbg(codec, "%s: on.", __func__); + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_16_000); + + if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) + tmp = FLOAT_TWO; + else + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + msleep(20); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + return 1; +} + /* * Select the active microphone. * If autodetect is enabled, mic will be selected based on jack detection. @@ -3363,6 +4268,125 @@ static int ca0132_select_mic(struct hda_codec *codec) } /* + * Select the active input. + * Mic detection isn't used, because it's kind of pointless on the SBZ. + * The front mic has no jack-detection, so the only way to switch to it + * is to do it manually in alsamixer. + */ +static int ca0132_alt_select_in(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + codec_dbg(codec, "%s\n", __func__); + + snd_hda_power_up_pm(codec); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + spec->cur_mic_type = spec->in_enum_val; + + switch (spec->cur_mic_type) { + case REAR_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + tmp = FLOAT_THREE; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + tmp = FLOAT_ONE; + break; + default: + tmp = FLOAT_ONE; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x0000000C); + } + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + case REAR_LINE_IN: + ca0132_mic_boost_set(codec, 0); + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x00000000); + chipio_write(codec, 0x18B09C, 0x00000000); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + break; + case FRONT_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0100, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + tmp = FLOAT_THREE; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_FRONT_MIC); + tmp = FLOAT_ONE; + break; + default: + tmp = FLOAT_ONE; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x000000CC); + } + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + } + ca0132_cvoice_switch_set(codec); + + snd_hda_power_down_pm(codec); + return 0; + +} + +/* * Check if VNODE settings take effect immediately. */ static bool ca0132_is_vnode_effective(struct hda_codec *codec, @@ -3418,7 +4442,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) { struct ca0132_spec *spec = codec->spec; - unsigned int on; + unsigned int on, tmp; int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; int err = 0; int idx = nid - EFFECT_START_NID; @@ -3442,6 +4466,46 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) /* Voice Focus applies to 2-ch Mic, Digital Mic */ if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC)) val = 0; + + /* If Voice Focus on SBZ, set to two channel. */ + if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + + if (spec->effects_switch[VOICE_FOCUS - + EFFECT_START_NID]) { + tmp = FLOAT_TWO; + val = 1; + } else + tmp = FLOAT_ONE; + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + } + } + /* + * For SBZ noise reduction, there's an extra command + * to module ID 0x47. No clue why. + */ + if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + if (spec->effects_switch[NOISE_REDUCTION - + EFFECT_START_NID]) + tmp = FLOAT_ONE; + else + tmp = FLOAT_ZERO; + } else + tmp = FLOAT_ZERO; + + dspio_set_uint_param(codec, 0x47, 0x00, tmp); + } + + /* If rear line in disable effects. */ + if (spec->use_alt_functions && + spec->in_enum_val == REAR_LINE_IN) + val = 0; } codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n", @@ -3469,6 +4533,9 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + i = OUT_EFFECT_START_NID - EFFECT_START_NID; nid = OUT_EFFECT_START_NID; /* PE affects all out effects */ @@ -3526,7 +4593,10 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - ret |= ca0132_set_vipsource(codec, 1); + if (spec->use_alt_functions) + ret |= ca0132_alt_set_vipsource(codec, 1); + else + ret |= ca0132_set_vipsource(codec, 1); resume_mic1(codec, oldval); return ret; } @@ -3546,6 +4616,16 @@ static int ca0132_mic_boost_set(struct hda_codec *codec, long val) return ret; } +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val) +{ + struct ca0132_spec *spec = codec->spec; + int ret = 0; + + ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0, + HDA_INPUT, 0, HDA_AMP_VOLMASK, val); + return ret; +} + static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3560,8 +4640,12 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, if (nid == VNID_HP_SEL) { auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; - if (!auto_jack) - ca0132_select_out(codec); + if (!auto_jack) { + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + else + ca0132_select_out(codec); + } return 1; } @@ -3574,7 +4658,10 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - ca0132_select_out(codec); + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + else + ca0132_select_out(codec); return 1; } @@ -3602,6 +4689,432 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, return ret; } /* End of control change helpers. */ +/* + * Below I've added controls to mess with the effect levels, I've only enabled + * them on the Sound Blaster Z, but they would probably also work on the + * Chromebook. I figured they were probably tuned specifically for it, and left + * out for a reason. + */ + +/* Sets DSP effect level from the sliders above the controls */ +static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid, + const unsigned int *lookup, int idx) +{ + int i = 0; + unsigned int y; + /* + * For X_BASS, req 2 is actually crossover freq instead of + * effect level + */ + if (nid == X_BASS) + y = 2; + else + y = 1; + + snd_hda_power_up(codec); + if (nid == XBASS_XOVER) { + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (ca0132_effects[i].nid == X_BASS) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[1], + &(lookup[idx - 1]), sizeof(unsigned int)); + } else { + /* Find the actual effect structure */ + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (nid == ca0132_effects[i].nid) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[y], + &(lookup[idx]), sizeof(unsigned int)); + } + + snd_hda_power_down(codec); + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->xbass_xover_freq; + return 0; +} + +static int ca0132_alt_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx = nid - OUT_EFFECT_START_NID; + + *valp = spec->fx_ctl_val[idx]; + return 0; +} + +/* + * The X-bass crossover starts at 10hz, so the min is 1. The + * frequency is set in multiples of 10. + */ +static int ca0132_alt_xbass_xover_slider_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int chs = get_amp_channels(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + /* any change? */ + if (spec->xbass_xover_freq == *valp) + return 0; + + spec->xbass_xover_freq = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx); + + return 0; +} + +static int ca0132_alt_effect_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + idx = nid - EFFECT_START_NID; + /* any change? */ + if (spec->fx_ctl_val[idx] == *valp) + return 0; + + spec->fx_ctl_val[idx] = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx); + + return 0; +} + + +/* + * Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original + * only has off or full 30 dB, and didn't like making a volume slider that has + * traditional 0-100 in alsamixer that goes in big steps. I like enum better. + */ +#define MIC_BOOST_NUM_OF_STEPS 4 +#define MIC_BOOST_ENUM_MAX_STRLEN 10 + +static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + char *sfx = "dB"; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS; + if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS) + uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1; + sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx); + strcpy(uinfo->value.enumerated.name, namestr); + return 0; +} + +static int ca0132_alt_mic_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->mic_boost_enum_val; + return 0; +} + +static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = MIC_BOOST_NUM_OF_STEPS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n", + sel); + + spec->mic_boost_enum_val = sel; + + if (spec->in_enum_val != REAR_LINE_IN) + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + + return 1; +} + + +/* + * Input Select Control for alternative ca0132 codecs. This exists because + * front microphone has no auto-detect, and we need a way to set the rear + * as line-in + */ +static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS; + if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS) + uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1; + strcpy(uinfo->value.enumerated.name, + in_src_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_input_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->in_enum_val; + return 0; +} + +static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = IN_SRC_NUM_OF_INPUTS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n", + sel, in_src_str[sel]); + + spec->in_enum_val = sel; + + ca0132_alt_select_in(codec); + + return 1; +} + +/* Sound Blaster Z Output Select Control */ +static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = NUM_OF_OUTPUTS; + if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS) + uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1; + strcpy(uinfo->value.enumerated.name, + alt_out_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_output_select_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->out_enum_val; + return 0; +} + +static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = NUM_OF_OUTPUTS; + unsigned int auto_jack; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n", + sel, alt_out_presets[sel].name); + + spec->out_enum_val = sel; + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + if (!auto_jack) + ca0132_alt_select_out(codec); + + return 1; +} + +/* + * Smart Volume output setting control. Three different settings, Normal, + * which takes the value from the smart volume slider. The two others, loud + * and night, disregard the slider value and have uneditable values. + */ +#define NUM_OF_SVM_SETTINGS 3 +static const char *const out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" }; + +static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS; + if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS) + uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1; + strcpy(uinfo->value.enumerated.name, + out_svm_set_enum_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_svm_setting_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->smart_volume_setting; + return 0; +} + +static int ca0132_alt_svm_setting_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = NUM_OF_SVM_SETTINGS; + unsigned int idx = SMART_VOLUME - EFFECT_START_NID; + unsigned int tmp; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n", + sel, out_svm_set_enum_str[sel]); + + spec->smart_volume_setting = sel; + + switch (sel) { + case 0: + tmp = FLOAT_ZERO; + break; + case 1: + tmp = FLOAT_ONE; + break; + case 2: + tmp = FLOAT_TWO; + break; + default: + tmp = FLOAT_ZERO; + break; + } + /* Req 2 is the Smart Volume Setting req. */ + dspio_set_uint_param(codec, ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[2], tmp); + return 1; +} + +/* Sound Blaster Z EQ preset controls */ +static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + ca0132_alt_eq_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_eq_preset_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->eq_preset_val; + return 0; +} + +static int ca0132_alt_eq_preset_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int i, err = 0; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets); + + if (sel >= items) + return 0; + + codec_dbg(codec, "%s: sel=%d, preset=%s\n", __func__, sel, + ca0132_alt_eq_presets[sel].name); + /* + * Idx 0 is default. + * Default needs to qualify with CrystalVoice state. + */ + for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) { + err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid, + ca0132_alt_eq_enum.reqs[i], + ca0132_alt_eq_presets[sel].vals[i]); + if (err < 0) + break; + } + + if (err >= 0) + spec->eq_preset_val = sel; + + return 1; +} static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -3753,10 +5266,15 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, /* mic boost */ if (nid == spec->input_pins[0]) { spec->cur_mic_boost = *valp; + if (spec->use_alt_functions) { + if (spec->in_enum_val != REAR_LINE_IN) + changed = ca0132_mic_boost_set(codec, *valp); + } else { + /* Mic boost does not apply to Digital Mic */ + if (spec->cur_mic_type != DIGITAL_MIC) + changed = ca0132_mic_boost_set(codec, *valp); + } - /* Mic boost does not apply to Digital Mic */ - if (spec->cur_mic_type != DIGITAL_MIC) - changed = ca0132_mic_boost_set(codec, *valp); goto exit; } @@ -3768,6 +5286,41 @@ exit: /* * Volume related */ +/* + * Sets the internal DSP decibel level to match the DAC for output, and the + * ADC for input. Currently only the SBZ sets dsp capture volume level, and + * all alternative codecs set DSP playback volume. + */ +static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_dir; + unsigned int lookup_val; + + if (nid == VNID_SPK) + dsp_dir = DSP_VOL_OUT; + else + dsp_dir = DSP_VOL_IN; + + lookup_val = spec->vnode_lvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[0], + float_vol_db_lookup[lookup_val]); + + lookup_val = spec->vnode_rvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[1], + float_vol_db_lookup[lookup_val]); + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO); +} + static int ca0132_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -3869,6 +5422,51 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol, return changed; } +/* + * This function is the same as the one above, because using an if statement + * inside of the above volume control for the DSP volume would cause too much + * lag. This is a lot more smooth. + */ +static int ca0132_alt_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + int ch = get_amp_channels(kcontrol); + long *valp = ucontrol->value.integer.value; + hda_nid_t vnid = 0; + int changed = 1; + + switch (nid) { + case 0x02: + vnid = VNID_SPK; + break; + case 0x07: + vnid = VNID_MIC; + break; + } + + /* store the left and right volume */ + if (ch & 1) { + spec->vnode_lvol[vnid - VNODE_START_NID] = *valp; + valp++; + } + if (ch & 2) { + spec->vnode_rvol[vnid - VNODE_START_NID] = *valp; + valp++; + } + + snd_hda_power_up(codec); + ca0132_alt_dsp_volume_put(codec, vnid); + mutex_lock(&codec->control_mutex); + changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); + mutex_unlock(&codec->control_mutex); + snd_hda_power_down(codec); + + return changed; +} + static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -3907,14 +5505,59 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, return err; } +/* Add volume slider control for effect level */ +static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid, + const char *pfx, int dir) +{ + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type); + + sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]); + + knew.tlv.c = 0; + knew.tlv.p = 0; + + switch (nid) { + case XBASS_XOVER: + knew.info = ca0132_alt_xbass_xover_slider_info; + knew.get = ca0132_alt_xbass_xover_slider_ctl_get; + knew.put = ca0132_alt_xbass_xover_slider_put; + break; + default: + knew.info = ca0132_alt_effect_slider_info; + knew.get = ca0132_alt_slider_ctl_get; + knew.put = ca0132_alt_effect_slider_put; + knew.private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, type); + break; + } + + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +/* + * Added FX: prefix for the alternative codecs, because otherwise the surround + * effect would conflict with the Surround sound volume control. Also seems more + * clear as to what the switches do. Left alone for others. + */ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int dir) { + struct ca0132_spec *spec = codec->spec; char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type); - sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + /* If using alt_controls, add FX: prefix. But, don't add FX: + * prefix to OutFX or InFX enable controls. + */ + if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID)) + sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]); + else + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -3929,11 +5572,141 @@ static int add_voicefx(struct hda_codec *codec) return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec)); } +/* Create the EQ Preset control */ +static int add_ca0132_alt_eq_presets(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name, + EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_eq_preset_info; + knew.get = ca0132_alt_eq_preset_get; + knew.put = ca0132_alt_eq_preset_put; + return snd_hda_ctl_add(codec, EQ_PRESET_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add enumerated control for the three different settings of the smart volume + * output effect. Normal just uses the slider value, and loud and night are + * their own things that ignore that value. + */ +static int ca0132_alt_add_svm_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting", + SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_svm_setting_info; + knew.get = ca0132_alt_svm_setting_get; + knew.put = ca0132_alt_svm_setting_put; + return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM, + snd_ctl_new1(&knew, codec)); + +} + +/* + * Create an Output Select enumerated control for codecs with surround + * out capabilities. + */ +static int ca0132_alt_add_output_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Output Select", + OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_output_select_get_info; + knew.get = ca0132_alt_output_select_get; + knew.put = ca0132_alt_output_select_put; + return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Create an Input Source enumerated control for the alternate ca0132 codecs + * because the front microphone has no auto-detect, and Line-in has to be set + * somehow. + */ +static int ca0132_alt_add_input_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Input Source", + INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_input_source_info; + knew.get = ca0132_alt_input_source_get; + knew.put = ca0132_alt_input_source_put; + return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add mic boost enumerated control. Switches through 0dB to 30dB. This adds + * more control than the original mic boost, which is either full 30dB or off. + */ +static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch", + MIC_BOOST_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_mic_boost_info; + knew.get = ca0132_alt_mic_boost_get; + knew.put = ca0132_alt_mic_boost_put; + return snd_hda_ctl_add(codec, MIC_BOOST_ENUM, + snd_ctl_new1(&knew, codec)); + +} + +/* + * Need to create slave controls for the alternate codecs that have surround + * capabilities. + */ +static const char * const ca0132_alt_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", NULL, +}; + +/* + * Also need special channel map, because the default one is incorrect. + * I think this has to do with the pin for rear surround being 0x11, + * and the center/lfe being 0x10. Usually the pin order is the opposite. + */ +static const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +/* Add the correct chmap for streams with 6 channels. */ +static void ca0132_alt_add_chmap_ctls(struct hda_codec *codec) +{ + int err = 0; + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + struct hda_pcm_stream *hinfo = + &pcm->stream[SNDRV_PCM_STREAM_PLAYBACK]; + struct snd_pcm_chmap *chmap; + const struct snd_pcm_chmap_elem *elem; + + elem = ca0132_alt_chmaps; + if (hinfo->channels_max == 6) { + err = snd_pcm_add_chmap_ctls(pcm->pcm, + SNDRV_PCM_STREAM_PLAYBACK, + elem, hinfo->channels_max, 0, &chmap); + if (err < 0) + codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!"); + } + } +} + /* * When changing Node IDs for Mixer Controls below, make sure to update * Node IDs in ca0132_config() as well. */ -static struct snd_kcontrol_new ca0132_mixer[] = { +static const struct snd_kcontrol_new ca0132_mixer[] = { CA0132_CODEC_VOL("Master Playback Volume", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_MUTE("Master Playback Switch", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), @@ -3955,10 +5728,55 @@ static struct snd_kcontrol_new ca0132_mixer[] = { { } /* end */ }; +/* + * SBZ specific control mixer. Removes auto-detect for mic, and adds surround + * controls. Also sets both the Front Playback and Capture Volume controls to + * alt so they set the DSP's decibel level. + */ +static const struct snd_kcontrol_new sbz_mixer[] = { + CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), + CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT), + CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT), + CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT), + HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT), + HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT), + CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch", + VNID_HP_ASEL, 1, HDA_OUTPUT), + { } /* end */ +}; + +/* + * Same as the Sound Blaster Z, except doesn't use the alt volume for capture + * because it doesn't set decibel levels for the DSP for capture. + */ +static const struct snd_kcontrol_new r3di_mixer[] = { + CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), + CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT), + CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), + CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT), + HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT), + HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT), + CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch", + VNID_HP_ASEL, 1, HDA_OUTPUT), + { } /* end */ +}; + static int ca0132_build_controls(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - int i, num_fx; + int i, num_fx, num_sliders; int err = 0; /* Add Mixer controls */ @@ -3967,29 +5785,94 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + /* Setup vmaster with surround slaves for desktop ca0132 devices */ + if (spec->use_alt_functions) { + snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, + spec->tlv); + snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->tlv, ca0132_alt_slave_pfxs, + "Playback Volume"); + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, ca0132_alt_slave_pfxs, + "Playback Switch", + true, &spec->vmaster_mute.sw_kctl); + + } /* Add in and out effects controls. * VoiceFX, PE and CrystalVoice are added separately. */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { + /* SBZ breaks if Echo Cancellation is used */ + if (spec->quirk == QUIRK_SBZ) { + if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + + OUT_EFFECTS_COUNT)) + continue; + } + err = add_fx_switch(codec, ca0132_effects[i].nid, ca0132_effects[i].name, ca0132_effects[i].direct); if (err < 0) return err; } + /* + * If codec has use_alt_controls set to true, add effect level sliders, + * EQ presets, and Smart Volume presets. Also, change names to add FX + * prefix, and change PlayEnhancement and CrystalVoice to match. + */ + if (spec->use_alt_controls) { + ca0132_alt_add_svm_enum(codec); + add_ca0132_alt_eq_presets(codec); + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "Enable OutFX", 0); + if (err < 0) + return err; - err = add_fx_switch(codec, PLAY_ENHANCEMENT, "PlayEnhancement", 0); - if (err < 0) - return err; + err = add_fx_switch(codec, CRYSTAL_VOICE, + "Enable InFX", 1); + if (err < 0) + return err; - err = add_fx_switch(codec, CRYSTAL_VOICE, "CrystalVoice", 1); - if (err < 0) - return err; + num_sliders = OUT_EFFECTS_COUNT - 1; + for (i = 0; i < num_sliders; i++) { + err = ca0132_alt_add_effect_slider(codec, + ca0132_effects[i].nid, + ca0132_effects[i].name, + ca0132_effects[i].direct); + if (err < 0) + return err; + } + err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER, + "X-Bass Crossover", EFX_DIR_OUT); + + if (err < 0) + return err; + } else { + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "PlayEnhancement", 0); + if (err < 0) + return err; + + err = add_fx_switch(codec, CRYSTAL_VOICE, + "CrystalVoice", 1); + if (err < 0) + return err; + } add_voicefx(codec); + /* + * If the codec uses alt_functions, you need the enumerated controls + * to select the new outputs and inputs, plus add the new mic boost + * setting control. + */ + if (spec->use_alt_functions) { + ca0132_alt_add_output_enum(codec); + ca0132_alt_add_input_enum(codec); + ca0132_alt_add_mic_boost_enum(codec); + } #ifdef ENABLE_TUNING_CONTROLS add_tuning_ctls(codec); #endif @@ -4014,6 +5897,10 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + if (spec->use_alt_functions) + ca0132_alt_add_chmap_ctls(codec); + return 0; } @@ -4068,6 +5955,11 @@ static int ca0132_build_pcms(struct hda_codec *codec) info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; + if (spec->use_alt_functions) { + info->own_chmap = true; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap + = ca0132_alt_chmaps; + } info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = @@ -4076,12 +5968,16 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); - if (!info) - return -ENOMEM; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + /* With the DSP enabled, desktops don't use this ADC. */ + if (!spec->use_alt_functions) { + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + } info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); if (!info) @@ -4237,7 +6133,10 @@ static void ca0132_init_dmic(struct hda_codec *codec) * Bit 6: set to select Data2, clear for Data1 * Bit 7: set to enable DMic, clear for AMic */ - val = 0x23; + if (spec->quirk == QUIRK_ALIENWARE_M17XR4) + val = 0x33; + else + val = 0x23; /* keep a copy of dmic ctl val for enable/disable dmic purpuse */ spec->dmic_ctl = val; snd_hda_codec_write(codec, spec->input_pins[0], 0, @@ -4288,6 +6187,196 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* + * Recon3Di r3di_setup_defaults sub functions. + */ + +static void r3di_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void r3di_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */ + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); +} + +/* + * Initialize Sound Blaster Z analog microphones. + */ +static void sbz_init_analog_mics(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + tmp = FLOAT_THREE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); + +} + +/* + * Sets the source of stream 0x14 to connpointID 0x48, and the destination + * connpointID to 0x91. If this isn't done, the destination is 0x71, and + * you get no sound. I'm guessing this has to do with the Sound Blaster Z + * having an updated DAC, which changes the destination to that DAC. + */ +static void sbz_connect_streams(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n"); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + + /* This value is 0x43 for 96khz, and 0x83 for 192khz. */ + chipio_write_no_mutex(codec, 0x18a020, 0x00000043); + + /* Setup stream 0x14 with it's source and destination points */ + chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91); + chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000); + chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000); + chipio_set_stream_channels(codec, 0x14, 2); + chipio_set_stream_control(codec, 0x14, 1); + + codec_dbg(codec, "Connect Streams exited, mutex released.\n"); + + mutex_unlock(&spec->chipio_mutex); + +} + +/* + * Write data through ChipIO to setup proper stream destinations. + * Not sure how it exactly works, but it seems to direct data + * to different destinations. Example is f8 to c0, e0 to c0. + * All I know is, if you don't set these, you get no sound. + */ +static void sbz_chipio_startup_data(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n"); + + /* These control audio output */ + chipio_write_no_mutex(codec, 0x190060, 0x0001f8c0); + chipio_write_no_mutex(codec, 0x190064, 0x0001f9c1); + chipio_write_no_mutex(codec, 0x190068, 0x0001fac6); + chipio_write_no_mutex(codec, 0x19006c, 0x0001fbc7); + /* Signal to update I think */ + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + /* No clue what these control */ + chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); + chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); + chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); + chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3); + chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4); + chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5); + chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6); + chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7); + chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8); + chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); + chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); + chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); + + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + codec_dbg(codec, "Startup Data exited, mutex released.\n"); + mutex_unlock(&spec->chipio_mutex); +} + +/* + * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands + * without a 0x20 source like normal. + */ +static void sbz_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void sbz_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + + tmp = FLOAT_THREE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + chipio_write(codec, 0x18b098, 0x0000000c); + chipio_write(codec, 0x18b09C, 0x0000000c); +} + +/* * Setup default parameters for DSP */ static void ca0132_setup_defaults(struct hda_codec *codec) @@ -4332,16 +6421,159 @@ static void ca0132_setup_defaults(struct hda_codec *codec) } /* + * Setup default parameters for Recon3Di DSP. + */ + +static void r3di_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + r3di_dsp_scp_startup(codec); + + r3di_dsp_initial_mic_setup(codec); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + + /* Setup effect defaults */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + +} + +/* + * Setup default parameters for the Sound Blaster Z DSP. A lot more going on + * than the Chromebook setup. + */ +static void sbz_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp, stream_format; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + sbz_dsp_scp_startup(codec); + + sbz_init_analog_mics(codec); + + sbz_connect_streams(codec); + + sbz_chipio_startup_data(codec); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + /* + * Sets internal input loopback to off, used to have a switch to + * enable input loopback, but turned out to be way too buggy. + */ + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x37, 0x08, tmp); + dspio_set_uint_param(codec, 0x37, 0x10, tmp); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + sbz_dsp_initial_mic_setup(codec); + + + /* out, in effects + voicefx */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + + /* + * Have to make a stream to bind the sound output to, otherwise + * you'll get dead audio. Before I did this, it would bind to an + * audio input, and would never work + */ + stream_format = snd_hdac_calc_stream_format(48000, 2, + SNDRV_PCM_FORMAT_S32_LE, 32, 0); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); + + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); + + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); +} + +/* * Initialization of flags in chip */ static void ca0132_init_flags(struct hda_codec *codec) { - chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) { + chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1); + } else { + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + } } /* @@ -4349,6 +6581,16 @@ static void ca0132_init_flags(struct hda_codec *codec) */ static void ca0132_init_params(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) { + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + chipio_set_conn_rate(codec, 0x0B, SR_48_000); + chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); + chipio_set_control_param(codec, 0, 0); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + } + chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6); chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6); } @@ -4370,11 +6612,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* + * Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (!spec->alt_firmware_present) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { @@ -4402,13 +6682,17 @@ static void ca0132_download_dsp(struct hda_codec *codec) return; /* don't retry failures */ chipio_enable_clocks(codec); - spec->dsp_state = DSP_DOWNLOADING; - if (!ca0132_download_dsp_images(codec)) - spec->dsp_state = DSP_DOWNLOAD_FAILED; - else - spec->dsp_state = DSP_DOWNLOADED; + if (spec->dsp_state != DSP_DOWNLOADED) { + spec->dsp_state = DSP_DOWNLOADING; - if (spec->dsp_state == DSP_DOWNLOADED) + if (!ca0132_download_dsp_images(codec)) + spec->dsp_state = DSP_DOWNLOAD_FAILED; + else + spec->dsp_state = DSP_DOWNLOADED; + } + + /* For codecs using alt functions, this is already done earlier */ + if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions)) ca0132_set_dsp_msr(codec, true); } @@ -4454,6 +6738,10 @@ static void ca0132_init_unsol(struct hda_codec *codec) amic_callback); snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); + /* Front headphone jack detection */ + if (spec->use_alt_functions) + snd_hda_jack_detect_enable_callback(codec, + spec->unsol_tag_front_hp, hp_callback); } /* @@ -4476,7 +6764,8 @@ static struct hda_verb ca0132_base_exit_verbs[] = { {} }; -/* Other verbs tables. Sends after DSP download. */ +/* Other verbs tables. Sends after DSP download. */ + static struct hda_verb ca0132_init_verbs0[] = { /* chip init verbs */ {0x15, 0x70D, 0xF0}, @@ -4506,8 +6795,27 @@ static struct hda_verb ca0132_init_verbs0[] = { {0x15, 0x546, 0xC9}, {0x15, 0x53B, 0xCE}, {0x15, 0x5E8, 0xC9}, - {0x15, 0x717, 0x0D}, - {0x15, 0x718, 0x20}, + {} +}; + +/* Extra init verbs for SBZ */ +static struct hda_verb sbz_init_verbs[] = { + {0x15, 0x70D, 0x20}, + {0x15, 0x70E, 0x19}, + {0x15, 0x707, 0x00}, + {0x15, 0x539, 0xCE}, + {0x15, 0x546, 0xC9}, + {0x15, 0x70D, 0xB7}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x10}, + {0x15, 0x70D, 0xAF}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x01}, + {0x15, 0x707, 0x05}, + {0x15, 0x70D, 0x73}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x14}, + {0x15, 0x6FF, 0xC4}, {} }; @@ -4521,7 +6829,11 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - spec->cur_mic_type = DIGITAL_MIC; + if (!spec->use_alt_functions) + spec->cur_mic_type = DIGITAL_MIC; + else + spec->cur_mic_type = REAR_MIC; + spec->cur_mic_boost = 0; for (i = 0; i < VNODES_COUNT; i++) { @@ -4539,6 +6851,15 @@ static void ca0132_init_chip(struct hda_codec *codec) on = (unsigned int)ca0132_effects[i].reqs[0]; spec->effects_switch[i] = on ? 1 : 0; } + /* + * Sets defaults for the effect slider controls, only for alternative + * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. + */ + if (spec->use_alt_controls) { + spec->xbass_xover_freq = 8; + for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) + spec->fx_ctl_val[i] = effect_slider_defaults[i]; + } spec->voicefx_val = 0; spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1; @@ -4549,6 +6870,120 @@ static void ca0132_init_chip(struct hda_codec *codec) #endif } +/* + * Recon3Di exit specific commands. + */ +/* prevents popping noise on shutdown */ +static void r3di_gpio_shutdown(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00); +} + +/* + * Sound Blaster Z exit specific commands. + */ +static void sbz_region2_exit(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i; + + for (i = 0; i < 4; i++) + writeb(0x0, spec->mem_base + 0x100); + for (i = 0; i < 8; i++) + writeb(0xb3, spec->mem_base + 0x304); + /* + * I believe these are GPIO, with the right most hex digit being the + * gpio pin, and the second digit being on or off. We see this more in + * the input/output select functions. + */ + writew(0x0000, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + writew(0x0007, spec->mem_base + 0x320); +} + +static void sbz_set_pin_ctl_default(struct hda_codec *codec) +{ + hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13}; + unsigned int i; + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40); + + for (i = 0; i < 5; i++) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); +} + +static void sbz_clear_unsolicited(struct hda_codec *codec) +{ + hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13}; + unsigned int i; + + for (i = 0; i < 7; i++) { + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, 0x00); + } +} + +/* On shutdown, sends commands in sets of three */ +static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir, + int mask, int data) +{ + if (dir >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, dir); + if (mask >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, mask); + + if (data >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, data); +} + +static void sbz_exit_chip(struct hda_codec *codec) +{ + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* Mess with GPIO */ + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01); + + chipio_set_stream_control(codec, 0x14, 0); + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_conn_rate(codec, 0x41, SR_192_000); + chipio_set_conn_rate(codec, 0x91, SR_192_000); + + chipio_write(codec, 0x18a020, 0x00000083); + + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06); + + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_control_param(codec, 0x0D, 0x24); + + sbz_clear_unsolicited(codec); + sbz_set_pin_ctl_default(codec); + + snd_hda_codec_write(codec, 0x0B, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + if (dspload_is_loaded(codec)) + dsp_reset(codec); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00); + + sbz_region2_exit(codec); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -4557,28 +6992,265 @@ static void ca0132_exit_chip(struct hda_codec *codec) dsp_reset(codec); } +/* + * This fixes a problem that was hard to reproduce. Very rarely, I would + * boot up, and there would be no sound, but the DSP indicated it had loaded + * properly. I did a few memory dumps to see if anything was different, and + * there were a few areas of memory uninitialized with a1a2a3a4. This function + * checks if those areas are uninitialized, and if they are, it'll attempt to + * reload the card 3 times. Usually it fixes by the second. + */ +static void sbz_dsp_startup_check(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_data_check[4]; + unsigned int cur_address = 0x390; + unsigned int i; + unsigned int failure = 0; + unsigned int reload = 3; + + if (spec->startup_check_entered) + return; + + spec->startup_check_entered = true; + + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + + codec_dbg(codec, "Startup Check: %d ", failure); + if (failure) + codec_info(codec, "DSP not initialized properly. Attempting to fix."); + /* + * While the failure condition is true, and we haven't reached our + * three reload limit, continue trying to reload the driver and + * fix the issue. + */ + while (failure && (reload != 0)) { + codec_info(codec, "Reloading... Tries left: %d", reload); + sbz_exit_chip(codec); + spec->dsp_state = DSP_DOWNLOAD_INIT; + codec->patch_ops.init(codec); + failure = 0; + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + reload--; + } + + if (!failure && reload < 3) + codec_info(codec, "DSP fixed."); + + if (!failure) + return; + + codec_info(codec, "DSP failed to initialize properly. Either try a full shutdown or a suspend to clear the internal memory."); +} + +/* + * This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add + * extra precision for decibel values. If you had the dB value in floating point + * you would take the value after the decimal point, multiply by 64, and divide + * by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to + * implement fixed point or floating point dB volumes. For now, I'll set them + * to 0 just incase a value has lingered from a boot into Windows. + */ +static void ca0132_alt_vol_setup(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00); +} + +/* + * Extra commands that don't really fit anywhere else. + */ +static void sbz_pre_dsp_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00820680, spec->mem_base + 0x01C); + writel(0x00820680, spec->mem_base + 0x01C); + + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); + + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); +} + +/* + * Extra commands that don't really fit anywhere else. + */ +static void r3di_pre_dsp_setup(struct hda_codec *codec) +{ + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x20); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x19); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x00); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x40); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); +} + + +/* + * These are sent before the DSP is downloaded. Not sure + * what they do, or if they're necessary. Could possibly + * be removed. Figure they're better to leave in. + */ +static void sbz_region2_startup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00000000, spec->mem_base + 0x400); + writel(0x00000000, spec->mem_base + 0x408); + writel(0x00000000, spec->mem_base + 0x40C); + writel(0x00880680, spec->mem_base + 0x01C); + writel(0x00000083, spec->mem_base + 0xC0C); + writel(0x00000030, spec->mem_base + 0xC00); + writel(0x00000000, spec->mem_base + 0xC04); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x000000F1, spec->mem_base + 0xC08); + writel(0x00000001, spec->mem_base + 0xC08); + writel(0x000000C7, spec->mem_base + 0xC08); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x00000080, spec->mem_base + 0xC04); +} + +/* + * Extra init functions for alternative ca0132 codecs. Done + * here so they don't clutter up the main ca0132_init function + * anymore than they have to. + */ +static void ca0132_alt_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_alt_vol_setup(codec); + + switch (spec->quirk) { + case QUIRK_SBZ: + codec_dbg(codec, "SBZ alt_init"); + ca0132_gpio_init(codec); + sbz_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->sbz_init_verbs); + break; + case QUIRK_R3DI: + codec_dbg(codec, "R3DI alt_init"); + ca0132_gpio_init(codec); + ca0132_gpio_setup(codec); + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING); + r3di_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4); + break; + } +} + static int ca0132_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i; + bool dsp_loaded; + + /* + * If the DSP is already downloaded, and init has been entered again, + * there's only two reasons for it. One, the codec has awaken from a + * suspended state, and in that case dspload_is_loaded will return + * false, and the init will be ran again. The other reason it gets + * re entered is on startup for some reason it triggers a suspend and + * resume state. In this case, it will check if the DSP is downloaded, + * and not run the init function again. For codecs using alt_functions, + * it will check if the DSP is loaded properly. + */ + if (spec->dsp_state == DSP_DOWNLOADED) { + dsp_loaded = dspload_is_loaded(codec); + if (!dsp_loaded) { + spec->dsp_reload = true; + spec->dsp_state = DSP_DOWNLOAD_INIT; + } else { + if (spec->quirk == QUIRK_SBZ) + sbz_dsp_startup_check(codec); + return 0; + } + } if (spec->dsp_state != DSP_DOWNLOAD_FAILED) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; + if (spec->quirk == QUIRK_SBZ) + sbz_region2_startup(codec); + snd_hda_power_up_pm(codec); ca0132_init_unsol(codec); - ca0132_init_params(codec); ca0132_init_flags(codec); + snd_hda_sequence_write(codec, spec->base_init_verbs); + + if (spec->use_alt_functions) + ca0132_alt_init(codec); + ca0132_download_dsp(codec); + ca0132_refresh_widget_caps(codec); - ca0132_setup_defaults(codec); - ca0132_init_analog_mic2(codec); - ca0132_init_dmic(codec); + + if (spec->quirk == QUIRK_SBZ) + writew(0x0107, spec->mem_base + 0x320); + + switch (spec->quirk) { + case QUIRK_R3DI: + r3di_setup_defaults(codec); + break; + case QUIRK_SBZ: + break; + default: + ca0132_setup_defaults(codec); + ca0132_init_analog_mic2(codec); + ca0132_init_dmic(codec); + break; + } for (i = 0; i < spec->num_outputs; i++) init_output(codec, spec->out_pins[i], spec->dacs[0]); @@ -4590,14 +7262,45 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - snd_hda_sequence_write(codec, spec->chip_init_verbs); - snd_hda_sequence_write(codec, spec->spec_init_verbs); + if (!spec->use_alt_functions) { + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); + } - ca0132_select_out(codec); - ca0132_select_mic(codec); + if (spec->quirk == QUIRK_SBZ) + ca0132_gpio_setup(codec); + + snd_hda_sequence_write(codec, spec->spec_init_verbs); + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_setup_defaults(codec); + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + default: + ca0132_select_out(codec); + ca0132_select_mic(codec); + break; + } snd_hda_jack_report_sync(codec); + /* + * Re set the PlayEnhancement switch on a resume event, because the + * controls will not be reloaded. + */ + if (spec->dsp_reload) { + spec->dsp_reload = false; + ca0132_pe_switch_set(codec); + } + snd_hda_power_down_pm(codec); return 0; @@ -4609,25 +7312,44 @@ static void ca0132_free(struct hda_codec *codec) cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_exit_chip(codec); + break; + case QUIRK_R3DI: + r3di_gpio_shutdown(codec); + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + break; + default: + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + break; + } snd_hda_power_down(codec); + if (spec->mem_base) + iounmap(spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); } +static void ca0132_reboot_notify(struct hda_codec *codec) +{ + codec->patch_ops.free(codec); +} + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, + .reboot_notify = ca0132_reboot_notify, }; static void ca0132_config(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; spec->dacs[0] = 0x2; spec->dacs[1] = 0x3; @@ -4635,9 +7357,14 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - spec->multiout.max_channels = 2; - if (spec->quirk == QUIRK_ALIENWARE) { + if (!spec->use_alt_functions) + spec->multiout.max_channels = 2; + else + spec->multiout.max_channels = 6; + + switch (spec->quirk) { + case QUIRK_ALIENWARE: codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); snd_hda_apply_pincfgs(codec, alienware_pincfgs); @@ -4657,7 +7384,63 @@ static void ca0132_config(struct hda_codec *codec) spec->input_pins[2] = 0x13; spec->shared_mic_nid = 0x7; spec->unsol_tag_amic1 = 0x11; - } else { + break; + case QUIRK_SBZ: + codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x7; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + spec->dig_in = 0x09; + break; + case QUIRK_R3DI: + codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3di_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x07; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0x0a; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + break; + default: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ @@ -4678,54 +7461,44 @@ static void ca0132_config(struct hda_codec *codec) /* SPDIF I/O */ spec->dig_out = 0x05; spec->multiout.dig_out_nid = spec->dig_out; - cfg->dig_out_pins[0] = 0x0c; - cfg->dig_outs = 1; - cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; spec->dig_in = 0x09; - cfg->dig_in_pin = 0x0e; - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; } } static int ca0132_prepare_verbs(struct hda_codec *codec) { /* Verbs + terminator (an empty element) */ -#define NUM_SPEC_VERBS 4 +#define NUM_SPEC_VERBS 2 struct ca0132_spec *spec = codec->spec; spec->chip_init_verbs = ca0132_init_verbs0; - spec->spec_init_verbs = kzalloc(sizeof(struct hda_verb) * NUM_SPEC_VERBS, GFP_KERNEL); + if (spec->quirk == QUIRK_SBZ) + spec->sbz_init_verbs = sbz_init_verbs; + spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, + sizeof(struct hda_verb), + GFP_KERNEL); if (!spec->spec_init_verbs) return -ENOMEM; - /* HP jack autodetection */ - spec->spec_init_verbs[0].nid = spec->unsol_tag_hp; - spec->spec_init_verbs[0].param = AC_VERB_SET_UNSOLICITED_ENABLE; - spec->spec_init_verbs[0].verb = AC_USRSP_EN | spec->unsol_tag_hp; - - /* MIC1 jack autodetection */ - spec->spec_init_verbs[1].nid = spec->unsol_tag_amic1; - spec->spec_init_verbs[1].param = AC_VERB_SET_UNSOLICITED_ENABLE; - spec->spec_init_verbs[1].verb = AC_USRSP_EN | spec->unsol_tag_amic1; - /* config EAPD */ - spec->spec_init_verbs[2].nid = 0x0b; - spec->spec_init_verbs[2].param = 0x78D; - spec->spec_init_verbs[2].verb = 0x00; + spec->spec_init_verbs[0].nid = 0x0b; + spec->spec_init_verbs[0].param = 0x78D; + spec->spec_init_verbs[0].verb = 0x00; /* Previously commented configuration */ /* - spec->spec_init_verbs[3].nid = 0x0b; - spec->spec_init_verbs[3].param = AC_VERB_SET_EAPD_BTLENABLE; + spec->spec_init_verbs[2].nid = 0x0b; + spec->spec_init_verbs[2].param = AC_VERB_SET_EAPD_BTLENABLE; + spec->spec_init_verbs[2].verb = 0x02; + + spec->spec_init_verbs[3].nid = 0x10; + spec->spec_init_verbs[3].param = 0x78D; spec->spec_init_verbs[3].verb = 0x02; spec->spec_init_verbs[4].nid = 0x10; - spec->spec_init_verbs[4].param = 0x78D; + spec->spec_init_verbs[4].param = AC_VERB_SET_EAPD_BTLENABLE; spec->spec_init_verbs[4].verb = 0x02; - - spec->spec_init_verbs[5].nid = 0x10; - spec->spec_init_verbs[5].param = AC_VERB_SET_EAPD_BTLENABLE; - spec->spec_init_verbs[5].verb = 0x02; */ /* Terminator: spec->spec_init_verbs[NUM_SPEC_VERBS-1] */ @@ -4757,9 +7530,46 @@ static int patch_ca0132(struct hda_codec *codec) else spec->quirk = QUIRK_NONE; + /* Setup BAR Region 2 for Sound Blaster Z */ + if (spec->quirk == QUIRK_SBZ) { + spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); + if (spec->mem_base == NULL) { + codec_warn(codec, "pci_iomap failed!"); + codec_info(codec, "perhaps this is not an SBZ?"); + spec->quirk = QUIRK_NONE; + } + } + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; - spec->mixers[0] = ca0132_mixer; + + /* Set which mixers each quirk uses. */ + switch (spec->quirk) { + case QUIRK_SBZ: + spec->mixers[0] = sbz_mixer; + snd_hda_codec_set_name(codec, "Sound Blaster Z"); + break; + case QUIRK_R3DI: + spec->mixers[0] = r3di_mixer; + snd_hda_codec_set_name(codec, "Recon3Di"); + break; + default: + spec->mixers[0] = ca0132_mixer; + break; + } + + /* Setup whether or not to use alt functions/controls */ + switch (spec->quirk) { + case QUIRK_SBZ: + case QUIRK_R3DI: + spec->use_alt_controls = true; + spec->use_alt_functions = true; + break; + default: + spec->use_alt_controls = false; + spec->use_alt_functions = false; + break; + } spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5b4dbcec6de8..e7fcfc3b8885 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -588,6 +588,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct conexant_spec *spec = codec->spec; + struct snd_kcontrol_new *kctl; int i; if (action != HDA_FIXUP_ACT_PROBE) @@ -606,9 +607,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x1a, PIN_VREF50); /* override mic boost control */ - for (i = 0; i < spec->gen.kctls.used; i++) { - struct snd_kcontrol_new *kctl = - snd_array_elem(&spec->gen.kctls, i); + snd_array_for_each(&spec->gen.kctls, i, kctl) { if (!strcmp(kctl->name, "Mic Boost Volume")) { kctl->put = olpc_xo_mic_boost_put; break; @@ -959,12 +958,15 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -998,6 +1000,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, + { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7d7eb1354eee..1de5491fb9bf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -33,6 +33,7 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/asoundef.h> @@ -176,13 +177,13 @@ struct hdmi_spec { /* i915/powerwell (Haswell+/Valleyview+) specific */ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ - struct i915_audio_component_audio_ops i915_audio_ops; + struct drm_audio_component_audio_ops drm_audio_ops; struct hdac_chmap chmap; hda_nid_t vendor_nid; }; -#ifdef CONFIG_SND_HDA_I915 +#ifdef CONFIG_SND_HDA_COMPONENT static inline bool codec_has_acomp(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -510,7 +511,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) snd_info_set_text_ops(entry, per_pin, print_eld_info); entry->c.text.write = write_eld_info; - entry->mode |= S_IWUSR; + entry->mode |= 0200; per_pin->proc_entry = entry; return 0; @@ -764,8 +765,10 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) snd_hda_jack_report_sync(codec); + mutex_unlock(&spec->pcm_lock); } static void jack_callback(struct hda_codec *codec, @@ -1628,21 +1631,23 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - struct hdmi_spec *spec = codec->spec; int ret; /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) - snd_hda_power_up_pm(codec); + if (!codec_has_acomp(codec)) { + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { + snd_hda_power_down_pm(codec); + return false; + } + } - mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { sync_eld_via_acomp(codec, per_pin); ret = false; /* don't call snd_hda_jack_report_sync() */ } else { ret = hdmi_present_sense_via_verbs(per_pin, repoll); } - mutex_unlock(&spec->pcm_lock); if (!codec_has_acomp(codec)) snd_hda_power_down_pm(codec); @@ -1654,12 +1659,16 @@ static void hdmi_repoll_eld(struct work_struct *work) { struct hdmi_spec_per_pin *per_pin = container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); + struct hda_codec *codec = per_pin->codec; + struct hdmi_spec *spec = codec->spec; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(per_pin, per_pin->repoll_count)) snd_hda_jack_report_sync(per_pin->codec); + mutex_unlock(&spec->pcm_lock); } static void intel_haswell_fixup_connect_list(struct hda_codec *codec, @@ -2279,7 +2288,7 @@ static void generic_hdmi_free(struct hda_codec *codec) int pin_idx, pcm_idx; if (codec_has_acomp(codec)) - snd_hdac_i915_register_notifier(NULL); + snd_hdac_acomp_register_notifier(&codec->bus->core, NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2462,6 +2471,38 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: + * Pin Widget 5 - PORT B (port = 1 in i915 driver) + * Pin Widget 6 - PORT C (port = 2 in i915 driver) + * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) + */ +static int intel_base_nid(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + return 4; + default: + return 5; + } +} + +static int intel_pin2port(void *audio_ptr, int pin_nid) +{ + int base_nid = intel_base_nid(audio_ptr); + + if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) + return -1; + return pin_nid - base_nid + 1; /* intel port is 1-based */ +} + static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; @@ -2472,16 +2513,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (port < 1 || port > 3) return; - switch (codec->core.vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - pin_nid = port + 0x03; - break; - default: - pin_nid = port + 0x04; - break; - } + pin_nid = port + intel_base_nid(codec) - 1; /* intel port is 1-based */ /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume @@ -2502,14 +2534,16 @@ static void register_i915_notifier(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; spec->use_acomp_notifier = true; - spec->i915_audio_ops.audio_ptr = codec; + spec->drm_audio_ops.audio_ptr = codec; /* intel_audio_codec_enable() or intel_audio_codec_disable() * will call pin_eld_notify with using audio_ptr pointer * We need make sure audio_ptr is really setup */ wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + spec->drm_audio_ops.pin2port = intel_pin2port; + spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_acomp_register_notifier(&codec->bus->core, + &spec->drm_audio_ops); } /* setup_stream ops override for HSW+ */ @@ -3741,6 +3775,11 @@ static int patch_atihdmi(struct hda_codec *codec) spec->chmap.channels_max = max(spec->chmap.channels_max, 8u); + /* AMD GPUs have neither EPSS nor CLKSTOP bits, hence preventing + * the link-down as is. Tell the core to allow it. + */ + codec->link_down_at_suspend = 1; + return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01a6643fc7d4..7496be4491b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -793,6 +793,9 @@ static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + if (!snd_hda_get_bool_hint(codec, "shutup")) + return; /* disabled explicitly by hints */ + if (spec && spec->shutup) spec->shutup(codec); else @@ -2542,6 +2545,7 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu Lifebook S7110", ALC262_FIXUP_FSC_S7110), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), + SND_PCI_QUIRK(0x1734, 0x1141, "FSC ESPRIMO U9210", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), @@ -2830,6 +2834,7 @@ static int find_ext_mic_pin(struct hda_codec *codec); static void alc286_shutup(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; int mic_pin = find_ext_mic_pin(codec); /* don't shut up pins when unloading the driver; otherwise it breaks @@ -2837,8 +2842,7 @@ static void alc286_shutup(struct hda_codec *codec) */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ if (pin->nid != mic_pin) snd_hda_codec_read(codec, pin->nid, 0, @@ -3653,30 +3657,37 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, } } -static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc269_fixup_hp_mute_led_micx(struct hda_codec *codec, + const struct hda_fixup *fix, + int action, hda_nid_t pin) { struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x18; + spec->mute_led_nid = pin; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; codec->power_filter = led_power_filter; } } +static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x18); +} + static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x19; - spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->gen.vmaster_mute_enum = 1; - codec->power_filter = led_power_filter; - } + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x19); +} + +static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1b); } /* update LED status via GPIO */ @@ -4985,7 +4996,6 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->shutup = alc_no_shutup; /* reduce click noise */ spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ @@ -5384,9 +5394,19 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" +static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_no_shutup(codec, fix, action); /* reduce click noise */ + hda_fixup_thinkpad_acpi(codec, fix, action); +} + /* for dell wmi mic mute led */ #include "dell_wmi_helper.c" +/* for alc295_fixup_hp_top_speakers */ +#include "hp_x360_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5413,6 +5433,7 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_MUTE_LED_MIC3, ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_HP_GPIO_MIC1_LED, ALC269_FIXUP_HP_LINE1_MIC1_LED, @@ -5506,6 +5527,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC295_FIXUP_HP_X360, }; static const struct hda_fixup alc269_fixups[] = { @@ -5672,6 +5694,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic3, + }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, @@ -5927,7 +5953,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC269_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, - .v.func = hda_fixup_thinkpad_acpi, + .v.func = alc_fixup_thinkpad_acpi, .chained = true, .chain_id = ALC269_FIXUP_SKU_IGNORE, }, @@ -6375,6 +6401,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC295_FIXUP_HP_X360] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_hp_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HP_MUTE_LED_MIC3 + } }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6494,6 +6526,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), + SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -6577,10 +6610,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3138, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), @@ -6752,6 +6784,22 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01111010}, {0x1e, 0x01451130}, {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + {0x12, 0x90a60140}, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x1a, 0x02a11040}, + {0x1b, 0x01014020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11020}, + {0x1a, 0x02a11030}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 5101f40f6fbd..93b8cfc6636f 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -662,7 +662,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 8dabd4d0211d..d7366ade5a25 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -926,7 +926,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index c241dc06dd92..904e3c4f4dfe 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -214,7 +214,7 @@ void lola_proc_debug_new(struct lola *chip) snd_info_set_text_ops(entry, chip, lola_proc_codec_read); if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = lola_proc_codec_rw_write; } if (!snd_card_proc_new(chip->card, "regs", &entry)) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 9655b08a1c52..54f6252faca6 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1016,6 +1016,11 @@ static int snd_lx6464es_create(struct snd_card *card, /* dsp port */ chip->port_dsp_bar = pci_ioremap_bar(pci, 2); + if (!chip->port_dsp_bar) { + dev_err(card->dev, "cannot remap PCI memory region\n"); + err = -ENOMEM; + goto remap_pci_failed; + } err = request_threaded_irq(pci->irq, lx_interrupt, lx_threaded_irq, IRQF_SHARED, KBUILD_MODNAME, chip); @@ -1055,6 +1060,9 @@ device_new_failed: free_irq(pci->irq, chip); request_irq_failed: + iounmap(chip->port_dsp_bar); + +remap_pci_failed: pci_release_regions(pci); request_regions_failed: diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 8f20dec97843..224e942f556d 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2657,7 +2657,10 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, chip->irq = pci->irq; #ifdef CONFIG_PM_SLEEP - chip->suspend_mem = vmalloc(sizeof(u16) * (REV_B_CODE_MEMORY_LENGTH + REV_B_DATA_MEMORY_LENGTH)); + chip->suspend_mem = + vmalloc(array_size(sizeof(u16), + REV_B_CODE_MEMORY_LENGTH + + REV_B_DATA_MEMORY_LENGTH)); if (chip->suspend_mem == NULL) dev_warn(card->dev, "can't allocate apm buffer\n"); #endif diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca12665ff73..81af21ac1439 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -1052,10 +1052,10 @@ static int add_controls(struct oxygen *chip, [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; - unsigned int i, j; + unsigned int i; struct snd_kcontrol_new template; struct snd_kcontrol *ctl; - int err; + int j, err; for (i = 0; i < count; ++i) { template = controls[i]; @@ -1086,11 +1086,11 @@ static int add_controls(struct oxygen *chip, err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - for (j = 0; j < CONTROL_COUNT; ++j) - if (!strcmp(ctl->id.name, known_ctl_names[j])) { - chip->controls[j] = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + j = match_string(known_ctl_names, CONTROL_COUNT, ctl->id.name); + if (j >= 0) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return 0; } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index f9ae72f28ddc..e57da4036231 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1465,7 +1465,7 @@ static void pcxhr_proc_init(struct snd_pcxhr *chip) !snd_card_proc_new(chip->card, "gpio", &entry)) { snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); entry->c.text.write = pcxhr_proc_gpo_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (!snd_card_proc_new(chip->card, "ltc", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a8abb15e3c3a..7fbdb703bfcd 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1188,6 +1188,7 @@ SONICVIBES_SINGLE("Joystick Speed", 0, SV_IREG_GAME_PORT, 1, 15, 0); static int snd_sonicvibes_create_gameport(struct sonicvibes *sonic) { struct gameport *gp; + int err; sonic->gameport = gp = gameport_allocate_port(); if (!gp) { @@ -1203,7 +1204,10 @@ static int snd_sonicvibes_create_gameport(struct sonicvibes *sonic) gameport_register_port(gp); - snd_ctl_add(sonic->card, snd_ctl_new1(&snd_sonicvibes_game_control, sonic)); + err = snd_ctl_add(sonic->card, + snd_ctl_new1(&snd_sonicvibes_game_control, sonic)); + if (err < 0) + return err; return 0; } @@ -1515,7 +1519,11 @@ static int snd_sonic_probe(struct pci_dev *pci, return err; } - snd_sonicvibes_create_gameport(sonic); + err = snd_sonicvibes_create_gameport(sonic); + if (err < 0) { + snd_card_free(card); + return err; + } if ((err = snd_card_register(card)) < 0) { snd_card_free(card); diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index eabd84d9ffee..49c64fae3466 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3362,7 +3362,9 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) trident->tlb.entries = (unsigned int*)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4); trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4); /* allocate shadow TLB page table (virtual addresses) */ - trident->tlb.shadow_entries = vmalloc(SNDRV_TRIDENT_MAX_PAGES*sizeof(unsigned long)); + trident->tlb.shadow_entries = + vmalloc(array_size(SNDRV_TRIDENT_MAX_PAGES, + sizeof(unsigned long))); if (!trident->tlb.shadow_entries) return -ENOMEM; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 3a1c0b8b4ea2..c488c5afa195 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -439,7 +439,9 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -ENOMEM; } if (! dev->idx_table) { - dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL); + dev->idx_table = kmalloc_array(VIA_TABLE_SIZE, + sizeof(*dev->idx_table), + GFP_KERNEL); if (! dev->idx_table) return -ENOMEM; } diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 8a69221c1b86..b13c8688cc8d 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -292,7 +292,9 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -ENOMEM; } if (! dev->idx_table) { - dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL); + dev->idx_table = kmalloc_array(VIA_TABLE_SIZE, + sizeof(*dev->idx_table), + GFP_KERNEL); if (! dev->idx_table) return -ENOMEM; } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 8ca2e41e5827..6f81396aadc9 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2435,8 +2435,8 @@ int snd_ymfpci_create(struct snd_card *card, goto free_chip; #ifdef CONFIG_PM_SLEEP - chip->saved_regs = kmalloc(YDSXGR_NUM_SAVED_REGS * sizeof(u32), - GFP_KERNEL); + chip->saved_regs = kmalloc_array(YDSXGR_NUM_SAVED_REGS, sizeof(u32), + GFP_KERNEL); if (chip->saved_regs == NULL) { err = -ENOMEM; goto free_chip; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 41af6b9cc350..1cf11cf51e1d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" +source "sound/soc/meson/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/qcom/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 06389a5385d7..62a5f87c3cfc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ +obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4..58c1dcb4d255 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c..8e3275a96a82 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -148,7 +150,8 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -163,7 +166,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -172,13 +175,24 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } -static int cz_dmic_startup(struct snd_pcm_substream *substream) +static int cz_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->i2s_instance = I2S_BT_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_dmic1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -197,23 +211,39 @@ static const struct snd_soc_ops cz_max_play_ops = { .shutdown = cz_max_shutdown, }; -static const struct snd_soc_ops cz_dmic_cap_ops = { - .startup = cz_dmic_startup, +static const struct snd_soc_ops cz_dmic0_cap_ops = { + .startup = cz_dmic0_startup, + .shutdown = cz_dmic_shutdown, +}; + +static const struct snd_soc_ops cz_dmic1_cap_ops = { + .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play-cap", - .stream_name = "Playback and Capture", + .name = "amd-da7219-play", + .stream_name = "Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.3.auto", + .cpu_dai_name = "designware-i2s.1.auto", .codec_dai_name = "da7219-hifi", .codec_name = "i2c-DLGS7219:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .ops = &cz_da7219_cap_ops, + }, + { + .name = "amd-da7219-cap", + .stream_name = "Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "da7219-hifi", + .codec_name = "i2c-DLGS7219:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, @@ -221,7 +251,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.1.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "HiFi", .codec_name = "MX98357A:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -230,8 +260,22 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .ops = &cz_max_play_ops, }, { - .name = "dmic", - .stream_name = "DMIC Capture", + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.3.auto", + .codec_dai_name = "adau7002-hifi", + .codec_name = "ADAU7002:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_dmic0_cap_ops, + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.2.auto", .codec_dai_name = "adau7002-hifi", @@ -239,7 +283,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, - .ops = &cz_dmic_cap_ops, + .ops = &cz_dmic1_cap_ops, }, }; @@ -278,11 +322,52 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 1458b5048498..e359938e3d7e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) | (size / 2); } @@ -322,22 +320,87 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } +static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg, res_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + ch_reg = mmACP_I2SMICSP_RER1; + res_reg = mmACP_I2SMICSP_RCR1; + imr_reg = mmACP_I2SMICSP_IMR1; + break; + case CAP_CHANNEL0: + default: + ch_reg = mmACP_I2SMICSP_RER0; + res_reg = mmACP_I2SMICSP_RCR0; + imr_reg = mmACP_I2SMICSP_IMR0; + break; + } + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, ch_reg); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, res_reg); + } + val = acp_reg_read(acp_mmio, imr_reg); + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x1, acp_mmio, ch_reg); +} + +static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + imr_reg = mmACP_I2SMICSP_IMR1; + ch_reg = mmACP_I2SMICSP_RER1; + break; + case CAP_CHANNEL0: + default: + imr_reg = mmACP_I2SMICSP_IMR0; + ch_reg = mmACP_I2SMICSP_RER0; + break; + } + val = acp_reg_read(acp_mmio, imr_reg); + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x0, acp_mmio, ch_reg); +} + /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -356,10 +419,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: - case ACP_TO_SYSRAM_CH_NUM: case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: - case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; @@ -368,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -613,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -644,32 +709,39 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & - BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if (valid_irq) return IRQ_HANDLED; else @@ -773,8 +845,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - if (pinfo) + if (pinfo) { rtd->i2s_instance = pinfo->i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); @@ -842,8 +916,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS; rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; @@ -852,13 +926,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; rtd->byte_cnt_low_reg_offset = mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: default: rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_CH_NUM; switch (adata->asic_type) { case CHIP_STONEY: rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET; @@ -875,6 +950,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; rtd->byte_cnt_low_reg_offset = mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } } @@ -928,6 +1004,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 buffersize; u32 pos = 0; u64 bytescount = 0; + u16 dscr; + u32 period_bytes, delay; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -935,12 +1013,25 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + period_bytes = frames_to_bytes(runtime, runtime->period_size); + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } else { + buffersize = frames_to_bytes(runtime, runtime->buffer_size); + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + pos = do_div(bytescount, buffersize); + } return bytes_to_frames(runtime, pos); } @@ -954,16 +1045,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; @@ -972,7 +1071,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -983,37 +1081,32 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); + rtd->bytescount = acp_get_byte_count(rtd); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (rtd->capture_channel == CAP_CHANNEL0) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL1); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL0); + } + if (rtd->capture_channel == CAP_CHANNEL1) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL0); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL1); + } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); } else { - acp_dma_start(rtd->acp_mmio, rtd->ch2); - acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } - rtd->bytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96c84d4..be3963e8f4fa 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,8 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define CAP_CHANNEL0 0x00 +#define CAP_CHANNEL1 0x01 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -72,16 +74,16 @@ #define ACP_TO_I2S_DMA_CH_NUM 13 /* Capture DMA channels */ -#define ACP_TO_SYSRAM_CH_NUM 14 -#define I2S_TO_ACP_DMA_CH_NUM 15 +#define I2S_TO_ACP_DMA_CH_NUM 14 +#define ACP_TO_SYSRAM_CH_NUM 15 /* Playback DMA Channels for I2S BT instance */ #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 /* Capture DMA Channels for I2S BT Instance */ -#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 -#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10 +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11 #define NUM_DSCRS_PER_CHANNEL 2 @@ -125,6 +127,7 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 i2s_instance; + u16 capture_channel; u16 direction; u16 ch1; u16 ch2; @@ -135,6 +138,7 @@ struct audio_substream_data { u32 sram_bank; u32 byte_cnt_high_reg_offset; u32 byte_cnt_low_reg_offset; + u32 dma_curr_dscr; uint64_t size; u64 bytescount; void __iomem *acp_mmio; @@ -155,6 +159,7 @@ struct audio_drv_data { */ struct acp_platform_info { u16 i2s_instance; + u16 capture_channel; }; union acp_dma_count { diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 5d3b5af9fd92..d88c1d995036 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -206,7 +206,6 @@ struct atmel_i2s_dev { struct regmap *regmap; struct clk *pclk; struct clk *gclk; - struct clk *aclk; struct snd_dmaengine_dai_dma_data playback; struct snd_dmaengine_dai_dma_data capture; unsigned int fmt; @@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs) { int i, best; - if (!dev->gclk || !dev->aclk) { + if (!dev->gclk) { dev_err(dev->dev, "cannot generate the I2S Master Clock\n"); return -EINVAL; } @@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, bool enabled) { unsigned int mr, mr_mask; - unsigned long aclk_rate; + unsigned long gclk_rate; int ret; mr = 0; @@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, /* Disable/unprepare the PMC generated clock. */ clk_disable_unprepare(dev->gclk); - /* Disable/unprepare the PLL audio clock. */ - clk_disable_unprepare(dev->aclk); return 0; } if (!dev->gck_param) return -EINVAL; - aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); + gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); - /* Fist change the PLL audio clock frequency ... */ - ret = clk_set_rate(dev->aclk, aclk_rate); + ret = clk_set_rate(dev->gclk, gclk_rate); if (ret) return ret; - /* - * ... then set the PMC generated clock rate to the very same frequency - * to set the gclk parent to aclk. - */ - ret = clk_set_rate(dev->gclk, aclk_rate); - if (ret) - return ret; - - /* Prepare and enable the PLL audio clock first ... */ - ret = clk_prepare_enable(dev->aclk); - if (ret) - return ret; - - /* ... then prepare and enable the PMC generated clock. */ ret = clk_prepare_enable(dev->gclk); if (ret) return ret; @@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev) return err; } - /* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */ - dev->aclk = devm_clk_get(&pdev->dev, "aclk"); + /* Get audio clock to generate the I2S Master Clock (I2S_MCK) */ dev->gclk = devm_clk_get(&pdev->dev, "gclk"); - if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) { - if (PTR_ERR(dev->aclk) == -EPROBE_DEFER || - PTR_ERR(dev->gclk) == -EPROBE_DEFER) + if (IS_ERR(dev->gclk)) { + if (PTR_ERR(dev->gclk) == -EPROBE_DEFER) return -EPROBE_DEFER; /* Master Mode not supported */ - dev->aclk = NULL; dev->gclk = NULL; - } else if (IS_ERR(dev->gclk)) { - err = PTR_ERR(dev->gclk); - dev_err(&pdev->dev, - "failed to get the PMC generated clock: %d\n", err); - return err; - } else if (IS_ERR(dev->aclk)) { - err = PTR_ERR(dev->aclk); - dev_err(&pdev->dev, - "failed to get the PLL audio clock: %d\n", err); - return err; } - dev->dev = &pdev->dev; dev->regmap = regmap; platform_set_drvdata(pdev, dev); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index fb650659c3a3..a906560d0cdd 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -339,8 +339,8 @@ static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct au1xpsc_audio_dmadata *dmadata; - dmadata = devm_kzalloc(&pdev->dev, - 2 * sizeof(struct au1xpsc_audio_dmadata), + dmadata = devm_kcalloc(&pdev->dev, + 2, sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); if (!dmadata) return -ENOMEM; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63cf62e9c9aa..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,12 +74,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DIO2125 select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 + select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI select SND_SOC_ICS43432 @@ -141,8 +141,10 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER + select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIMPLE_AMPLIFIER select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF select SND_SOC_SSM2305 @@ -572,10 +574,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DIO2125 - tristate "Dioo DIO2125 Amplifier" - select GPIOLIB - config SND_SOC_DMIC tristate @@ -588,6 +586,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES7241 + tristate "Everest Semi ES7241 CODEC" + config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C @@ -778,6 +779,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default y if SND_SOC_RT5682=y default y if SND_SOC_RT1305=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m @@ -791,6 +793,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m + default m if SND_SOC_RT5682=m default m if SND_SOC_RT1305=m config SND_SOC_RL6347A @@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI tristate default SND_SOC_RT5677 && SPI +config SND_SOC_RT5682 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" @@ -891,6 +897,10 @@ config SND_SOC_SIGMADSP_REGMAP tristate select SND_SOC_SIGMADSP +config SND_SOC_SIMPLE_AMPLIFIER + tristate "Simple Audio Amplifier" + select GPIOLIB + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO @@ -953,8 +963,11 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + tristate "Texas Instruments TAS571x power amplifiers" depends on I2C + help + Enable support for Texas Instruments TAS5707, TAS5711, TAS5717, + TAS5719 and TAS5721 power amplifiers config SND_SOC_TAS5720 tristate "Texas Instruments TAS5720 Mono Audio amplifier" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e023fdf85221..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -146,6 +147,7 @@ snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o +snd-soc-rt5682-objs := rt5682.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -249,9 +251,9 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp -snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o @@ -329,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o @@ -405,6 +408,7 @@ obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o +obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o @@ -507,7 +511,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp -obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o +obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index ae41edd1c406..57169b8ff14e 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { { "DSP", NULL, "Left Decimator" }, { "DSP", NULL, "Right Decimator" }, + { "DSP", NULL, "Playback" }, }; static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 31ec0ba2e639..299ada4dfaa0 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ -struct snd_soc_component_driver soc_codec_dev_ak4458 = { +static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { .probe = ak4458_probe, .remove = ak4458_remove, .controls = ak4458_snd_controls, diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee13406d93..2fa83a1a84cf 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> #include <linux/module.h> #include <sound/soc.h> @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9351cf..c1181a20714d 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include <linux/clk.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 605055964529..353237025514 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index f4ed5cc40661..448bb90c9c8e 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return regcache_sync(ak5558->regmap); } -const struct dev_pm_ops ak5558_pm = { +static const struct dev_pm_ops ak5558_pm = { SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; -struct snd_soc_component_driver soc_codec_dev_ak5558 = { +static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .probe = ak5558_probe, .remove = ak5558_remove, .controls = ak5558_snd_controls, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2a7a4168c072..3c266eeb89bf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) - return 1; + return true; return reg == CS4270_CHIPID; } diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index feca0a672976..80dc42197154 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1733,10 +1733,10 @@ static ssize_t cs43130_show_ac_r(struct device *dev, return cs43130_show_ac(dev, buf, HP_RIGHT); } -static DEVICE_ATTR(hpload_dc_l, S_IRUGO, cs43130_show_dc_l, NULL); -static DEVICE_ATTR(hpload_dc_r, S_IRUGO, cs43130_show_dc_r, NULL); -static DEVICE_ATTR(hpload_ac_l, S_IRUGO, cs43130_show_ac_l, NULL); -static DEVICE_ATTR(hpload_ac_r, S_IRUGO, cs43130_show_ac_r, NULL); +static DEVICE_ATTR(hpload_dc_l, 0444, cs43130_show_dc_l, NULL); +static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); +static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); +static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0da52ead91e0..45e50fe3bf25 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 07dd33b09596..ab174b5114dc 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component) return -ENOMEM; cx20442->por = regulator_get(component->dev, "POR"); - if (IS_ERR(cx20442->por)) - dev_warn(component->dev, "failed to get the regulator"); + if (IS_ERR(cx20442->por)) { + int err = PTR_ERR(cx20442->por); + + dev_warn(component->dev, "failed to get POR supply (%d)", err); + /* + * When running on a non-dt platform and requested regulator + * is not available, regulator_get() never returns + * -EPROBE_DEFER as it is not able to justify if the regulator + * may still appear later. On the other hand, the board can + * still set full constraints flag at late_initcall in order + * to instruct regulator_get() to return a dummy one if + * sufficient. Hence, if we get -ENODEV here, let's convert + * it to -EPROBE_DEFER and wait for the board to decide or + * let Deferred Probe infrastructure handle this error. + */ + if (err == -ENODEV) + err = -EPROBE_DEFER; + kfree(cx20442); + return err; + } + cx20442->tty = NULL; snd_soc_component_set_drvdata(component, cx20442); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111b7184..e172913d04a4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen <Dajun.chen@diasemi.com> - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen <Dajun.chen@diasemi.com> +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include <linux/delay.h> #include <linux/i2c.h> diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 54cb5f24969f..92d006a5283e 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg) case DA7213_ALC_OFFSET_AUTO_M_R: case DA7213_ALC_OFFSET_AUTO_U_R: case DA7213_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d..c0144f2f8174 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2143,9 +2143,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg) case DA7219_ACCDET_IRQ_EVENT_B: case DA7219_ACCDET_CONFIG_8: case DA7219_SYSTEM_STATUS: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index afdf90c78884..f6a7bf9560e7 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev, case DA9055_HP_R_GAIN_STATUS: case DA9055_LINE_GAIN_STATUS: case DA9055_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 58515bb1a303..6d7bca7b78ca 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -17,6 +17,7 @@ * in the file called COPYING. */ +#include <linux/of_platform.h> #include <linux/module.h> #include <sound/soc.h> @@ -24,6 +25,82 @@ * The everest 7134 is a very simple DA converter with no register */ +struct es7134_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *mclk_fs; + unsigned int mclk_fs_num; +}; + +struct es7134_chip { + struct snd_soc_dai_driver *dai_drv; + const struct es7134_clock_mode *modes; + unsigned int mode_num; + const struct snd_soc_dapm_widget *extra_widgets; + unsigned int extra_widget_num; + const struct snd_soc_dapm_route *extra_routes; + unsigned int extra_route_num; +}; + +struct es7134_data { + unsigned int mclk; + const struct es7134_chip *chip; +}; + +static int es7134_check_mclk(struct snd_soc_dai *dai, + struct es7134_data *priv, + unsigned int rate) +{ + unsigned int mfs = priv->mclk / rate; + int i, j; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7134_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate > mode->rate_max) + continue; + + for (j = 0; j < mode->mclk_fs_num; j++) { + if (mode->mclk_fs[j] == mfs) + return 0; + } + + dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n", + mfs, rate); + return -EINVAL; + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7134_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + /* mclk has not been provided, assume it is OK */ + if (!priv->mclk) + return 0; + + return es7134_check_mclk(dai, priv, params_rate(params)); +} + +static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | @@ -38,8 +115,38 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int es7134_component_probe(struct snd_soc_component *c) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c); + struct es7134_data *priv = snd_soc_component_get_drvdata(c); + const struct es7134_chip *chip = priv->chip; + int ret; + + if (chip->extra_widget_num) { + ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets, + chip->extra_widget_num); + if (ret) { + dev_err(c->dev, "failed to add extra widgets\n"); + return ret; + } + } + + if (chip->extra_route_num) { + ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes, + chip->extra_route_num); + if (ret) { + dev_err(c->dev, "failed to add extra routes\n"); + return ret; + } + } + + return 0; +} + static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, + .hw_params = es7134_hw_params, + .set_sysclk = es7134_set_sysclk, }; static struct snd_soc_dai_driver es7134_dai = { @@ -48,7 +155,11 @@ static struct snd_soc_dai_driver es7134_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | @@ -58,18 +169,56 @@ static struct snd_soc_dai_driver es7134_dai = { .ops = &es7134_dai_ops, }; +static const struct es7134_clock_mode es7134_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .mclk_fs_num = 5, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 }, + .mclk_fs_num = 5, + }, { + /* Quad speed mode */ + .rate_min = 167000, + .rate_max = 192000, + .mclk_fs = (unsigned int[]) { 128, 192, 256 }, + .mclk_fs_num = 3, + }, +}; + +/* Digital I/O are also supplied by VDD on the es7134 */ +static const struct snd_soc_dapm_route es7134_extra_routes[] = { + { "Playback", NULL, "VDD", } +}; + +static const struct es7134_chip es7134_chip = { + .dai_drv = &es7134_dai, + .modes = es7134_modes, + .mode_num = ARRAY_SIZE(es7134_modes), + .extra_routes = es7134_extra_routes, + .extra_route_num = ARRAY_SIZE(es7134_extra_routes), +}; + static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, + { "DAC", NULL, "VDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { + .probe = es7134_component_probe, .dapm_widgets = es7134_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets), .dapm_routes = es7134_dapm_routes, @@ -80,17 +229,87 @@ static const struct snd_soc_component_driver es7134_component_driver = { .non_legacy_dai_naming = 1, }; +static struct snd_soc_dai_driver es7154_dai = { + .name = "es7154-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S18_3LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7134_dai_ops, +}; + +static const struct es7134_clock_mode es7154_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256, + 384, 512, 768, 1024 }, + .mclk_fs_num = 9, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512, + 768, 1024}, + .mclk_fs_num = 7, + } +}; + +/* Es7154 has a separate supply for digital I/O */ +static const struct snd_soc_dapm_widget es7154_extra_widgets[] = { + SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0), +}; + +static const struct snd_soc_dapm_route es7154_extra_routes[] = { + { "Playback", NULL, "PVDD", } +}; + +static const struct es7134_chip es7154_chip = { + .dai_drv = &es7154_dai, + .modes = es7154_modes, + .mode_num = ARRAY_SIZE(es7154_modes), + .extra_routes = es7154_extra_routes, + .extra_route_num = ARRAY_SIZE(es7154_extra_routes), + .extra_widgets = es7154_extra_widgets, + .extra_widget_num = ARRAY_SIZE(es7154_extra_widgets), +}; + static int es7134_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct es7134_data *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, - &es7134_dai, 1); + priv->chip->dai_drv, 1); } #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { - { .compatible = "everest,es7134", }, - { .compatible = "everest,es7144", }, + { .compatible = "everest,es7134", .data = &es7134_chip }, + { .compatible = "everest,es7144", .data = &es7134_chip }, + { .compatible = "everest,es7154", .data = &es7154_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c new file mode 100644 index 000000000000..87991bd4acef --- /dev/null +++ b/sound/soc/codecs/es7241.c @@ -0,0 +1,322 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/gpio/consumer.h> +#include <linux/of_platform.h> +#include <linux/module.h> +#include <sound/soc.h> + +struct es7241_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *slv_mfs; + unsigned int slv_mfs_num; + unsigned int mst_mfs; + unsigned int mst_m0:1; + unsigned int mst_m1:1; +}; + +struct es7241_chip { + const struct es7241_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7241_data { + struct gpio_desc *reset; + struct gpio_desc *m0; + struct gpio_desc *m1; + unsigned int fmt; + unsigned int mclk; + bool is_slave; + const struct es7241_chip *chip; +}; + +static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) +{ + /* put the device in reset */ + gpiod_set_value_cansleep(priv->reset, 0); + + /* set the mode */ + gpiod_set_value_cansleep(priv->m0, m0); + gpiod_set_value_cansleep(priv->m1, m1); + + /* take the device out of reset - datasheet does not specify a delay */ + gpiod_set_value_cansleep(priv->reset, 1); +} + +static int es7241_set_slave_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + int j; + + if (!mfs) + goto out_ok; + + for (j = 0; j < mode->slv_mfs_num; j++) { + if (mode->slv_mfs[j] == mfs) + goto out_ok; + } + + return -EINVAL; + +out_ok: + es7241_set_mode(priv, 1, 1); + return 0; +} + +static int es7241_set_master_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + /* + * We can't really set clock ratio, if the mclk/lrclk is different + * from what we provide, then error out + */ + if (mfs && mfs != mode->mst_mfs) + return -EINVAL; + + es7241_set_mode(priv, mode->mst_m0, mode->mst_m1); + + return 0; +} + +static int es7241_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int mfs = priv->mclk / rate; + int i; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7241_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate >= mode->rate_max) + continue; + + if (priv->is_slave) + return es7241_set_slave_mode(priv, mode, mfs); + else + return es7241_set_master_mode(priv, mode, mfs); + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + +static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(dai->dev, "Unsupported dai clock inversion\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) { + dev_err(dai->dev, "Invalid dai format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + priv->is_slave = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + priv->is_slave = false; + break; + + default: + dev_err(dai->dev, "Unsupported clock configuration\n"); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops es7241_dai_ops = { + .set_fmt = es7241_set_fmt, + .hw_params = es7241_hw_params, + .set_sysclk = es7241_set_sysclk, +}; + +static struct snd_soc_dai_driver es7241_dai = { + .name = "es7241-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7241_dai_ops, +}; + +static const struct es7241_clock_mode es7241_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .slv_mfs_num = 5, + .mst_mfs = 256, + .mst_m0 = 0, + .mst_m1 = 0, + }, { + /* Double speed mode */ + .rate_min = 50000, + .rate_max = 100000, + .slv_mfs = (unsigned int[]) { 128, 192 }, + .slv_mfs_num = 2, + .mst_mfs = 128, + .mst_m0 = 1, + .mst_m1 = 0, + }, { + /* Quad speed mode */ + .rate_min = 100000, + .rate_max = 200000, + .slv_mfs = (unsigned int[]) { 64 }, + .slv_mfs_num = 1, + .mst_mfs = 64, + .mst_m0 = 0, + .mst_m1 = 1, + }, +}; + +static const struct es7241_chip es7241_chip = { + .modes = es7241_modes, + .mode_num = ARRAY_SIZE(es7241_modes), +}; + +static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AINL"), + SND_SOC_DAPM_INPUT("AINR"), + SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0), +}; + +static const struct snd_soc_dapm_route es7241_dapm_routes[] = { + { "ADC", NULL, "AINL", }, + { "ADC", NULL, "AINR", }, + { "ADC", NULL, "VDDA", }, + { "Capture", NULL, "VDDP", }, + { "Capture", NULL, "VDDD", }, +}; + +static const struct snd_soc_component_driver es7241_component_driver = { + .dapm_widgets = es7241_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es7241_dapm_widgets), + .dapm_routes = es7241_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv) +{ + bool is_leftj; + + /* + * The format is given by a pull resistor on the SDOUT pin: + * pull-up for i2s, pull-down for left justified. + */ + is_leftj = of_property_read_bool(dev->of_node, + "everest,sdout-pull-down"); + if (is_leftj) + priv->fmt = SND_SOC_DAIFMT_LEFT_J; + else + priv->fmt = SND_SOC_DAIFMT_I2S; +} + +static int es7241_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct es7241_data *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + es7241_parse_fmt(dev, priv); + + priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset)) { + err = PTR_ERR(priv->reset); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'reset' gpio: %d", err); + return err; + } + + priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW); + if (IS_ERR(priv->m0)) { + err = PTR_ERR(priv->m0); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm0' gpio: %d", err); + return err; + } + + priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW); + if (IS_ERR(priv->m1)) { + err = PTR_ERR(priv->m1); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm1' gpio: %d", err); + return err; + } + + return devm_snd_soc_register_component(&pdev->dev, + &es7241_component_driver, + &es7241_dai, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id es7241_ids[] = { + { .compatible = "everest,es7241", .data = &es7241_chip }, + { } +}; +MODULE_DEVICE_TABLE(of, es7241_ids); +#endif + +static struct platform_driver es7241_driver = { + .driver = { + .name = "es7241", + .of_match_table = of_match_ptr(es7241_ids), + }, + .probe = es7241_probe, +}; + +module_platform_driver(es7241_driver); + +MODULE_DESCRIPTION("ASoC ES7241 audio codec driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..7b8533abf637 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -85,7 +85,7 @@ struct hdac_hdmi_pin { bool mst_capable; struct hdac_hdmi_port *ports; int num_ports; - struct hdac_ext_device *edev; + struct hdac_device *hdev; }; struct hdac_hdmi_port { @@ -126,6 +126,9 @@ struct hdac_hdmi_drv_data { }; struct hdac_hdmi_priv { + struct hdac_device *hdev; + struct snd_soc_component *component; + struct snd_card *card; struct hdac_hdmi_dai_port_map dai_map[HDA_MAX_CVTS]; struct list_head pin_list; struct list_head cvt_list; @@ -139,7 +142,7 @@ struct hdac_hdmi_priv { struct snd_soc_dai_driver *dai_drv; }; -#define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +#define hdev_to_hdmi_priv(_hdev) dev_get_drvdata(&(_hdev)->dev) static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, @@ -158,7 +161,7 @@ hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { - struct hdac_ext_device *edev = port->pin->edev; + struct hdac_device *hdev = port->pin->hdev; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -172,7 +175,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, * ports. */ if (pcm->jack_event == 0) { - dev_dbg(&edev->hdev.dev, + dev_dbg(&hdev->dev, "jack report for pcm=%d\n", pcm->pcm_id); snd_soc_jack_report(pcm->jack, SND_JACK_AVOUT, @@ -198,19 +201,18 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, /* * Get the no devices that can be connected to a port on the Pin widget. */ -static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_get_port_len(struct hdac_device *hdev, hda_nid_t nid) { unsigned int caps; unsigned int type, param; - caps = get_wcaps(&edev->hdev, nid); + caps = get_wcaps(hdev, nid); type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL) || (type != AC_WID_PIN)) return 0; - param = snd_hdac_read_parm_uncached(&edev->hdev, nid, - AC_PAR_DEVLIST_LEN); + param = snd_hdac_read_parm_uncached(hdev, nid, AC_PAR_DEVLIST_LEN); if (param == -1) return param; @@ -222,10 +224,10 @@ static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) * id selected on the pin. Return 0 means the first port entry * is selected or MST is not supported. */ -static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_get(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - return snd_hdac_codec_read(&edev->hdev, port->pin->nid, + return snd_hdac_codec_read(hdev, port->pin->nid, 0, AC_VERB_GET_DEVICE_SEL, 0); } @@ -233,7 +235,7 @@ static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, * Sets the selected port entry for the configuring Pin widget verb. * returns error if port set is not equal to port get otherwise success */ -static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_set(struct hdac_device *hdev, struct hdac_hdmi_port *port) { int num_ports; @@ -242,8 +244,7 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, return 0; /* AC_PAR_DEVLIST_LEN is 0 based. */ - num_ports = hdac_hdmi_get_port_len(edev, port->pin->nid); - + num_ports = hdac_hdmi_get_port_len(hdev, port->pin->nid); if (num_ports < 0) return -EIO; /* @@ -253,13 +254,13 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, if (num_ports + 1 < port->id) return 0; - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_DEVICE_SEL, port->id); - if (port->id != hdac_hdmi_port_select_get(edev, port)) + if (port->id != hdac_hdmi_port_select_get(hdev, port)) return -EIO; - dev_dbg(&edev->hdev.dev, "Selected the port=%d\n", port->id); + dev_dbg(&hdev->dev, "Selected the port=%d\n", port->id); return 0; } @@ -277,13 +278,6 @@ static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, return NULL; } -static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - return to_ehdac_device(hdev); -} - static unsigned int sad_format(const u8 *sad) { return ((sad[0] >> 0x3) & 0x1f); @@ -324,15 +318,13 @@ format_constraint: } static void -hdac_hdmi_set_dip_index(struct hdac_ext_device *edev, hda_nid_t pin_nid, +hdac_hdmi_set_dip_index(struct hdac_device *hdev, hda_nid_t pin_nid, int packet_index, int byte_index) { int val; val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hdac_codec_write(&edev->hdev, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_INDEX, val); + snd_hdac_codec_write(hdev, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); } struct dp_audio_infoframe { @@ -347,14 +339,14 @@ struct dp_audio_infoframe { u8 LFEPBL01_LSV36_DM_INH7; }; -static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, +static int hdac_hdmi_setup_audio_infoframe(struct hdac_device *hdev, struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port) { uint8_t buffer[HDMI_INFOFRAME_HEADER_SIZE + HDMI_AUDIO_INFOFRAME_SIZE]; struct hdmi_audio_infoframe frame; struct hdac_hdmi_pin *pin = port->pin; struct dp_audio_infoframe dp_ai; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt = pcm->cvt; u8 *dip; int ret; @@ -363,11 +355,11 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, u8 conn_type; int channels, ca; - ca = snd_hdac_channel_allocation(&edev->hdev, port->eld.info.spk_alloc, + ca = snd_hdac_channel_allocation(hdev, port->eld.info.spk_alloc, pcm->channels, pcm->chmap_set, true, pcm->chmap); channels = snd_hdac_get_active_channels(ca); - hdmi->chmap.ops.set_channel_count(&edev->hdev, cvt->nid, channels); + hdmi->chmap.ops.set_channel_count(hdev, cvt->nid, channels); snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, pcm->channels, pcm->chmap, pcm->chmap_set); @@ -400,32 +392,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, break; default: - dev_err(&edev->hdev.dev, "Invalid connection type: %d\n", - conn_type); + dev_err(&hdev->dev, "Invalid connection type: %d\n", conn_type); return -EIO; } /* stop infoframe transmission */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); /* Fill infoframe. Index auto-incremented */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { for (i = 0; i < sizeof(buffer); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, dip[i]); } /* Start infoframe */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); return 0; @@ -435,12 +426,12 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: strm_tag: %d\n", __func__, tx_mask); + dev_dbg(&hdev->dev, "%s: strm_tag: %d\n", __func__, tx_mask); dai_map = &hdmi->dai_map[dai->id]; @@ -455,8 +446,8 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; @@ -469,7 +460,7 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return -ENODEV; if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "device is not configured for this pin:port%d:%d\n", port->pin->nid, port->id); return -ENODEV; @@ -489,28 +480,28 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return 0; } -static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, +static int hdac_hdmi_query_port_connlist(struct hdac_device *hdev, struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - if (!(get_wcaps(&edev->hdev, pin->nid) & AC_WCAP_CONN_LIST)) { - dev_warn(&edev->hdev.dev, + if (!(get_wcaps(hdev, pin->nid) & AC_WCAP_CONN_LIST)) { + dev_warn(&hdev->dev, "HDMI: pin %d wcaps %#x does not support connection list\n", - pin->nid, get_wcaps(&edev->hdev, pin->nid)); + pin->nid, get_wcaps(hdev, pin->nid)); return -EINVAL; } - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; - port->num_mux_nids = snd_hdac_get_connections(&edev->hdev, pin->nid, + port->num_mux_nids = snd_hdac_get_connections(hdev, pin->nid, port->mux_nids, HDA_MAX_CONNECTIONS); if (port->num_mux_nids == 0) - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "No connections found for pin:port %d:%d\n", pin->nid, port->id); - dev_dbg(&edev->hdev.dev, "num_mux_nids %d for pin:port %d:%d\n", + dev_dbg(&hdev->dev, "num_mux_nids %d for pin:port %d:%d\n", port->num_mux_nids, pin->nid, port->id); return port->num_mux_nids; @@ -526,7 +517,7 @@ static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, * connected. */ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( - struct hdac_ext_device *edev, + struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { @@ -541,7 +532,7 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( list_for_each_entry(port, &pcm->port_list, head) { mutex_lock(&pcm->lock); - ret = hdac_hdmi_query_port_connlist(edev, + ret = hdac_hdmi_query_port_connlist(hdev, port->pin, port); mutex_unlock(&pcm->lock); if (ret < 0) @@ -568,8 +559,8 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; struct hdac_hdmi_port *port; @@ -578,7 +569,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, dai_map = &hdmi->dai_map[dai->id]; cvt = dai_map->cvt; - port = hdac_hdmi_get_port_from_cvt(edev, hdmi, cvt); + port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt); /* * To make PA and other userland happy. @@ -589,7 +580,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Failed: present?:%d ELD valid?:%d pin:port: %d:%d\n", port->eld.monitor_present, port->eld.eld_valid, port->pin->nid, port->id); @@ -611,8 +602,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; @@ -695,10 +685,10 @@ static void hdac_hdmi_fill_route(struct snd_soc_dapm_route *route, route->connected = handler; } -static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, +static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; struct hdac_hdmi_port *p; @@ -715,33 +705,32 @@ static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, return NULL; } -static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, +static void hdac_hdmi_set_power_state(struct hdac_device *hdev, hda_nid_t nid, unsigned int pwr_state) { int count; unsigned int state; - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) { - if (!snd_hdac_check_power_state(&edev->hdev, nid, pwr_state)) { + if (get_wcaps(hdev, nid) & AC_WCAP_POWER) { + if (!snd_hdac_check_power_state(hdev, nid, pwr_state)) { for (count = 0; count < 10; count++) { - snd_hdac_codec_read(&edev->hdev, nid, 0, + snd_hdac_codec_read(hdev, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); - state = snd_hdac_sync_power_state(&edev->hdev, + state = snd_hdac_sync_power_state(hdev, nid, pwr_state); if (!(state & AC_PWRST_ERROR)) break; } } - } } -static void hdac_hdmi_set_amp(struct hdac_ext_device *edev, +static void hdac_hdmi_set_amp(struct hdac_device *hdev, hda_nid_t nid, int val) { - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_OUT_AMP) - snd_hdac_codec_write(&edev->hdev, nid, 0, + if (get_wcaps(hdev, nid) & AC_WCAP_OUT_AMP) + snd_hdac_codec_write(hdev, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); } @@ -750,40 +739,40 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!pcm) return -EIO; /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D0); /* Enable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_UNMUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_UNMUTE); - return hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + return hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); case SND_SOC_DAPM_POST_PMD: - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_MUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_MUTE); /* Disable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D3); break; } @@ -795,11 +784,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_cvt *cvt = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); pcm = hdac_hdmi_get_pcm_from_cvt(hdmi, cvt); @@ -808,29 +797,29 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0); /* Enable transmission */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_1, 1); /* Category Code (CC) to zero */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, pcm->format); break; case SND_SOC_DAPM_POST_PMD: - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D3); break; } @@ -842,10 +831,10 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); int mux_idx; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); if (!kc) @@ -854,11 +843,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, mux_idx = dapm_kcontrol_get_value(kc); /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; if (mux_idx > 0) { - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_CONNECT_SEL, (mux_idx - 1)); } @@ -877,8 +866,8 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); struct snd_soc_dapm_context *dapm = w->dapm; struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; const char *cvt_name = e->texts[ucontrol->value.enumerated.item[0]]; @@ -931,12 +920,12 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, * care of selecting the right one and leaving all other inputs selected to * "NONE" */ -static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, +static int hdac_hdmi_create_pin_port_muxs(struct hdac_device *hdev, struct hdac_hdmi_port *port, struct snd_soc_dapm_widget *widget, const char *widget_name) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = port->pin; struct snd_kcontrol_new *kc; struct hdac_hdmi_cvt *cvt; @@ -948,17 +937,17 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, int i = 0; int num_items = hdmi->num_cvt + 1; - kc = devm_kzalloc(&edev->hdev.dev, sizeof(*kc), GFP_KERNEL); + kc = devm_kzalloc(&hdev->dev, sizeof(*kc), GFP_KERNEL); if (!kc) return -ENOMEM; - se = devm_kzalloc(&edev->hdev.dev, sizeof(*se), GFP_KERNEL); + se = devm_kzalloc(&hdev->dev, sizeof(*se), GFP_KERNEL); if (!se) return -ENOMEM; snprintf(kc_name, NAME_SIZE, "Pin %d port %d Input", pin->nid, port->id); - kc->name = devm_kstrdup(&edev->hdev.dev, kc_name, GFP_KERNEL); + kc->name = devm_kstrdup(&hdev->dev, kc_name, GFP_KERNEL); if (!kc->name) return -ENOMEM; @@ -976,35 +965,35 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, se->mask = roundup_pow_of_two(se->items) - 1; sprintf(mux_items, "NONE"); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; list_for_each_entry(cvt, &hdmi->cvt_list, head) { i++; sprintf(mux_items, "cvt %d", cvt->nid); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; } - se->texts = devm_kmemdup(&edev->hdev.dev, items, + se->texts = devm_kmemdup(&hdev->dev, items, (num_items * sizeof(char *)), GFP_KERNEL); if (!se->texts) return -ENOMEM; - return hdac_hdmi_fill_widget_info(&edev->hdev.dev, widget, + return hdac_hdmi_fill_widget_info(&hdev->dev, widget, snd_soc_dapm_mux, port, widget_name, NULL, kc, 1, hdac_hdmi_pin_mux_widget_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_REG); } /* Add cvt <- input <- mux route map */ -static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_ext_device *edev, +static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_device *hdev, struct snd_soc_dapm_widget *widgets, struct snd_soc_dapm_route *route, int rindex) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); const struct snd_kcontrol_new *kc; struct soc_enum *se; int mux_index = hdmi->num_cvt + hdmi->num_ports; @@ -1046,8 +1035,8 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct snd_soc_dai_driver *dai_drv = hdmi->dai_drv; char widget_name[NAME_SIZE]; struct hdac_hdmi_cvt *cvt; @@ -1099,7 +1088,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) for (j = 0; j < pin->num_ports; j++) { sprintf(widget_name, "Pin%d-Port%d Mux", pin->nid, pin->ports[j].id); - ret = hdac_hdmi_create_pin_port_muxs(edev, + ret = hdac_hdmi_create_pin_port_muxs(hdev, &pin->ports[j], &widgets[i], widget_name); if (ret < 0) @@ -1134,7 +1123,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } } - hdac_hdmi_add_pinmux_cvt_route(edev, widgets, route, i); + hdac_hdmi_add_pinmux_cvt_route(hdev, widgets, route, i); snd_soc_dapm_new_controls(dapm, widgets, ((2 * hdmi->num_ports) + hdmi->num_cvt)); @@ -1146,9 +1135,9 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } -static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) +static int hdac_hdmi_init_dai_map(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; int dai_id = 0; @@ -1164,7 +1153,7 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) dai_id++; if (dai_id == HDA_MAX_CVTS) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Max dais supported: %d\n", dai_id); break; } @@ -1173,9 +1162,9 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) return 0; } -static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt; char name[NAME_SIZE]; @@ -1190,10 +1179,10 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) list_add_tail(&cvt->head, &hdmi->cvt_list); hdmi->num_cvt++; - return hdac_hdmi_query_cvt_params(&edev->hdev, cvt); + return hdac_hdmi_query_cvt_params(hdev, cvt); } -static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_eld(struct hdac_device *hdev, struct hdac_hdmi_port *port) { unsigned int ver, mnl; @@ -1202,7 +1191,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, >> DRM_ELD_VER_SHIFT; if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) { - dev_err(&edev->hdev.dev, "HDMI: Unknown ELD version %d\n", ver); + dev_err(&hdev->dev, "HDMI: Unknown ELD version %d\n", ver); return -EINVAL; } @@ -1210,7 +1199,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT; if (mnl > ELD_MAX_MNL) { - dev_err(&edev->hdev.dev, "HDMI: MNL Invalid %d\n", mnl); + dev_err(&hdev->dev, "HDMI: MNL Invalid %d\n", mnl); return -EINVAL; } @@ -1222,8 +1211,8 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - struct hdac_ext_device *edev = pin->edev; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = pin->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; @@ -1241,14 +1230,14 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, if (pin->mst_capable) port_id = port->id; - size = snd_hdac_acomp_get_eld(&edev->hdev, pin->nid, port_id, + size = snd_hdac_acomp_get_eld(hdev, pin->nid, port_id, &port->eld.monitor_present, port->eld.eld_buffer, ELD_MAX_SIZE); if (size > 0) { size = min(size, ELD_MAX_SIZE); - if (hdac_hdmi_parse_eld(edev, port) < 0) + if (hdac_hdmi_parse_eld(hdev, port) < 0) size = -EINVAL; } @@ -1260,11 +1249,11 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { - dev_err(&edev->hdev.dev, "%s: disconnect for pin:port %d:%d\n", + dev_err(&hdev->dev, "%s: disconnect for pin:port %d:%d\n", __func__, pin->nid, port->id); /* @@ -1316,9 +1305,9 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, return 0; } -static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin; int ret; @@ -1328,7 +1317,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) pin->nid = nid; pin->mst_capable = false; - pin->edev = edev; + pin->hdev = hdev; ret = hdac_hdmi_add_ports(hdmi, pin); if (ret < 0) return ret; @@ -1459,15 +1448,14 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, * Parse all nodes and store the cvt/pin nids in array * Add one time initialization for pin and cvt widgets */ -static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, struct snd_soc_dai_driver **dais, int *num_dais) { hda_nid_t nid; int i, num_nodes; struct hdac_hdmi_cvt *temp_cvt, *cvt_next; struct hdac_hdmi_pin *temp_pin, *pin_next; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int ret; hdac_hdmi_skl_enable_all_pins(hdev); @@ -1492,13 +1480,13 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, switch (type) { case AC_WID_AUD_OUT: - ret = hdac_hdmi_add_cvt(edev, nid); + ret = hdac_hdmi_add_cvt(hdev, nid); if (ret < 0) goto free_widgets; break; case AC_WID_PIN: - ret = hdac_hdmi_add_pin(edev, nid); + ret = hdac_hdmi_add_pin(hdev, nid); if (ret < 0) goto free_widgets; break; @@ -1518,7 +1506,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, } *num_dais = hdmi->num_cvt; - ret = hdac_hdmi_init_dai_map(edev); + ret = hdac_hdmi_init_dai_map(hdev); if (ret < 0) goto free_widgets; @@ -1542,19 +1530,24 @@ free_widgets: return ret; } +static int hdac_hdmi_pin2port(void *aptr, int pin) +{ + return pin - 4; /* map NID 0x05 -> port #1 */ +} + static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { - struct hdac_ext_device *edev = aptr; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = NULL; struct hdac_hdmi_port *hport = NULL; - struct snd_soc_component *component = edev->scodec; + struct snd_soc_component *component = hdmi->component; int i; /* Don't know how this mapping is derived */ hda_nid_t pin_nid = port + 0x04; - dev_dbg(&edev->hdev.dev, "%s: for pin:%d port=%d\n", __func__, + dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); /* @@ -1567,7 +1560,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) SNDRV_CTL_POWER_D0) return; - if (atomic_read(&edev->hdev.in_pm)) + if (atomic_read(&hdev->in_pm)) return; list_for_each_entry(pin, &hdmi->pin_list, head) { @@ -1595,7 +1588,8 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) } -static struct i915_audio_component_audio_ops aops = { +static struct drm_audio_component_audio_ops aops = { + .pin2port = hdac_hdmi_pin2port, .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; @@ -1614,15 +1608,15 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, /* create jack pin kcontrols */ static int create_fill_jack_kcontrols(struct snd_soc_card *card, - struct hdac_ext_device *edev) + struct hdac_device *hdev) { struct hdac_hdmi_pin *pin; struct snd_kcontrol_new *kc; char kc_name[NAME_SIZE], xname[NAME_SIZE]; char *name; int i = 0, j; - struct snd_soc_component *component = edev->scodec; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + struct snd_soc_component *component = hdmi->component; kc = devm_kcalloc(component->dev, hdmi->num_ports, sizeof(*kc), GFP_KERNEL); @@ -1659,8 +1653,8 @@ static int create_fill_jack_kcontrols(struct snd_soc_card *card, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pin *pin; struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; @@ -1715,7 +1709,7 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, return ret; /* Add Jack Pin switch Kcontrol */ - ret = create_fill_jack_kcontrols(dapm->card, edev); + ret = create_fill_jack_kcontrols(dapm->card, hdev); if (ret < 0) return ret; @@ -1735,8 +1729,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, struct snd_soc_jack *jack) { struct snd_soc_component *component = dai->component; - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pcm *pcm; struct snd_pcm *snd_pcm; int err; @@ -1758,7 +1752,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, if (snd_pcm) { err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); if (err < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "chmap control add failed with err: %d for pcm: %d\n", err, device); kfree(pcm); @@ -1772,7 +1766,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); -static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, +static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { int i; @@ -1781,7 +1775,7 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, list_for_each_entry(pin, &hdmi->pin_list, head) { if (detect_pin_caps) { - if (hdac_hdmi_get_port_len(edev, pin->nid) == 0) + if (hdac_hdmi_get_port_len(hdev, pin->nid) == 0) pin->mst_capable = false; else pin->mst_capable = true; @@ -1798,68 +1792,67 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, static int hdmi_codec_probe(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct hdac_ext_link *hlink = NULL; int ret; - edev->scodec = component; + hdmi->component = component; /* * hold the ref while we probe, also no need to drop the ref on * exit, we call pm_runtime_suspend() so that will do for us */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; - aops.audio_ptr = edev; - ret = snd_hdac_i915_register_notifier(&aops); + aops.audio_ptr = hdev; + ret = snd_hdac_acomp_register_notifier(hdev->bus, &aops); if (ret < 0) { - dev_err(&edev->hdev.dev, "notifier register failed: err: %d\n", - ret); + dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret); return ret; } - hdac_hdmi_present_sense_all_pins(edev, hdmi, true); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, true); /* Imp: Store the card pointer in hda_codec */ - edev->card = dapm->card->snd_card; + hdmi->card = dapm->card->snd_card; /* * hdac_device core already sets the state to active and calls * get_noresume. So enable runtime and set the device to suspend. */ - pm_runtime_enable(&edev->hdev.dev); - pm_runtime_put(&edev->hdev.dev); - pm_runtime_suspend(&edev->hdev.dev); + pm_runtime_enable(&hdev->dev); + pm_runtime_put(&hdev->dev); + pm_runtime_suspend(&hdev->dev); return 0; } static void hdmi_codec_remove(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; - pm_runtime_disable(&edev->hdev.dev); + pm_runtime_disable(&hdev->dev); } #ifdef CONFIG_PM static int hdmi_codec_prepare(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); - pm_runtime_get_sync(&edev->hdev.dev); + pm_runtime_get_sync(&hdev->dev); /* * Power down afg. @@ -1876,16 +1869,15 @@ static int hdmi_codec_prepare(struct device *dev) static void hdmi_codec_complete(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* * As the ELD notify callback request is not entertained while the @@ -1893,9 +1885,9 @@ static void hdmi_codec_complete(struct device *dev) * all pins here. pin capablity change is not support, so use the * already set pin caps. */ - hdac_hdmi_present_sense_all_pins(edev, hdmi, false); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - pm_runtime_put_sync(&edev->hdev.dev); + pm_runtime_put_sync(&hdev->dev); } #else #define hdmi_codec_prepare NULL @@ -1922,7 +1914,6 @@ static void hdac_hdmi_get_chmap(struct hdac_device *hdev, int pcm_idx, static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, unsigned char *chmap, int prepared) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_port *port; @@ -1938,7 +1929,7 @@ static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, memcpy(pcm->chmap, chmap, ARRAY_SIZE(pcm->chmap)); list_for_each_entry(port, &pcm->port_list, head) if (prepared) - hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); mutex_unlock(&pcm->lock); } @@ -1987,10 +1978,9 @@ static struct hdac_hdmi_drv_data intel_drv_data = { .vendor_nid = INTEL_VENDOR_NID, }; -static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_device *hdev = &edev->hdev; - struct hdac_hdmi_priv *hdmi_priv; + struct hdac_hdmi_priv *hdmi_priv = NULL; struct snd_soc_dai_driver *hdmi_dais = NULL; struct hdac_ext_link *hlink = NULL; int num_dais = 0; @@ -1999,24 +1989,24 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); /* hold the ref while we probe */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); hdmi_priv = devm_kzalloc(&hdev->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; - edev->private_data = hdmi_priv; snd_hdac_register_chmap_ops(hdev, &hdmi_priv->chmap); hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; + hdmi_priv->hdev = hdev; if (!hdac_id) return -ENODEV; @@ -2027,7 +2017,7 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) else hdmi_priv->drv_data = &intel_drv_data; - dev_set_drvdata(&hdev->dev, edev); + dev_set_drvdata(&hdev->dev, hdmi_priv); INIT_LIST_HEAD(&hdmi_priv->pin_list); INIT_LIST_HEAD(&hdmi_priv->cvt_list); @@ -2038,15 +2028,15 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) * Turned off in the runtime_suspend during the first explicit * pm_runtime_suspend call. */ - ret = snd_hdac_display_power(edev->hdev.bus, true); + ret = snd_hdac_display_power(hdev->bus, true); if (ret < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "Cannot turn on display power on i915 err: %d\n", ret); return ret; } - ret = hdac_hdmi_parse_and_map_nid(edev, &hdmi_dais, &num_dais); + ret = hdac_hdmi_parse_and_map_nid(hdev, &hdmi_dais, &num_dais); if (ret < 0) { dev_err(&hdev->dev, "Failed in parse and map nid with err: %d\n", ret); @@ -2058,14 +2048,14 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec, hdmi_dais, num_dais); - snd_hdac_ext_bus_link_put(edev->ebus, hlink); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); return ret; } -static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin, *pin_next; struct hdac_hdmi_cvt *cvt, *cvt_next; struct hdac_hdmi_pcm *pcm, *pcm_next; @@ -2103,12 +2093,79 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2129,27 +2186,25 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) AC_PWRST_D3); err = snd_hdac_display_power(bus, false); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_put(ebus, hlink); + snd_hdac_ext_bus_link_put(bus, hlink); return 0; } static int hdac_hdmi_runtime_resume(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2159,22 +2214,22 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(ebus, hlink); + snd_hdac_ext_bus_link_get(bus, hlink); err = snd_hdac_display_power(bus, true); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, @@ -2206,14 +2261,12 @@ static const struct hda_device_id hdmi_list[] = { MODULE_DEVICE_TABLE(hdaudio, hdmi_list); -static struct hdac_ext_driver hdmi_driver = { - . hdac = { - .driver = { - .name = "HDMI HDA Codec", - .pm = &hdac_hdmi_pm, - }, - .id_table = hdmi_list, +static struct hdac_driver hdmi_driver = { + .driver = { + .name = "HDMI HDA Codec", + .pm = &hdac_hdmi_pm, }, + .id_table = hdmi_list, .probe = hdac_hdmi_dev_probe, .remove = hdac_hdmi_dev_remove, }; diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 3e5b12de71bb..d00734d31e04 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -780,7 +780,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) hcp->hcd = *hcd; mutex_init(&hcp->current_stream_lock); - hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + hcp->daidrv = devm_kcalloc(dev, dai_count, sizeof(*hcp->daidrv), GFP_KERNEL); if (!hcp->daidrv) return -ENOMEM; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 74d7f52c7e73..6e6134589588 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case MAX9850_STATUSA: case MAX9850_STATUSB: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 17104f8dc1a9..e3c8cd17daf2 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) return -EINVAL; - osrate = osr_adc_sel[osr].osr; if (rate * osr > CLK_ADC_MAX) { dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd14453f06e..468d5143e2c4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea4987b7d..b9fed99d8b5e 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c index 21f15219b3ad..8df6447c76a6 100644 --- a/sound/soc/codecs/pcm1789.c +++ b/sound/soc/codecs/pcm1789.c @@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev) { struct pcm1789_private *priv = dev_get_drvdata(dev); - if (&priv->work) - flush_work(&priv->work); + flush_work(&priv->work); return 0; } diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 88fde70b1e9e..690c26e7389e 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component); unsigned int rate = params_rate(params); - unsigned int format = params_format(params); + snd_pcm_format_t format = params_format(params); unsigned int width = params_width(params); unsigned int channels = params_channels(params); unsigned int div_lrck; diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index f4c8c45f4010..c4452efc7970 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0l = 562949953421312; + r0l = 562949953421312ULL; if (rhl != 0) do_div(r0l, rhl); pr_debug("Left_r0 = 0x%llx\n", r0l); @@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0r = 562949953421312; + r0r = 562949953421312ULL; if (rhl != 0) do_div(r0r, rhl); pr_debug("Right_r0 = 0x%llx\n", r0r); @@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c, rt1305_reset(rt1305->regmap); rt1305_calibrate(rt1305); - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1305, rt1305_dai, ARRAY_SIZE(rt1305_dai)); } -static int rt1305_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt1305_i2c_shutdown(struct i2c_client *client) { struct rt1305_priv *rt1305 = i2c_get_clientdata(client); @@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = { #endif }, .probe = rt1305_i2c_probe, - .remove = rt1305_i2c_remove, .shutdown = rt1305_i2c_shutdown, .id_table = rt1305_i2c_id, }; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index cf6dce69eb2a..865f49ac38dd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg) case RT5631_INDEX_ADD: case RT5631_INDEX_DATA: case RT5631_EQ_CTRL: - return 1; + return true; default: - return 0; + return false; } } @@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg) case RT5631_VENDOR_ID: case RT5631_VENDOR_ID1: case RT5631_VENDOR_ID2: - return 1; + return true; default: - return 0; + return false; } } @@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), - SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), - SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), /* MONO IN */ SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8bf8d360c25f..27770143ae8f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 712384581ebf..1dc70f452c1b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3449,8 +3449,9 @@ static int rt5645_probe(struct snd_soc_component *component) if (rt5645->pdata.long_name) component->card->long_name = rt5645->pdata.long_name; - rt5645->eq_param = devm_kzalloc(component->dev, - RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + rt5645->eq_param = devm_kcalloc(component->dev, + RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), + GFP_KERNEL); return 0; } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 6b5669f3e85d..985852fd9723 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL, RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT, @@ -1581,6 +1583,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component snd_soc_dapm_mutex_unlock(dapm); } +static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR); + rt5651->ovcd_irq_enabled = true; +} + +static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP); + rt5651->ovcd_irq_enabled = false; +} + static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component) { snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, @@ -1622,10 +1642,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) return val == 0; } -/* Jack detect timings */ +/* Jack detect and button-press timings */ #define JACK_SETTLE_TIME 100 /* milli seconds */ #define JACK_DETECT_COUNT 5 #define JACK_DETECT_MAXCOUNT 20 /* Aprox. 2 seconds worth of tries */ +#define JACK_UNPLUG_TIME 80 /* milli seconds */ +#define BP_POLL_TIME 10 /* milli seconds */ +#define BP_POLL_MAXCOUNT 200 /* assume something is wrong after this */ +#define BP_THRESHOLD 3 + +static void rt5651_start_button_press_work(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + rt5651->poll_count = 0; + rt5651->press_count = 0; + rt5651->release_count = 0; + rt5651->pressed = false; + rt5651->press_reported = false; + rt5651_clear_micbias1_ovcd(component); + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} + +static void rt5651_button_press_work(struct work_struct *work) +{ + struct rt5651_priv *rt5651 = + container_of(work, struct rt5651_priv, bp_work.work); + struct snd_soc_component *component = rt5651->component; + + /* Check the jack was not removed underneath us */ + if (!rt5651_jack_inserted(component)) + return; + + if (rt5651_micbias1_ovcd(component)) { + rt5651->release_count = 0; + rt5651->press_count++; + /* Remember till after JACK_UNPLUG_TIME wait */ + if (rt5651->press_count >= BP_THRESHOLD) + rt5651->pressed = true; + rt5651_clear_micbias1_ovcd(component); + } else { + rt5651->press_count = 0; + rt5651->release_count++; + } + + /* + * The pins get temporarily shorted on jack unplug, so we poll for + * at least JACK_UNPLUG_TIME milli-seconds before reporting a press. + */ + rt5651->poll_count++; + if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) { + schedule_delayed_work(&rt5651->bp_work, + msecs_to_jiffies(BP_POLL_TIME)); + return; + } + + if (rt5651->pressed && !rt5651->press_reported) { + dev_dbg(component->dev, "headset button press\n"); + snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + rt5651->press_reported = true; + } + + if (rt5651->release_count >= BP_THRESHOLD) { + if (rt5651->press_reported) { + dev_dbg(component->dev, "headset button release\n"); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + /* Re-enable OVCD IRQ to detect next press */ + rt5651_enable_micbias1_ovcd_irq(component); + return; /* Stop polling */ + } + + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} static int rt5651_detect_headset(struct snd_soc_component *component) { @@ -1676,15 +1766,58 @@ static void rt5651_jack_detect_work(struct work_struct *work) { struct rt5651_priv *rt5651 = container_of(work, struct rt5651_priv, jack_detect_work); + struct snd_soc_component *component = rt5651->component; int report = 0; - if (rt5651_jack_inserted(rt5651->component)) { - rt5651_enable_micbias1_for_ovcd(rt5651->component); - report = rt5651_detect_headset(rt5651->component); - rt5651_disable_micbias1_for_ovcd(rt5651->component); + if (!rt5651_jack_inserted(component)) { + /* Jack removed, or spurious IRQ? */ + if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) { + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + cancel_delayed_work_sync(&rt5651->bp_work); + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + } + snd_soc_jack_report(rt5651->hp_jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + dev_dbg(component->dev, "jack unplugged\n"); + } + } else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) { + /* Jack inserted */ + WARN_ON(rt5651->ovcd_irq_enabled); + rt5651_enable_micbias1_for_ovcd(component); + report = rt5651_detect_headset(component); + if (report == SND_JACK_HEADSET) { + /* Enable ovcd IRQ for button press detect. */ + rt5651_enable_micbias1_ovcd_irq(component); + } else { + /* No more need for overcurrent detect. */ + rt5651_disable_micbias1_for_ovcd(component); + } + dev_dbg(component->dev, "detect report %#02x\n", report); + snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) { + dev_dbg(component->dev, "OVCD IRQ\n"); + + /* + * The ovcd IRQ keeps firing while the button is pressed, so + * we disable it and start polling the button until released. + * + * The disable will make the IRQ pin 0 again and since we get + * IRQs on both edges (so as to detect both jack plugin and + * unplug) this means we will immediately get another IRQ. + * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP. + */ + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_start_button_press_work(component); + + /* + * If the jack-detect IRQ flag goes high (unplug) after our + * above rt5651_jack_inserted() check and before we have + * disabled the OVCD IRQ, the IRQ pin will stay high and as + * we react to edges, we miss the unplug event -> recheck. + */ + queue_work(system_long_wq, &rt5651->jack_detect_work); } - - snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); } static irqreturn_t rt5651_irq(int irq, void *data) @@ -1696,14 +1829,18 @@ static irqreturn_t rt5651_irq(int irq, void *data) return IRQ_HANDLED; } -static int rt5651_set_jack(struct snd_soc_component *component, - struct snd_soc_jack *hp_jack, void *data) +static void rt5651_cancel_work(void *data) { - struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); - int ret; + struct rt5651_priv *rt5651 = data; - if (!rt5651->irq) - return -EINVAL; + cancel_work_sync(&rt5651->jack_detect_work); + cancel_delayed_work_sync(&rt5651->bp_work); +} + +static void rt5651_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hp_jack) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); /* IRQ output on GPIO1 */ snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, @@ -1730,10 +1867,10 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); break; case RT5651_JD_NULL: - return 0; + return; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); - return -EINVAL; + return; } /* Enable jack detect power */ @@ -1767,19 +1904,39 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; - - ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL, - rt5651_irq, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, "rt5651", rt5651); - if (ret) { - dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret); - return ret; + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_enable_micbias1_for_ovcd(component); + rt5651_enable_micbias1_ovcd_irq(component); } + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); +} + +static void rt5651_disable_jack_detect(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + disable_irq(rt5651->irq); + rt5651_cancel_work(rt5651); + + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + + rt5651->hp_jack = NULL; +} + +static int rt5651_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + rt5651_enable_jack_detect(component, jack); + else + rt5651_disable_jack_detect(component); return 0; } @@ -2034,8 +2191,26 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, rt5651->irq = i2c->irq; rt5651->hp_mute = 1; + INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work); INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); + /* Make sure work is stopped on probe-error / remove */ + ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651); + if (ret) + return ret; + + ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5651", rt5651); + if (ret == 0) { + /* Gets re-enabled by rt5651_set_jack() */ + disable_irq(rt5651->irq); + } else { + dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + rt5651->irq, ret); + rt5651->irq = -ENXIO; + } + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); @@ -2043,15 +2218,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, return ret; } -static int rt5651_i2c_remove(struct i2c_client *i2c) -{ - struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c); - - cancel_work_sync(&rt5651->jack_detect_work); - - return 0; -} - static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", @@ -2059,7 +2225,6 @@ static struct i2c_driver rt5651_i2c_driver = { .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, - .remove = rt5651_i2c_remove, .id_table = rt5651_i2c_id, }; module_i2c_driver(rt5651_i2c_driver); diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 3a0968c53fde..ac6de6fb5414 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2071,8 +2071,16 @@ struct rt5651_pll_code { struct rt5651_priv { struct snd_soc_component *component; struct regmap *regmap; + /* Jack and button detect data */ struct snd_soc_jack *hp_jack; struct work_struct jack_detect_work; + struct delayed_work bp_work; + bool ovcd_irq_enabled; + bool pressed; + bool press_reported; + int press_count; + int release_count; + int poll_count; unsigned int jd_src; unsigned int ovcd_th; unsigned int ovcd_sf; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..9b7a1833d331 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; + /* fall through */ case 24: val |= (2 << 8); break; @@ -5007,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", .data = (const void *)RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c new file mode 100644 index 000000000000..640d400ca013 --- /dev/null +++ b/sound/soc/codecs/rt5682.c @@ -0,0 +1,2681 @@ +/* + * rt5682.c -- RT5682 ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao@realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5682.h> + +#include "rl6231.h" +#include "rt5682.h" + +#define RT5682_NUM_SUPPLIES 3 + +static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_sequence patch_list[] = { + {0x01c1, 0x1000}, +}; + +static const struct reg_default rt5682_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5682_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_CBJ_CTRL_2: + case RT5682_INT_ST_1: + case RT5682_4BTN_IL_CMD_1: + case RT5682_AJD1_CTRL: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_DEVICE_ID: + case RT5682_I2C_MODE: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_EFUSE_CTRL_2: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11: + case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5682_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_VERSION_ID: + case RT5682_VENDOR_ID: + case RT5682_DEVICE_ID: + case RT5682_HP_CTRL_1: + case RT5682_HP_CTRL_2: + case RT5682_HPL_GAIN: + case RT5682_HPR_GAIN: + case RT5682_I2C_CTRL: + case RT5682_CBJ_BST_CTRL: + case RT5682_CBJ_CTRL_1: + case RT5682_CBJ_CTRL_2: + case RT5682_CBJ_CTRL_3: + case RT5682_CBJ_CTRL_4: + case RT5682_CBJ_CTRL_5: + case RT5682_CBJ_CTRL_6: + case RT5682_CBJ_CTRL_7: + case RT5682_DAC1_DIG_VOL: + case RT5682_STO1_ADC_DIG_VOL: + case RT5682_STO1_ADC_BOOST: + case RT5682_HP_IMP_GAIN_1: + case RT5682_HP_IMP_GAIN_2: + case RT5682_SIDETONE_CTRL: + case RT5682_STO1_ADC_MIXER: + case RT5682_AD_DA_MIXER: + case RT5682_STO1_DAC_MIXER: + case RT5682_A_DAC1_MUX: + case RT5682_DIG_INF2_DATA: + case RT5682_REC_MIXER: + case RT5682_CAL_REC: + case RT5682_ALC_BACK_GAIN: + case RT5682_PWR_DIG_1: + case RT5682_PWR_DIG_2: + case RT5682_PWR_ANLG_1: + case RT5682_PWR_ANLG_2: + case RT5682_PWR_ANLG_3: + case RT5682_PWR_MIXER: + case RT5682_PWR_VOL: + case RT5682_CLK_DET: + case RT5682_RESET_LPF_CTRL: + case RT5682_RESET_HPF_CTRL: + case RT5682_DMIC_CTRL_1: + case RT5682_I2S1_SDP: + case RT5682_I2S2_SDP: + case RT5682_ADDA_CLK_1: + case RT5682_ADDA_CLK_2: + case RT5682_I2S1_F_DIV_CTRL_1: + case RT5682_I2S1_F_DIV_CTRL_2: + case RT5682_TDM_CTRL: + case RT5682_TDM_ADDA_CTRL_1: + case RT5682_TDM_ADDA_CTRL_2: + case RT5682_DATA_SEL_CTRL_1: + case RT5682_TDM_TCON_CTRL: + case RT5682_GLB_CLK: + case RT5682_PLL_CTRL_1: + case RT5682_PLL_CTRL_2: + case RT5682_PLL_TRACK_1: + case RT5682_PLL_TRACK_2: + case RT5682_PLL_TRACK_3: + case RT5682_PLL_TRACK_4: + case RT5682_PLL_TRACK_5: + case RT5682_PLL_TRACK_6: + case RT5682_PLL_TRACK_11: + case RT5682_SDW_REF_CLK: + case RT5682_DEPOP_1: + case RT5682_DEPOP_2: + case RT5682_HP_CHARGE_PUMP_1: + case RT5682_HP_CHARGE_PUMP_2: + case RT5682_MICBIAS_1: + case RT5682_MICBIAS_2: + case RT5682_PLL_TRACK_12: + case RT5682_PLL_TRACK_14: + case RT5682_PLL2_CTRL_1: + case RT5682_PLL2_CTRL_2: + case RT5682_PLL2_CTRL_3: + case RT5682_PLL2_CTRL_4: + case RT5682_RC_CLK_CTRL: + case RT5682_I2S_M_CLK_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_2: + case RT5682_EQ_CTRL_1: + case RT5682_EQ_CTRL_2: + case RT5682_IRQ_CTRL_1: + case RT5682_IRQ_CTRL_2: + case RT5682_IRQ_CTRL_3: + case RT5682_IRQ_CTRL_4: + case RT5682_INT_ST_1: + case RT5682_GPIO_CTRL_1: + case RT5682_GPIO_CTRL_2: + case RT5682_GPIO_CTRL_3: + case RT5682_HP_AMP_DET_CTRL_1: + case RT5682_HP_AMP_DET_CTRL_2: + case RT5682_MID_HP_AMP_DET: + case RT5682_LOW_HP_AMP_DET: + case RT5682_DELAY_BUF_CTRL: + case RT5682_SV_ZCD_1: + case RT5682_SV_ZCD_2: + case RT5682_IL_CMD_1: + case RT5682_IL_CMD_2: + case RT5682_IL_CMD_3: + case RT5682_IL_CMD_4: + case RT5682_IL_CMD_5: + case RT5682_IL_CMD_6: + case RT5682_4BTN_IL_CMD_1: + case RT5682_4BTN_IL_CMD_2: + case RT5682_4BTN_IL_CMD_3: + case RT5682_4BTN_IL_CMD_4: + case RT5682_4BTN_IL_CMD_5: + case RT5682_4BTN_IL_CMD_6: + case RT5682_4BTN_IL_CMD_7: + case RT5682_ADC_STO1_HP_CTRL_1: + case RT5682_ADC_STO1_HP_CTRL_2: + case RT5682_AJD1_CTRL: + case RT5682_JD1_THD: + case RT5682_JD2_THD: + case RT5682_JD_CTRL_1: + case RT5682_DUMMY_1: + case RT5682_DUMMY_2: + case RT5682_DUMMY_3: + case RT5682_DAC_ADC_DIG_VOL1: + case RT5682_BIAS_CUR_CTRL_2: + case RT5682_BIAS_CUR_CTRL_3: + case RT5682_BIAS_CUR_CTRL_4: + case RT5682_BIAS_CUR_CTRL_5: + case RT5682_BIAS_CUR_CTRL_6: + case RT5682_BIAS_CUR_CTRL_7: + case RT5682_BIAS_CUR_CTRL_8: + case RT5682_BIAS_CUR_CTRL_9: + case RT5682_BIAS_CUR_CTRL_10: + case RT5682_VREF_REC_OP_FB_CAP_CTRL: + case RT5682_CHARGE_PUMP_1: + case RT5682_DIG_IN_CTRL_1: + case RT5682_PAD_DRIVING_CTRL: + case RT5682_SOFT_RAMP_DEPOP: + case RT5682_CHOP_DAC: + case RT5682_CHOP_ADC: + case RT5682_CALIB_ADC_CTRL: + case RT5682_VOL_TEST: + case RT5682_SPKVDD_DET_STA: + case RT5682_TEST_MODE_CTRL_1: + case RT5682_TEST_MODE_CTRL_2: + case RT5682_TEST_MODE_CTRL_3: + case RT5682_TEST_MODE_CTRL_4: + case RT5682_TEST_MODE_CTRL_5: + case RT5682_PLL1_INTERNAL: + case RT5682_PLL2_INTERNAL: + case RT5682_STO_NG2_CTRL_1: + case RT5682_STO_NG2_CTRL_2: + case RT5682_STO_NG2_CTRL_3: + case RT5682_STO_NG2_CTRL_4: + case RT5682_STO_NG2_CTRL_5: + case RT5682_STO_NG2_CTRL_6: + case RT5682_STO_NG2_CTRL_7: + case RT5682_STO_NG2_CTRL_8: + case RT5682_STO_NG2_CTRL_9: + case RT5682_STO_NG2_CTRL_10: + case RT5682_STO1_DAC_SIL_DET: + case RT5682_SIL_PSV_CTRL1: + case RT5682_SIL_PSV_CTRL2: + case RT5682_SIL_PSV_CTRL3: + case RT5682_SIL_PSV_CTRL4: + case RT5682_SIL_PSV_CTRL5: + case RT5682_HP_IMP_SENS_CTRL_01: + case RT5682_HP_IMP_SENS_CTRL_02: + case RT5682_HP_IMP_SENS_CTRL_03: + case RT5682_HP_IMP_SENS_CTRL_04: + case RT5682_HP_IMP_SENS_CTRL_05: + case RT5682_HP_IMP_SENS_CTRL_06: + case RT5682_HP_IMP_SENS_CTRL_07: + case RT5682_HP_IMP_SENS_CTRL_08: + case RT5682_HP_IMP_SENS_CTRL_09: + case RT5682_HP_IMP_SENS_CTRL_10: + case RT5682_HP_IMP_SENS_CTRL_11: + case RT5682_HP_IMP_SENS_CTRL_12: + case RT5682_HP_IMP_SENS_CTRL_13: + case RT5682_HP_IMP_SENS_CTRL_14: + case RT5682_HP_IMP_SENS_CTRL_15: + case RT5682_HP_IMP_SENS_CTRL_16: + case RT5682_HP_IMP_SENS_CTRL_17: + case RT5682_HP_IMP_SENS_CTRL_18: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_HP_IMP_SENS_CTRL_20: + case RT5682_HP_IMP_SENS_CTRL_21: + case RT5682_HP_IMP_SENS_CTRL_22: + case RT5682_HP_IMP_SENS_CTRL_23: + case RT5682_HP_IMP_SENS_CTRL_24: + case RT5682_HP_IMP_SENS_CTRL_25: + case RT5682_HP_IMP_SENS_CTRL_26: + case RT5682_HP_IMP_SENS_CTRL_27: + case RT5682_HP_IMP_SENS_CTRL_28: + case RT5682_HP_IMP_SENS_CTRL_29: + case RT5682_HP_IMP_SENS_CTRL_30: + case RT5682_HP_IMP_SENS_CTRL_31: + case RT5682_HP_IMP_SENS_CTRL_32: + case RT5682_HP_IMP_SENS_CTRL_33: + case RT5682_HP_IMP_SENS_CTRL_34: + case RT5682_HP_IMP_SENS_CTRL_35: + case RT5682_HP_IMP_SENS_CTRL_36: + case RT5682_HP_IMP_SENS_CTRL_37: + case RT5682_HP_IMP_SENS_CTRL_38: + case RT5682_HP_IMP_SENS_CTRL_39: + case RT5682_HP_IMP_SENS_CTRL_40: + case RT5682_HP_IMP_SENS_CTRL_41: + case RT5682_HP_IMP_SENS_CTRL_42: + case RT5682_HP_IMP_SENS_CTRL_43: + case RT5682_HP_LOGIC_CTRL_1: + case RT5682_HP_LOGIC_CTRL_2: + case RT5682_HP_LOGIC_CTRL_3: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_HP_CALIB_CTRL_2: + case RT5682_HP_CALIB_CTRL_3: + case RT5682_HP_CALIB_CTRL_4: + case RT5682_HP_CALIB_CTRL_5: + case RT5682_HP_CALIB_CTRL_6: + case RT5682_HP_CALIB_CTRL_7: + case RT5682_HP_CALIB_CTRL_9: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_HP_CALIB_CTRL_11: + case RT5682_HP_CALIB_STA_1: + case RT5682_HP_CALIB_STA_2: + case RT5682_HP_CALIB_STA_3: + case RT5682_HP_CALIB_STA_4: + case RT5682_HP_CALIB_STA_5: + case RT5682_HP_CALIB_STA_6: + case RT5682_HP_CALIB_STA_7: + case RT5682_HP_CALIB_STA_8: + case RT5682_HP_CALIB_STA_9: + case RT5682_HP_CALIB_STA_10: + case RT5682_HP_CALIB_STA_11: + case RT5682_SAR_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_3: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_5: + case RT5682_SAR_IL_CMD_6: + case RT5682_SAR_IL_CMD_7: + case RT5682_SAR_IL_CMD_8: + case RT5682_SAR_IL_CMD_9: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_SAR_IL_CMD_12: + case RT5682_SAR_IL_CMD_13: + case RT5682_EFUSE_CTRL_1: + case RT5682_EFUSE_CTRL_2: + case RT5682_EFUSE_CTRL_3: + case RT5682_EFUSE_CTRL_4: + case RT5682_EFUSE_CTRL_5: + case RT5682_EFUSE_CTRL_6: + case RT5682_EFUSE_CTRL_7: + case RT5682_EFUSE_CTRL_8: + case RT5682_EFUSE_CTRL_9: + case RT5682_EFUSE_CTRL_10: + case RT5682_EFUSE_CTRL_11: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_DRC1_CTRL_0: + case RT5682_DRC1_CTRL_1: + case RT5682_DRC1_CTRL_2: + case RT5682_DRC1_CTRL_3: + case RT5682_DRC1_CTRL_4: + case RT5682_DRC1_CTRL_5: + case RT5682_DRC1_CTRL_6: + case RT5682_DRC1_HARD_LMT_CTRL_1: + case RT5682_DRC1_HARD_LMT_CTRL_2: + case RT5682_DRC1_PRIV_1: + case RT5682_DRC1_PRIV_2: + case RT5682_DRC1_PRIV_3: + case RT5682_DRC1_PRIV_4: + case RT5682_DRC1_PRIV_5: + case RT5682_DRC1_PRIV_6: + case RT5682_DRC1_PRIV_7: + case RT5682_DRC1_PRIV_8: + case RT5682_EQ_AUTO_RCV_CTRL1: + case RT5682_EQ_AUTO_RCV_CTRL2: + case RT5682_EQ_AUTO_RCV_CTRL3: + case RT5682_EQ_AUTO_RCV_CTRL4: + case RT5682_EQ_AUTO_RCV_CTRL5: + case RT5682_EQ_AUTO_RCV_CTRL6: + case RT5682_EQ_AUTO_RCV_CTRL7: + case RT5682_EQ_AUTO_RCV_CTRL8: + case RT5682_EQ_AUTO_RCV_CTRL9: + case RT5682_EQ_AUTO_RCV_CTRL10: + case RT5682_EQ_AUTO_RCV_CTRL11: + case RT5682_EQ_AUTO_RCV_CTRL12: + case RT5682_EQ_AUTO_RCV_CTRL13: + case RT5682_ADC_L_EQ_LPF1_A1: + case RT5682_R_EQ_LPF1_A1: + case RT5682_L_EQ_LPF1_H0: + case RT5682_R_EQ_LPF1_H0: + case RT5682_L_EQ_BPF1_A1: + case RT5682_R_EQ_BPF1_A1: + case RT5682_L_EQ_BPF1_A2: + case RT5682_R_EQ_BPF1_A2: + case RT5682_L_EQ_BPF1_H0: + case RT5682_R_EQ_BPF1_H0: + case RT5682_L_EQ_BPF2_A1: + case RT5682_R_EQ_BPF2_A1: + case RT5682_L_EQ_BPF2_A2: + case RT5682_R_EQ_BPF2_A2: + case RT5682_L_EQ_BPF2_H0: + case RT5682_R_EQ_BPF2_H0: + case RT5682_L_EQ_BPF3_A1: + case RT5682_R_EQ_BPF3_A1: + case RT5682_L_EQ_BPF3_A2: + case RT5682_R_EQ_BPF3_A2: + case RT5682_L_EQ_BPF3_H0: + case RT5682_R_EQ_BPF3_H0: + case RT5682_L_EQ_BPF4_A1: + case RT5682_R_EQ_BPF4_A1: + case RT5682_L_EQ_BPF4_A2: + case RT5682_R_EQ_BPF4_A2: + case RT5682_L_EQ_BPF4_H0: + case RT5682_R_EQ_BPF4_H0: + case RT5682_L_EQ_HPF1_A1: + case RT5682_R_EQ_HPF1_A1: + case RT5682_L_EQ_HPF1_H0: + case RT5682_R_EQ_HPF1_H0: + case RT5682_L_EQ_PRE_VOL: + case RT5682_R_EQ_PRE_VOL: + case RT5682_L_EQ_POST_VOL: + case RT5682_R_EQ_POST_VOL: + case RT5682_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5682_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum, + RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select); + +static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); + +static void rt5682_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5682_RESET, 0); + regmap_write(regmap, RT5682_I2C_MODE, 1); +} +/** + * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5682_CLK_SEL_SYS: + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src); + +static int rt5682_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); + + return btn_type; +} + +static void rt5682_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, + RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE); + } +} + +/** + * rt5682_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2) + & RT5682_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5682->jack_type = SND_JACK_HEADSET; + rt5682_enable_push_button_irq(component, true); + break; + default: + rt5682->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5682_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5682->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); + return rt5682->jack_type; +} + +static irqreturn_t rt5682_irq(int irq, void *data) +{ + struct rt5682_priv *rt5682 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5682_jd_check_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK) { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682->jd_check_work, 500); + } +} + +static int rt5682_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5682->hs_jack = hs_jack; + + return 0; +} + +static void rt5682_jack_detect_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = + container_of(work, struct rt5682_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682->component) + usleep_range(10000, 15000); + + while (!rt5682->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682->calibrate_mutex); + + val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682->jack_type == 0) { + /* jack was out, report jack type */ + rt5682->jack_type = + rt5682_headset_detect(rt5682->component, 1); + } else { + /* jack is already in, report button event */ + rt5682->jack_type = SND_JACK_HEADSET; + btn_type = rt5682_button_detect(rt5682->component); + /** + * rt5682 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5682->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + } + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + + mutex_unlock(&rt5682->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5682_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, + RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, + RT5682_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, + RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5682_div_sel(struct rt5682_priv *rt5682, + int target, const int div[], int size) +{ + int i; + + if (rt5682->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5682->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682->sysclk) + return i; + if (target * div[i + 1] > rt5682->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5682->sysclk); + return i; + } + } + + if (target * div[i] < rt5682->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5682->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, + RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int ref, val, reg, sft, mask, idx = -EINVAL; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & + RT5682_GP4_PIN_MASK; + if (w->shift == RT5682_PWR_ADC_S1F_BIT && + val == RT5682_GP4_PIN_ADCDAT2) + ref = 256 * rt5682->lrck[RT5682_AIF2]; + else + ref = 256 * rt5682->lrck[RT5682_AIF1]; + + idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682_PWR_ADC_S1F_BIT) { + reg = RT5682_PLL_TRACK_3; + sft = RT5682_ADC_OSR_SFT; + mask = RT5682_ADC_OSR_MASK; + } else { + reg = RT5682_PLL_TRACK_2; + sft = RT5682_DAC_OSR_SFT; + mask = RT5682_DAC_OSR_MASK; + } + + snd_soc_component_update_bits(component, reg, + RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + mask, idx << sft); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682_ADC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_3; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + case RT5682_DAC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_2; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER, + RT5682_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5682_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum, + RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK, + rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACL1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACR1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_R_MUTE_SFT, 1, 1); + +static int rt5682_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5682_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, + RT5682_NG2_EN_MASK, RT5682_NG2_EN); + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, + RT5682_PWR_FV1); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, + RT5682_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5682_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5682_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, + RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK, + rt5682_adcdat_pin_select, rt5682_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, + ARRAY_SIZE(rt5682_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2, + RT5682_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC, + RT5682_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix, + ARRAY_SIZE(rt5682_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, + RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, + ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5682_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC, + RT5682_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, + RT5682_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, + RT5682_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, + RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET, + RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET, + RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET, + RT5682_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "CLKDET"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, RT5682_TDM_EN); + else + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682_TDM_TX_CH_4; + val |= RT5682_TDM_RX_CH_4; + break; + case 6: + val |= RT5682_TDM_TX_CH_6; + val |= RT5682_TDM_RX_CH_6; + break; + case 8: + val |= RT5682_TDM_TX_CH_8; + val |= RT5682_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_CTRL, + RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682_TDM_CL_16; + cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682_TDM_CL_20; + cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682_TDM_CL_24; + cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682_TDM_CL_32; + cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl); + + return 0; +} + + +static int rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5682->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5682->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682_I2S1_DL_20; + len_2 |= RT5682_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682_I2S1_DL_24; + len_2 |= RT5682_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682_I2S1_DL_32; + len_2 |= RT5682_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682_I2S2_DL_8; + len_2 |= RT5682_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_DL_MASK, len_1); + if (rt5682->master[RT5682_AIF1]) { + snd_soc_component_update_bits(component, + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, + pre_div << RT5682_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_DIS); + break; + case RT5682_AIF2: + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_DL_MASK, len_2); + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK, + pre_div << RT5682_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682_I2S_BP_INV; + tdm_ctrl |= RT5682_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV | + RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682_I2S_DF_LEFT; + tdm_ctrl |= RT5682_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682_I2S_DF_PCM_A; + tdm_ctrl |= RT5682_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682_I2S_DF_PCM_B; + tdm_ctrl |= RT5682_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK | + RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK | + RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK, + tdm_ctrl | rt5682->master[dai->id]); + break; + case RT5682_AIF2: + if (rt5682->master[dai->id] == 0) + reg_val |= RT5682_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK | + RT5682_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682_SCLK_S_MCLK: + reg_val |= RT5682_SCLK_SRC_MCLK; + src = RT5682_CLK_SRC_MCLK; + break; + case RT5682_SCLK_S_PLL1: + reg_val |= RT5682_SCLK_SRC_PLL1; + src = RT5682_CLK_SRC_PLL1; + break; + case RT5682_SCLK_S_PLL2: + reg_val |= RT5682_SCLK_SRC_PLL2; + src = RT5682_CLK_SRC_PLL2; + break; + case RT5682_SCLK_S_RCCLK: + reg_val |= RT5682_SCLK_SRC_RCCLK; + src = RT5682_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, reg_val); + + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK, + src << RT5682_I2S2_SRC_SFT); + } + + rt5682->sysclk = freq; + rt5682->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5682_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5682->pll_src && freq_in == rt5682->pll_in && + freq_out == rt5682->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5682->pll_in = 0; + rt5682->pll_out = 0; + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + + rt5682->pll_in = freq_in; + rt5682->pll_out = freq_out; + rt5682->pll_src = source; + + return 0; +} + +static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, + RT5682_PWR_MB | RT5682_PWR_BG); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5682_probe(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->component = component; + + return 0; +} + +static void rt5682_remove(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_PM +static int rt5682_suspend(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + return 0; +} + +static int rt5682_resume(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} +#else +#define rt5682_suspend NULL +#define rt5682_resume NULL +#endif + +#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_tdm_slot = rt5682_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_bclk_ratio = rt5682_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5682_dai[] = { + { + .name = "rt5682-aif1", + .id = RT5682_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif1_dai_ops, + }, + { + .name = "rt5682-aif2", + .id = RT5682_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5682 = { + .probe = rt5682_probe, + .remove = rt5682_remove, + .suspend = rt5682_suspend, + .resume = rt5682_resume, + .set_bias_level = rt5682_set_bias_level, + .controls = rt5682_snd_controls, + .num_controls = ARRAY_SIZE(rt5682_snd_controls), + .dapm_widgets = rt5682_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets), + .dapm_routes = rt5682_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes), + .set_sysclk = rt5682_set_component_sysclk, + .set_pll = rt5682_set_component_pll, + .set_jack = rt5682_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5682_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5682_i2c_id[] = { + {"rt5682", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); + +static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) +{ + + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682->pdata.jd_src); + + rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5682_calibrate(struct rt5682_priv *rt5682) +{ + int value, count; + + mutex_lock(&rt5682->calibrate_mutex); + + rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); + regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); + regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5682->calibrate_mutex); + +} + +static int rt5682_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682_priv *rt5682; + int i, ret; + unsigned int val; + + rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv), + GFP_KERNEL); + + if (rt5682 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682); + + if (pdata) + rt5682->pdata = *pdata; + else + rt5682_parse_dt(rt5682, &i2c->dev); + + rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++) + rt5682->supplies[i].supply = rt5682_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + rt5682_reset(rt5682->regmap); + + rt5682_calibrate(rt5682); + + ret = regmap_register_patch(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) { + switch (rt5682->pdata.dmic1_data_pin) { + case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA); + break; + + case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5682->pdata.dmic1_clk_pin) { + case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK); + break; + + case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, + RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682->jd_check_work, + rt5682_jd_check_handler); + + mutex_init(&rt5682->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5682", rt5682); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); +} + +static void rt5682_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5682_of_match[] = { + {.compatible = "realtek,rt5682i"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5682_acpi_match[] = { + {"10EC5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); +#endif + +static struct i2c_driver rt5682_i2c_driver = { + .driver = { + .name = "rt5682", + .of_match_table = of_match_ptr(rt5682_of_match), + .acpi_match_table = ACPI_PTR(rt5682_acpi_match), + }, + .probe = rt5682_i2c_probe, + .shutdown = rt5682_i2c_shutdown, + .id_table = rt5682_i2c_id, +}; +module_i2c_driver(rt5682_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h new file mode 100644 index 000000000000..8068140ebe3f --- /dev/null +++ b/sound/soc/codecs/rt5682.h @@ -0,0 +1,1324 @@ +/* + * rt5682.h -- RT5682/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao <bardliao@realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5682_H__ +#define __RT5682_H__ + +#include <sound/rt5682.h> + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5682_RESET 0x0000 +#define RT5682_VERSION_ID 0x00fd +#define RT5682_VENDOR_ID 0x00fe +#define RT5682_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682_HP_CTRL_1 0x0002 +#define RT5682_HP_CTRL_2 0x0003 +#define RT5682_HPL_GAIN 0x0005 +#define RT5682_HPR_GAIN 0x0006 + +#define RT5682_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682_CBJ_BST_CTRL 0x000b +#define RT5682_CBJ_CTRL_1 0x0010 +#define RT5682_CBJ_CTRL_2 0x0011 +#define RT5682_CBJ_CTRL_3 0x0012 +#define RT5682_CBJ_CTRL_4 0x0013 +#define RT5682_CBJ_CTRL_5 0x0014 +#define RT5682_CBJ_CTRL_6 0x0015 +#define RT5682_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5682_DAC1_DIG_VOL 0x0019 +#define RT5682_STO1_ADC_DIG_VOL 0x001c +#define RT5682_STO1_ADC_BOOST 0x001f +#define RT5682_HP_IMP_GAIN_1 0x0022 +#define RT5682_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682_SIDETONE_CTRL 0x0024 +#define RT5682_STO1_ADC_MIXER 0x0026 +#define RT5682_AD_DA_MIXER 0x0029 +#define RT5682_STO1_DAC_MIXER 0x002a +#define RT5682_A_DAC1_MUX 0x002b +#define RT5682_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682_REC_MIXER 0x003c +#define RT5682_CAL_REC 0x0044 +#define RT5682_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5682_PWR_DIG_1 0x0061 +#define RT5682_PWR_DIG_2 0x0062 +#define RT5682_PWR_ANLG_1 0x0063 +#define RT5682_PWR_ANLG_2 0x0064 +#define RT5682_PWR_ANLG_3 0x0065 +#define RT5682_PWR_MIXER 0x0066 +#define RT5682_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5682_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682_RESET_LPF_CTRL 0x006c +#define RT5682_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5682_I2S1_SDP 0x0070 +#define RT5682_I2S2_SDP 0x0071 +#define RT5682_ADDA_CLK_1 0x0073 +#define RT5682_ADDA_CLK_2 0x0074 +#define RT5682_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682_TDM_CTRL 0x0079 +#define RT5682_TDM_ADDA_CTRL_1 0x007a +#define RT5682_TDM_ADDA_CTRL_2 0x007b +#define RT5682_DATA_SEL_CTRL_1 0x007c +#define RT5682_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5682_GLB_CLK 0x0080 +#define RT5682_PLL_CTRL_1 0x0081 +#define RT5682_PLL_CTRL_2 0x0082 +#define RT5682_PLL_TRACK_1 0x0083 +#define RT5682_PLL_TRACK_2 0x0084 +#define RT5682_PLL_TRACK_3 0x0085 +#define RT5682_PLL_TRACK_4 0x0086 +#define RT5682_PLL_TRACK_5 0x0087 +#define RT5682_PLL_TRACK_6 0x0088 +#define RT5682_PLL_TRACK_11 0x008c +#define RT5682_SDW_REF_CLK 0x008d +#define RT5682_DEPOP_1 0x008e +#define RT5682_DEPOP_2 0x008f +#define RT5682_HP_CHARGE_PUMP_1 0x0091 +#define RT5682_HP_CHARGE_PUMP_2 0x0092 +#define RT5682_MICBIAS_1 0x0093 +#define RT5682_MICBIAS_2 0x0094 +#define RT5682_PLL_TRACK_12 0x0098 +#define RT5682_PLL_TRACK_14 0x009a +#define RT5682_PLL2_CTRL_1 0x009b +#define RT5682_PLL2_CTRL_2 0x009c +#define RT5682_PLL2_CTRL_3 0x009d +#define RT5682_PLL2_CTRL_4 0x009e +#define RT5682_RC_CLK_CTRL 0x009f +#define RT5682_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5682_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5682_EQ_CTRL_1 0x00ae +#define RT5682_EQ_CTRL_2 0x00af +#define RT5682_IRQ_CTRL_1 0x00b6 +#define RT5682_IRQ_CTRL_2 0x00b7 +#define RT5682_IRQ_CTRL_3 0x00b8 +#define RT5682_IRQ_CTRL_4 0x00b9 +#define RT5682_INT_ST_1 0x00be +#define RT5682_GPIO_CTRL_1 0x00c0 +#define RT5682_GPIO_CTRL_2 0x00c1 +#define RT5682_GPIO_CTRL_3 0x00c2 +#define RT5682_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5682_MID_HP_AMP_DET 0x00d2 +#define RT5682_LOW_HP_AMP_DET 0x00d3 +#define RT5682_DELAY_BUF_CTRL 0x00d4 +#define RT5682_SV_ZCD_1 0x00d9 +#define RT5682_SV_ZCD_2 0x00da +#define RT5682_IL_CMD_1 0x00db +#define RT5682_IL_CMD_2 0x00dc +#define RT5682_IL_CMD_3 0x00dd +#define RT5682_IL_CMD_4 0x00de +#define RT5682_IL_CMD_5 0x00df +#define RT5682_IL_CMD_6 0x00e0 +#define RT5682_4BTN_IL_CMD_1 0x00e2 +#define RT5682_4BTN_IL_CMD_2 0x00e3 +#define RT5682_4BTN_IL_CMD_3 0x00e4 +#define RT5682_4BTN_IL_CMD_4 0x00e5 +#define RT5682_4BTN_IL_CMD_5 0x00e6 +#define RT5682_4BTN_IL_CMD_6 0x00e7 +#define RT5682_4BTN_IL_CMD_7 0x00e8 + +#define RT5682_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682_AJD1_CTRL 0x00f0 +#define RT5682_JD1_THD 0x00f1 +#define RT5682_JD2_THD 0x00f2 +#define RT5682_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682_DUMMY_1 0x00fa +#define RT5682_DUMMY_2 0x00fb +#define RT5682_DUMMY_3 0x00fc + +#define RT5682_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682_BIAS_CUR_CTRL_2 0x010b +#define RT5682_BIAS_CUR_CTRL_3 0x010c +#define RT5682_BIAS_CUR_CTRL_4 0x010d +#define RT5682_BIAS_CUR_CTRL_5 0x010e +#define RT5682_BIAS_CUR_CTRL_6 0x010f +#define RT5682_BIAS_CUR_CTRL_7 0x0110 +#define RT5682_BIAS_CUR_CTRL_8 0x0111 +#define RT5682_BIAS_CUR_CTRL_9 0x0112 +#define RT5682_BIAS_CUR_CTRL_10 0x0113 +#define RT5682_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5682_CHARGE_PUMP_1 0x0125 +#define RT5682_DIG_IN_CTRL_1 0x0132 +#define RT5682_PAD_DRIVING_CTRL 0x0136 +#define RT5682_SOFT_RAMP_DEPOP 0x0138 +#define RT5682_CHOP_DAC 0x013a +#define RT5682_CHOP_ADC 0x013b +#define RT5682_CALIB_ADC_CTRL 0x013c +#define RT5682_VOL_TEST 0x013f +#define RT5682_SPKVDD_DET_STA 0x0142 +#define RT5682_TEST_MODE_CTRL_1 0x0145 +#define RT5682_TEST_MODE_CTRL_2 0x0146 +#define RT5682_TEST_MODE_CTRL_3 0x0147 +#define RT5682_TEST_MODE_CTRL_4 0x0148 +#define RT5682_TEST_MODE_CTRL_5 0x0149 +#define RT5682_PLL1_INTERNAL 0x0150 +#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_STO_NG2_CTRL_1 0x0160 +#define RT5682_STO_NG2_CTRL_2 0x0161 +#define RT5682_STO_NG2_CTRL_3 0x0162 +#define RT5682_STO_NG2_CTRL_4 0x0163 +#define RT5682_STO_NG2_CTRL_5 0x0164 +#define RT5682_STO_NG2_CTRL_6 0x0165 +#define RT5682_STO_NG2_CTRL_7 0x0166 +#define RT5682_STO_NG2_CTRL_8 0x0167 +#define RT5682_STO_NG2_CTRL_9 0x0168 +#define RT5682_STO_NG2_CTRL_10 0x0169 +#define RT5682_STO1_DAC_SIL_DET 0x0190 +#define RT5682_SIL_PSV_CTRL1 0x0194 +#define RT5682_SIL_PSV_CTRL2 0x0195 +#define RT5682_SIL_PSV_CTRL3 0x0197 +#define RT5682_SIL_PSV_CTRL4 0x0198 +#define RT5682_SIL_PSV_CTRL5 0x0199 +#define RT5682_HP_IMP_SENS_CTRL_01 0x01af +#define RT5682_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5682_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5682_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5682_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5682_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5682_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5682_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5682_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5682_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5682_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5682_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5682_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5682_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5682_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5682_HP_IMP_SENS_CTRL_16 0x01be +#define RT5682_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5682_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5682_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5682_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5682_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5682_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5682_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5682_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5682_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5682_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5682_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5682_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5682_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5682_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5682_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5682_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5682_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5682_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5682_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5682_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5682_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5682_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5682_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5682_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5682_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5682_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5682_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5682_HP_LOGIC_CTRL_1 0x01da +#define RT5682_HP_LOGIC_CTRL_2 0x01db +#define RT5682_HP_LOGIC_CTRL_3 0x01dc +#define RT5682_HP_CALIB_CTRL_1 0x01de +#define RT5682_HP_CALIB_CTRL_2 0x01df +#define RT5682_HP_CALIB_CTRL_3 0x01e0 +#define RT5682_HP_CALIB_CTRL_4 0x01e1 +#define RT5682_HP_CALIB_CTRL_5 0x01e2 +#define RT5682_HP_CALIB_CTRL_6 0x01e3 +#define RT5682_HP_CALIB_CTRL_7 0x01e4 +#define RT5682_HP_CALIB_CTRL_9 0x01e6 +#define RT5682_HP_CALIB_CTRL_10 0x01e7 +#define RT5682_HP_CALIB_CTRL_11 0x01e8 +#define RT5682_HP_CALIB_STA_1 0x01ea +#define RT5682_HP_CALIB_STA_2 0x01eb +#define RT5682_HP_CALIB_STA_3 0x01ec +#define RT5682_HP_CALIB_STA_4 0x01ed +#define RT5682_HP_CALIB_STA_5 0x01ee +#define RT5682_HP_CALIB_STA_6 0x01ef +#define RT5682_HP_CALIB_STA_7 0x01f0 +#define RT5682_HP_CALIB_STA_8 0x01f1 +#define RT5682_HP_CALIB_STA_9 0x01f2 +#define RT5682_HP_CALIB_STA_10 0x01f3 +#define RT5682_HP_CALIB_STA_11 0x01f4 +#define RT5682_SAR_IL_CMD_1 0x0210 +#define RT5682_SAR_IL_CMD_2 0x0211 +#define RT5682_SAR_IL_CMD_3 0x0212 +#define RT5682_SAR_IL_CMD_4 0x0213 +#define RT5682_SAR_IL_CMD_5 0x0214 +#define RT5682_SAR_IL_CMD_6 0x0215 +#define RT5682_SAR_IL_CMD_7 0x0216 +#define RT5682_SAR_IL_CMD_8 0x0217 +#define RT5682_SAR_IL_CMD_9 0x0218 +#define RT5682_SAR_IL_CMD_10 0x0219 +#define RT5682_SAR_IL_CMD_11 0x021a +#define RT5682_SAR_IL_CMD_12 0x021b +#define RT5682_SAR_IL_CMD_13 0x021c +#define RT5682_EFUSE_CTRL_1 0x0250 +#define RT5682_EFUSE_CTRL_2 0x0251 +#define RT5682_EFUSE_CTRL_3 0x0252 +#define RT5682_EFUSE_CTRL_4 0x0253 +#define RT5682_EFUSE_CTRL_5 0x0254 +#define RT5682_EFUSE_CTRL_6 0x0255 +#define RT5682_EFUSE_CTRL_7 0x0256 +#define RT5682_EFUSE_CTRL_8 0x0257 +#define RT5682_EFUSE_CTRL_9 0x0258 +#define RT5682_EFUSE_CTRL_10 0x0259 +#define RT5682_EFUSE_CTRL_11 0x025a +#define RT5682_JD_TOP_VC_VTRL 0x0270 +#define RT5682_DRC1_CTRL_0 0x02ff +#define RT5682_DRC1_CTRL_1 0x0300 +#define RT5682_DRC1_CTRL_2 0x0301 +#define RT5682_DRC1_CTRL_3 0x0302 +#define RT5682_DRC1_CTRL_4 0x0303 +#define RT5682_DRC1_CTRL_5 0x0304 +#define RT5682_DRC1_CTRL_6 0x0305 +#define RT5682_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5682_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5682_DRC1_PRIV_1 0x0310 +#define RT5682_DRC1_PRIV_2 0x0311 +#define RT5682_DRC1_PRIV_3 0x0312 +#define RT5682_DRC1_PRIV_4 0x0313 +#define RT5682_DRC1_PRIV_5 0x0314 +#define RT5682_DRC1_PRIV_6 0x0315 +#define RT5682_DRC1_PRIV_7 0x0316 +#define RT5682_DRC1_PRIV_8 0x0317 +#define RT5682_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5682_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5682_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5682_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5682_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5682_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5682_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5682_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5682_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5682_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5682_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5682_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5682_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5682_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5682_R_EQ_LPF1_A1 0x03d1 +#define RT5682_L_EQ_LPF1_H0 0x03d2 +#define RT5682_R_EQ_LPF1_H0 0x03d3 +#define RT5682_L_EQ_BPF1_A1 0x03d4 +#define RT5682_R_EQ_BPF1_A1 0x03d5 +#define RT5682_L_EQ_BPF1_A2 0x03d6 +#define RT5682_R_EQ_BPF1_A2 0x03d7 +#define RT5682_L_EQ_BPF1_H0 0x03d8 +#define RT5682_R_EQ_BPF1_H0 0x03d9 +#define RT5682_L_EQ_BPF2_A1 0x03da +#define RT5682_R_EQ_BPF2_A1 0x03db +#define RT5682_L_EQ_BPF2_A2 0x03dc +#define RT5682_R_EQ_BPF2_A2 0x03dd +#define RT5682_L_EQ_BPF2_H0 0x03de +#define RT5682_R_EQ_BPF2_H0 0x03df +#define RT5682_L_EQ_BPF3_A1 0x03e0 +#define RT5682_R_EQ_BPF3_A1 0x03e1 +#define RT5682_L_EQ_BPF3_A2 0x03e2 +#define RT5682_R_EQ_BPF3_A2 0x03e3 +#define RT5682_L_EQ_BPF3_H0 0x03e4 +#define RT5682_R_EQ_BPF3_H0 0x03e5 +#define RT5682_L_EQ_BPF4_A1 0x03e6 +#define RT5682_R_EQ_BPF4_A1 0x03e7 +#define RT5682_L_EQ_BPF4_A2 0x03e8 +#define RT5682_R_EQ_BPF4_A2 0x03e9 +#define RT5682_L_EQ_BPF4_H0 0x03ea +#define RT5682_R_EQ_BPF4_H0 0x03eb +#define RT5682_L_EQ_HPF1_A1 0x03ec +#define RT5682_R_EQ_HPF1_A1 0x03ed +#define RT5682_L_EQ_HPF1_H0 0x03ee +#define RT5682_R_EQ_HPF1_H0 0x03ef +#define RT5682_L_EQ_PRE_VOL 0x03f0 +#define RT5682_R_EQ_PRE_VOL 0x03f1 +#define RT5682_L_EQ_POST_VOL 0x03f2 +#define RT5682_R_EQ_POST_VOL 0x03f3 +#define RT5682_I2C_MODE 0xffff + + +/* global definition */ +#define RT5682_L_MUTE (0x1 << 15) +#define RT5682_L_MUTE_SFT 15 +#define RT5682_VOL_L_MUTE (0x1 << 14) +#define RT5682_VOL_L_SFT 14 +#define RT5682_R_MUTE (0x1 << 7) +#define RT5682_R_MUTE_SFT 7 +#define RT5682_VOL_R_MUTE (0x1 << 6) +#define RT5682_VOL_R_SFT 6 +#define RT5682_L_VOL_MASK (0x3f << 8) +#define RT5682_L_VOL_SFT 8 +#define RT5682_R_VOL_MASK (0x3f) +#define RT5682_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682_G_HP (0xf << 8) +#define RT5682_G_HP_SFT 8 +#define RT5682_G_STO_DA_DMIX (0xf) +#define RT5682_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5682_BST_CBJ_MASK (0xf << 8) +#define RT5682_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5682_EMB_JD_EN (0x1 << 15) +#define RT5682_EMB_JD_EN_SFT 15 +#define RT5682_EMB_JD_RST (0x1 << 14) +#define RT5682_JD_MODE (0x1 << 13) +#define RT5682_JD_MODE_SFT 13 +#define RT5682_DET_TYPE (0x1 << 12) +#define RT5682_DET_TYPE_SFT 12 +#define RT5682_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682_EXT_JD_DIG (0x1 << 9) +#define RT5682_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682_FAST_OFF_MASK (0x1 << 7) +#define RT5682_FAST_OFF_EN (0x1 << 7) +#define RT5682_FAST_OFF_DIS (0x0 << 7) +#define RT5682_VREF_POW_MASK (0x1 << 6) +#define RT5682_VREF_POW_FSM (0x0 << 6) +#define RT5682_VREF_POW_REG (0x1 << 6) +#define RT5682_MB1_PATH_MASK (0x1 << 5) +#define RT5682_CTRL_MB1_REG (0x1 << 5) +#define RT5682_CTRL_MB1_FSM (0x0 << 5) +#define RT5682_MB2_PATH_MASK (0x1 << 4) +#define RT5682_CTRL_MB2_REG (0x1 << 4) +#define RT5682_CTRL_MB2_FSM (0x0 << 4) +#define RT5682_TRIG_JD_MASK (0x1 << 3) +#define RT5682_TRIG_JD_HIGH (0x1 << 3) +#define RT5682_TRIG_JD_LOW (0x0 << 3) +#define RT5682_MIC_CAP_MASK (0x1 << 1) +#define RT5682_MIC_CAP_HS (0x1 << 1) +#define RT5682_MIC_CAP_HP (0x0 << 1) +#define RT5682_MIC_CAP_SRC_MASK (0x1) +#define RT5682_MIC_CAP_SRC_REG (0x1) +#define RT5682_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5682_EXT_JD_SRC (0x7 << 4) +#define RT5682_EXT_JD_SRC_SFT 4 +#define RT5682_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5682_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5682_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682_DAC_L1_VOL_SFT 8 +#define RT5682_DAC_R1_VOL_MASK (0xff) +#define RT5682_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682_ADC_L_VOL_SFT 8 +#define RT5682_ADC_R_VOL_MASK (0x7f) +#define RT5682_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682_STO1_ADC_L_BST_SFT 14 +#define RT5682_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682_ST_SRC_SEL (0x1 << 8) +#define RT5682_ST_SRC_SFT 8 +#define RT5682_ST_EN_MASK (0x1 << 6) +#define RT5682_ST_DIS (0x0 << 6) +#define RT5682_ST_EN (0x1 << 6) +#define RT5682_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682_M_STO1_ADC_L1_SFT 15 +#define RT5682_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682_M_STO1_ADC_L2_SFT 14 +#define RT5682_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682_STO1_ADC1L_SRC_SFT 13 +#define RT5682_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682_STO1_ADC2L_SRC_SFT 12 +#define RT5682_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682_STO1_ADCL_SRC_SFT 10 +#define RT5682_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5682_STO1_DD_L_SRC_SFT 9 +#define RT5682_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_SFT 8 +#define RT5682_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5682_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682_M_STO1_ADC_R1_SFT 7 +#define RT5682_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682_M_STO1_ADC_R2_SFT 6 +#define RT5682_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682_STO1_ADC1R_SRC_SFT 5 +#define RT5682_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682_STO1_ADC2R_SRC_SFT 4 +#define RT5682_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682_M_ADCMIX_L (0x1 << 15) +#define RT5682_M_ADCMIX_L_SFT 15 +#define RT5682_M_DAC1_L (0x1 << 14) +#define RT5682_M_DAC1_L_SFT 14 +#define RT5682_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5682_DAC1_R_SEL_SFT 10 +#define RT5682_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5682_DAC1_L_SEL_SFT 8 +#define RT5682_M_ADCMIX_R (0x1 << 7) +#define RT5682_M_ADCMIX_R_SFT 7 +#define RT5682_M_DAC1_R (0x1 << 6) +#define RT5682_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682_M_DAC_L1_STO_L_SFT 15 +#define RT5682_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682_G_DAC_L1_STO_L_SFT 14 +#define RT5682_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682_M_DAC_R1_STO_L_SFT 13 +#define RT5682_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682_G_DAC_R1_STO_L_SFT 12 +#define RT5682_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682_M_DAC_L1_STO_R_SFT 7 +#define RT5682_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682_G_DAC_L1_STO_R_SFT 6 +#define RT5682_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682_M_DAC_R1_STO_R_SFT 5 +#define RT5682_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682_M_ST_STO_L (0x1 << 9) +#define RT5682_M_ST_STO_L_SFT 9 +#define RT5682_M_ST_STO_R (0x1 << 8) +#define RT5682_M_ST_STO_R_SFT 8 +#define RT5682_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5682_A_DACL1_SFT 4 +#define RT5682_DAC_R1_SRC_MASK (0x3) +#define RT5682_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5682_G_CBJ_RM1_L (0x7 << 10) +#define RT5682_G_CBJ_RM1_L_SFT 10 +#define RT5682_M_CBJ_RM1_L (0x1 << 7) +#define RT5682_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682_PWR_I2S1 (0x1 << 15) +#define RT5682_PWR_I2S1_BIT 15 +#define RT5682_PWR_I2S2 (0x1 << 14) +#define RT5682_PWR_I2S2_BIT 14 +#define RT5682_PWR_DAC_L1 (0x1 << 11) +#define RT5682_PWR_DAC_L1_BIT 11 +#define RT5682_PWR_DAC_R1 (0x1 << 10) +#define RT5682_PWR_DAC_R1_BIT 10 +#define RT5682_PWR_LDO (0x1 << 8) +#define RT5682_PWR_LDO_BIT 8 +#define RT5682_PWR_ADC_L1 (0x1 << 4) +#define RT5682_PWR_ADC_L1_BIT 4 +#define RT5682_PWR_ADC_R1 (0x1 << 3) +#define RT5682_PWR_ADC_R1_BIT 3 +#define RT5682_DIG_GATE_CTRL (0x1 << 0) +#define RT5682_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682_PWR_ADC_S1F (0x1 << 15) +#define RT5682_PWR_ADC_S1F_BIT 15 +#define RT5682_PWR_DAC_S1F (0x1 << 10) +#define RT5682_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682_PWR_VREF1 (0x1 << 15) +#define RT5682_PWR_VREF1_BIT 15 +#define RT5682_PWR_FV1 (0x1 << 14) +#define RT5682_PWR_FV1_BIT 14 +#define RT5682_PWR_VREF2 (0x1 << 13) +#define RT5682_PWR_VREF2_BIT 13 +#define RT5682_PWR_FV2 (0x1 << 12) +#define RT5682_PWR_FV2_BIT 12 +#define RT5682_LDO1_DBG_MASK (0x3 << 10) +#define RT5682_PWR_MB (0x1 << 9) +#define RT5682_PWR_MB_BIT 9 +#define RT5682_PWR_BG (0x1 << 7) +#define RT5682_PWR_BG_BIT 7 +#define RT5682_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682_LDO1_BYPASS (0x1 << 6) +#define RT5682_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5682_PWR_MA_BIT 6 +#define RT5682_LDO1_DVO_MASK (0x3 << 4) +#define RT5682_LDO1_DVO_09 (0x0 << 4) +#define RT5682_LDO1_DVO_10 (0x1 << 4) +#define RT5682_LDO1_DVO_12 (0x2 << 4) +#define RT5682_LDO1_DVO_14 (0x3 << 4) +#define RT5682_HP_DRIVER_MASK (0x3 << 2) +#define RT5682_HP_DRIVER_1X (0x0 << 2) +#define RT5682_HP_DRIVER_3X (0x1 << 2) +#define RT5682_HP_DRIVER_5X (0x3 << 2) +#define RT5682_PWR_HA_L (0x1 << 1) +#define RT5682_PWR_HA_L_BIT 1 +#define RT5682_PWR_HA_R (0x1 << 0) +#define RT5682_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682_PWR_MB1 (0x1 << 11) +#define RT5682_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682_PWR_MB1_BIT 11 +#define RT5682_PWR_MB2 (0x1 << 10) +#define RT5682_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682_PWR_MB2_BIT 10 +#define RT5682_PWR_JDH (0x1 << 3) +#define RT5682_PWR_JDH_BIT 3 +#define RT5682_PWR_JDL (0x1 << 2) +#define RT5682_PWR_JDL_BIT 2 +#define RT5682_PWR_RM1_L (0x1 << 1) +#define RT5682_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682_PWR_CBJ (0x1 << 9) +#define RT5682_PWR_CBJ_BIT 9 +#define RT5682_PWR_PLL (0x1 << 6) +#define RT5682_PWR_PLL_BIT 6 +#define RT5682_PWR_PLL2B (0x1 << 5) +#define RT5682_PWR_PLL2B_BIT 5 +#define RT5682_PWR_PLL2F (0x1 << 4) +#define RT5682_PWR_PLL2F_BIT 4 +#define RT5682_PWR_LDO2 (0x1 << 2) +#define RT5682_PWR_LDO2_BIT 2 +#define RT5682_PWR_DET_SPKVDD (0x1 << 1) +#define RT5682_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5682_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682_PWR_STO1_DAC_L_BIT 5 +#define RT5682_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682_SYS_CLK_DET (0x1 << 15) +#define RT5682_SYS_CLK_DET_SFT 15 +#define RT5682_PLL1_CLK_DET (0x1 << 14) +#define RT5682_PLL1_CLK_DET_SFT 14 +#define RT5682_PLL2_CLK_DET (0x1 << 13) +#define RT5682_PLL2_CLK_DET_SFT 13 +#define RT5682_POW_CLK_DET2_SFT 8 +#define RT5682_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682_DMIC_1_EN_SFT 15 +#define RT5682_DMIC_1_DIS (0x0 << 15) +#define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682_DMIC_1_DP_SFT 4 +#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682_DMIC_CLK_MASK (0xf << 0) +#define RT5682_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682_SEL_ADCDAT_SFT 15 +#define RT5682_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682_I2S1_TX_CHL_SFT 12 +#define RT5682_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682_I2S1_RX_CHL_SFT 8 +#define RT5682_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682_I2S1_MONO_MASK (0x1 << 7) +#define RT5682_I2S1_MONO_EN (0x1 << 7) +#define RT5682_I2S1_MONO_DIS (0x0 << 7) +#define RT5682_I2S2_MONO_MASK (0x1 << 6) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S1_DL_MASK (0x7 << 4) +#define RT5682_I2S1_DL_SFT 4 +#define RT5682_I2S1_DL_16 (0x0 << 4) +#define RT5682_I2S1_DL_20 (0x1 << 4) +#define RT5682_I2S1_DL_24 (0x2 << 4) +#define RT5682_I2S1_DL_32 (0x3 << 4) +#define RT5682_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5682_I2S2_MS_MASK (0x1 << 15) +#define RT5682_I2S2_MS_SFT 15 +#define RT5682_I2S2_MS_M (0x0 << 15) +#define RT5682_I2S2_MS_S (0x1 << 15) +#define RT5682_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682_I2S2_PIN_CFG_SFT 14 +#define RT5682_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5682_I2S2_CLK_SEL_SFT 11 +#define RT5682_I2S2_OUT_MASK (0x1 << 9) +#define RT5682_I2S2_OUT_SFT 9 +#define RT5682_I2S2_OUT_UM (0x0 << 9) +#define RT5682_I2S2_OUT_M (0x1 << 9) +#define RT5682_I2S_BP_MASK (0x1 << 8) +#define RT5682_I2S_BP_SFT 8 +#define RT5682_I2S_BP_NOR (0x0 << 8) +#define RT5682_I2S_BP_INV (0x1 << 8) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S2_DL_MASK (0x3 << 4) +#define RT5682_I2S2_DL_SFT 4 +#define RT5682_I2S2_DL_16 (0x0 << 4) +#define RT5682_I2S2_DL_20 (0x1 << 4) +#define RT5682_I2S2_DL_24 (0x2 << 4) +#define RT5682_I2S2_DL_8 (0x3 << 4) +#define RT5682_I2S_DF_MASK (0x7) +#define RT5682_I2S_DF_SFT 0 +#define RT5682_I2S_DF_I2S (0x0) +#define RT5682_I2S_DF_LEFT (0x1) +#define RT5682_I2S_DF_PCM_A (0x2) +#define RT5682_I2S_DF_PCM_B (0x3) +#define RT5682_I2S_DF_PCM_A_N (0x6) +#define RT5682_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682_ADC_OSR_MASK (0xf << 12) +#define RT5682_ADC_OSR_SFT 12 +#define RT5682_ADC_OSR_D_1 (0x0 << 12) +#define RT5682_ADC_OSR_D_2 (0x1 << 12) +#define RT5682_ADC_OSR_D_4 (0x2 << 12) +#define RT5682_ADC_OSR_D_6 (0x3 << 12) +#define RT5682_ADC_OSR_D_8 (0x4 << 12) +#define RT5682_ADC_OSR_D_12 (0x5 << 12) +#define RT5682_ADC_OSR_D_16 (0x6 << 12) +#define RT5682_ADC_OSR_D_24 (0x7 << 12) +#define RT5682_ADC_OSR_D_32 (0x8 << 12) +#define RT5682_ADC_OSR_D_48 (0x9 << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_SFT 8 +#define RT5682_I2S_M_D_1 (0x0 << 8) +#define RT5682_I2S_M_D_2 (0x1 << 8) +#define RT5682_I2S_M_D_3 (0x2 << 8) +#define RT5682_I2S_M_D_4 (0x3 << 8) +#define RT5682_I2S_M_D_6 (0x4 << 8) +#define RT5682_I2S_M_D_8 (0x5 << 8) +#define RT5682_I2S_M_D_12 (0x6 << 8) +#define RT5682_I2S_M_D_16 (0x7 << 8) +#define RT5682_I2S_M_D_24 (0x8 << 8) +#define RT5682_I2S_M_D_32 (0x9 << 8) +#define RT5682_I2S_M_D_48 (0x10 << 8) +#define RT5682_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5682_I2S_CLK_SRC_SFT 4 +#define RT5682_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5682_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5682_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5682_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5682_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5682_DAC_OSR_MASK (0xf << 0) +#define RT5682_DAC_OSR_SFT 0 +#define RT5682_DAC_OSR_D_1 (0x0 << 0) +#define RT5682_DAC_OSR_D_2 (0x1 << 0) +#define RT5682_DAC_OSR_D_4 (0x2 << 0) +#define RT5682_DAC_OSR_D_6 (0x3 << 0) +#define RT5682_DAC_OSR_D_8 (0x4 << 0) +#define RT5682_DAC_OSR_D_12 (0x5 << 0) +#define RT5682_DAC_OSR_D_16 (0x6 << 0) +#define RT5682_DAC_OSR_D_24 (0x7 << 0) +#define RT5682_DAC_OSR_D_32 (0x8 << 0) +#define RT5682_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682_I2S2_BCLK_MS2_SFT 11 +#define RT5682_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682_TDM_TX_CH_2 (0x0 << 12) +#define RT5682_TDM_TX_CH_4 (0x1 << 12) +#define RT5682_TDM_TX_CH_6 (0x2 << 12) +#define RT5682_TDM_TX_CH_8 (0x3 << 12) +#define RT5682_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682_TDM_RX_CH_2 (0x0 << 8) +#define RT5682_TDM_RX_CH_4 (0x1 << 8) +#define RT5682_TDM_RX_CH_6 (0x2 << 8) +#define RT5682_TDM_RX_CH_8 (0x3 << 8) +#define RT5682_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5682_TDM_ADC_LCA_SFT 4 +#define RT5682_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682_IF1_ADC1_SEL_SFT 14 +#define RT5682_IF1_ADC2_SEL_SFT 12 +#define RT5682_IF1_ADC3_SEL_SFT 10 +#define RT5682_IF1_ADC4_SEL_SFT 8 +#define RT5682_TDM_ADC_SEL_SFT 4 + +/* TDM control 3 (0x007b) */ +#define RT5682_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682_TDM_S_BP_MASK (0x1 << 15) +#define RT5682_TDM_S_BP_SFT 15 +#define RT5682_TDM_S_BP_NOR (0x0 << 15) +#define RT5682_TDM_S_BP_INV (0x1 << 15) +#define RT5682_TDM_S_LP_MASK (0x1 << 14) +#define RT5682_TDM_S_LP_SFT 14 +#define RT5682_TDM_S_LP_NOR (0x0 << 14) +#define RT5682_TDM_S_LP_INV (0x1 << 14) +#define RT5682_TDM_DF_MASK (0x7 << 11) +#define RT5682_TDM_DF_SFT 11 +#define RT5682_TDM_DF_I2S (0x0 << 11) +#define RT5682_TDM_DF_LEFT (0x1 << 11) +#define RT5682_TDM_DF_PCM_A (0x2 << 11) +#define RT5682_TDM_DF_PCM_B (0x3 << 11) +#define RT5682_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_CL_MASK (0x3 << 4) +#define RT5682_TDM_CL_16 (0x0 << 4) +#define RT5682_TDM_CL_20 (0x1 << 4) +#define RT5682_TDM_CL_24 (0x2 << 4) +#define RT5682_TDM_CL_32 (0x3 << 4) +#define RT5682_TDM_M_BP_MASK (0x1 << 2) +#define RT5682_TDM_M_BP_SFT 2 +#define RT5682_TDM_M_BP_NOR (0x0 << 2) +#define RT5682_TDM_M_BP_INV (0x1 << 2) +#define RT5682_TDM_M_LP_MASK (0x1 << 1) +#define RT5682_TDM_M_LP_SFT 1 +#define RT5682_TDM_M_LP_NOR (0x0 << 1) +#define RT5682_TDM_M_LP_INV (0x1 << 1) +#define RT5682_TDM_MS_MASK (0x1 << 0) +#define RT5682_TDM_MS_SFT 0 +#define RT5682_TDM_MS_M (0x0 << 0) +#define RT5682_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682_SCLK_SRC_MASK (0x7 << 13) +#define RT5682_SCLK_SRC_SFT 13 +#define RT5682_SCLK_SRC_MCLK (0x0 << 13) +#define RT5682_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5682_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5682_SCLK_SRC_SDW (0x3 << 13) +#define RT5682_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5682_PLL1_SRC_MASK (0x3 << 10) +#define RT5682_PLL1_SRC_SFT 10 +#define RT5682_PLL1_SRC_MCLK (0x0 << 10) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL1_SRC_SDW (0x2 << 10) +#define RT5682_PLL1_SRC_RC (0x3 << 10) +#define RT5682_PLL2_SRC_MASK (0x3 << 8) +#define RT5682_PLL2_SRC_SFT 8 +#define RT5682_PLL2_SRC_MCLK (0x0 << 8) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL2_SRC_SDW (0x2 << 8) +#define RT5682_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5682_PLL_INP_MAX 40000000 +#define RT5682_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5682_PLL_N_MAX 0x001ff +#define RT5682_PLL_N_MASK (RT5682_PLL_N_MAX << 7) +#define RT5682_PLL_N_SFT 7 +#define RT5682_PLL_K_MAX 0x001f +#define RT5682_PLL_K_MASK (RT5682_PLL_K_MAX) +#define RT5682_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5682_PLL_M_MAX 0x00f +#define RT5682_PLL_M_MASK (RT5682_PLL_M_MAX << 12) +#define RT5682_PLL_M_SFT 12 +#define RT5682_PLL_M_BP (0x1 << 11) +#define RT5682_PLL_M_BP_SFT 11 +#define RT5682_PLL_K_BP (0x1 << 10) +#define RT5682_PLL_K_BP_SFT 10 +#define RT5682_PLL_RST (0x1 << 1) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682_DA_ASRC_MASK (0x1 << 13) +#define RT5682_DA_ASRC_SFT 13 +#define RT5682_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682_DAC_STO1_ASRC_SFT 12 +#define RT5682_AD_ASRC_MASK (0x1 << 8) +#define RT5682_AD_ASRC_SFT 8 +#define RT5682_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682_AD_ASRC_SEL_SFT 4 +#define RT5682_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682_DMIC_ASRC_SFT 3 +#define RT5682_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682_ADC_STO1_ASRC_SFT 2 +#define RT5682_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682_FILTER_CLK_SEL_SFT 12 +#define RT5682_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682_ASRCIN_FTK_N1_SFT 14 +#define RT5682_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682_ASRCIN_FTK_N2_SFT 12 +#define RT5682_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682_ASRCIN_FTK_M1_SFT 8 +#define RT5682_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5682_PLL2_OUT_MASK (0x1 << 8) +#define RT5682_PLL2_OUT_98M (0x0 << 8) +#define RT5682_PLL2_OUT_49M (0x1 << 8) +#define RT5682_SDW_REF_2_MASK (0xf << 4) +#define RT5682_SDW_REF_2_SFT 4 +#define RT5682_SDW_REF_2_48K (0x0 << 4) +#define RT5682_SDW_REF_2_96K (0x1 << 4) +#define RT5682_SDW_REF_2_192K (0x2 << 4) +#define RT5682_SDW_REF_2_32K (0x3 << 4) +#define RT5682_SDW_REF_2_24K (0x4 << 4) +#define RT5682_SDW_REF_2_16K (0x5 << 4) +#define RT5682_SDW_REF_2_12K (0x6 << 4) +#define RT5682_SDW_REF_2_8K (0x7 << 4) +#define RT5682_SDW_REF_2_44K (0x8 << 4) +#define RT5682_SDW_REF_2_88K (0x9 << 4) +#define RT5682_SDW_REF_2_176K (0xa << 4) +#define RT5682_SDW_REF_2_353K (0xb << 4) +#define RT5682_SDW_REF_2_22K (0xc << 4) +#define RT5682_SDW_REF_2_384K (0xd << 4) +#define RT5682_SDW_REF_2_11K (0xe << 4) +#define RT5682_SDW_REF_1_MASK (0xf << 0) +#define RT5682_SDW_REF_1_SFT 0 +#define RT5682_SDW_REF_1_48K (0x0 << 0) +#define RT5682_SDW_REF_1_96K (0x1 << 0) +#define RT5682_SDW_REF_1_192K (0x2 << 0) +#define RT5682_SDW_REF_1_32K (0x3 << 0) +#define RT5682_SDW_REF_1_24K (0x4 << 0) +#define RT5682_SDW_REF_1_16K (0x5 << 0) +#define RT5682_SDW_REF_1_12K (0x6 << 0) +#define RT5682_SDW_REF_1_8K (0x7 << 0) +#define RT5682_SDW_REF_1_44K (0x8 << 0) +#define RT5682_SDW_REF_1_88K (0x9 << 0) +#define RT5682_SDW_REF_1_176K (0xa << 0) +#define RT5682_SDW_REF_1_353K (0xb << 0) +#define RT5682_SDW_REF_1_22K (0xc << 0) +#define RT5682_SDW_REF_1_384K (0xd << 0) +#define RT5682_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682_PUMP_EN (0x1 << 3) +#define RT5682_PUMP_EN_SFT 3 +#define RT5682_CAPLESS_EN (0x1 << 0) +#define RT5682_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682_RAMP_MASK (0x1 << 12) +#define RT5682_RAMP_SFT 12 +#define RT5682_RAMP_DIS (0x0 << 12) +#define RT5682_RAMP_EN (0x1 << 12) +#define RT5682_BPS_MASK (0x1 << 11) +#define RT5682_BPS_SFT 11 +#define RT5682_BPS_DIS (0x0 << 11) +#define RT5682_BPS_EN (0x1 << 11) +#define RT5682_FAST_UPDN_MASK (0x1 << 10) +#define RT5682_FAST_UPDN_SFT 10 +#define RT5682_FAST_UPDN_DIS (0x0 << 10) +#define RT5682_FAST_UPDN_EN (0x1 << 10) +#define RT5682_VLO_MASK (0x1 << 7) +#define RT5682_VLO_SFT 7 +#define RT5682_VLO_3V (0x0 << 7) +#define RT5682_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682_OSW_L_MASK (0x1 << 11) +#define RT5682_OSW_L_SFT 11 +#define RT5682_OSW_L_DIS (0x0 << 11) +#define RT5682_OSW_L_EN (0x1 << 11) +#define RT5682_OSW_R_MASK (0x1 << 10) +#define RT5682_OSW_R_SFT 10 +#define RT5682_OSW_R_DIS (0x0 << 10) +#define RT5682_OSW_R_EN (0x1 << 10) +#define RT5682_PM_HP_MASK (0x3 << 8) +#define RT5682_PM_HP_SFT 8 +#define RT5682_PM_HP_LV (0x0 << 8) +#define RT5682_PM_HP_MV (0x1 << 8) +#define RT5682_PM_HP_HV (0x2 << 8) +#define RT5682_IB_HP_MASK (0x3 << 6) +#define RT5682_IB_HP_SFT 6 +#define RT5682_IB_HP_125IL (0x0 << 6) +#define RT5682_IB_HP_25IL (0x1 << 6) +#define RT5682_IB_HP_5IL (0x2 << 6) +#define RT5682_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5682_MIC1_OV_MASK (0x3 << 14) +#define RT5682_MIC1_OV_SFT 14 +#define RT5682_MIC1_OV_2V7 (0x0 << 14) +#define RT5682_MIC1_OV_2V4 (0x1 << 14) +#define RT5682_MIC1_OV_2V25 (0x3 << 14) +#define RT5682_MIC1_OV_1V8 (0x4 << 14) +#define RT5682_MIC1_CLK_MASK (0x1 << 13) +#define RT5682_MIC1_CLK_SFT 13 +#define RT5682_MIC1_CLK_DIS (0x0 << 13) +#define RT5682_MIC1_CLK_EN (0x1 << 13) +#define RT5682_MIC1_OVCD_MASK (0x1 << 12) +#define RT5682_MIC1_OVCD_SFT 12 +#define RT5682_MIC1_OVCD_DIS (0x0 << 12) +#define RT5682_MIC1_OVCD_EN (0x1 << 12) +#define RT5682_MIC1_OVTH_MASK (0x3 << 10) +#define RT5682_MIC1_OVTH_SFT 10 +#define RT5682_MIC1_OVTH_768UA (0x0 << 10) +#define RT5682_MIC1_OVTH_960UA (0x1 << 10) +#define RT5682_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5682_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5682_MIC2_OV_MASK (0x3 << 8) +#define RT5682_MIC2_OV_SFT 8 +#define RT5682_MIC2_OV_2V7 (0x0 << 8) +#define RT5682_MIC2_OV_2V4 (0x1 << 8) +#define RT5682_MIC2_OV_2V25 (0x3 << 8) +#define RT5682_MIC2_OV_1V8 (0x4 << 8) +#define RT5682_MIC2_CLK_MASK (0x1 << 7) +#define RT5682_MIC2_CLK_SFT 7 +#define RT5682_MIC2_CLK_DIS (0x0 << 7) +#define RT5682_MIC2_CLK_EN (0x1 << 7) +#define RT5682_MIC2_OVTH_MASK (0x3 << 4) +#define RT5682_MIC2_OVTH_SFT 4 +#define RT5682_MIC2_OVTH_768UA (0x0 << 4) +#define RT5682_MIC2_OVTH_960UA (0x1 << 4) +#define RT5682_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5682_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5682_PWR_MB_MASK (0x1 << 3) +#define RT5682_PWR_MB_SFT 3 +#define RT5682_PWR_MB_PD (0x0 << 3) +#define RT5682_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5682_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682_PWR_CLK25M_SFT 9 +#define RT5682_PWR_CLK25M_PD (0x0 << 9) +#define RT5682_PWR_CLK25M_PU (0x1 << 9) +#define RT5682_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682_PWR_CLK1M_SFT 8 +#define RT5682_PWR_CLK1M_PD (0x0 << 8) +#define RT5682_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5682_POW_IRQ (0x1 << 15) +#define RT5682_POW_JDH (0x1 << 14) +#define RT5682_POW_JDL (0x1 << 13) +#define RT5682_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5682_CLK_SRC_MCLK (0x0) +#define RT5682_CLK_SRC_PLL1 (0x1) +#define RT5682_CLK_SRC_PLL2 (0x2) +#define RT5682_CLK_SRC_SDW (0x3) +#define RT5682_CLK_SRC_RCCLK (0x4) +#define RT5682_I2S_PD_1 (0x0) +#define RT5682_I2S_PD_2 (0x1) +#define RT5682_I2S_PD_3 (0x2) +#define RT5682_I2S_PD_4 (0x3) +#define RT5682_I2S_PD_6 (0x4) +#define RT5682_I2S_PD_8 (0x5) +#define RT5682_I2S_PD_12 (0x6) +#define RT5682_I2S_PD_16 (0x7) +#define RT5682_I2S_PD_24 (0x8) +#define RT5682_I2S_PD_32 (0x9) +#define RT5682_I2S_PD_48 (0xa) +#define RT5682_I2S2_SRC_MASK (0x3 << 4) +#define RT5682_I2S2_SRC_SFT 4 +#define RT5682_I2S2_M_PD_MASK (0xf << 0) +#define RT5682_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682_JD1_PULSE_EN_SFT 10 +#define RT5682_JD1_PULSE_DIS (0x0 << 10) +#define RT5682_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682_JD1_EN_MASK (0x1 << 15) +#define RT5682_JD1_EN_SFT 15 +#define RT5682_JD1_DIS (0x0 << 15) +#define RT5682_JD1_EN (0x1 << 15) +#define RT5682_JD1_POL_MASK (0x1 << 13) +#define RT5682_JD1_POL_NOR (0x0 << 13) +#define RT5682_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682_IL_IRQ_MASK (0x1 << 7) +#define RT5682_IL_IRQ_DIS (0x0 << 7) +#define RT5682_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682_GP1_PIN_MASK (0x3 << 14) +#define RT5682_GP1_PIN_SFT 14 +#define RT5682_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682_GP1_PIN_IRQ (0x1 << 14) +#define RT5682_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682_GP2_PIN_MASK (0x3 << 12) +#define RT5682_GP2_PIN_SFT 12 +#define RT5682_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682_GP3_PIN_MASK (0x3 << 10) +#define RT5682_GP3_PIN_SFT 10 +#define RT5682_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682_GP4_PIN_MASK (0x3 << 8) +#define RT5682_GP4_PIN_SFT 8 +#define RT5682_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682_GP5_PIN_MASK (0x3 << 6) +#define RT5682_GP5_PIN_SFT 6 +#define RT5682_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682_GP6_PIN_MASK (0x1 << 5) +#define RT5682_GP6_PIN_SFT 5 +#define RT5682_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682_GP1_PF_MASK (0x1 << 15) +#define RT5682_GP1_PF_IN (0x0 << 15) +#define RT5682_GP1_PF_OUT (0x1 << 15) +#define RT5682_GP1_OUT_MASK (0x1 << 14) +#define RT5682_GP1_OUT_L (0x0 << 14) +#define RT5682_GP1_OUT_H (0x1 << 14) +#define RT5682_GP2_PF_MASK (0x1 << 13) +#define RT5682_GP2_PF_IN (0x0 << 13) +#define RT5682_GP2_PF_OUT (0x1 << 13) +#define RT5682_GP2_OUT_MASK (0x1 << 12) +#define RT5682_GP2_OUT_L (0x0 << 12) +#define RT5682_GP2_OUT_H (0x1 << 12) +#define RT5682_GP3_PF_MASK (0x1 << 11) +#define RT5682_GP3_PF_IN (0x0 << 11) +#define RT5682_GP3_PF_OUT (0x1 << 11) +#define RT5682_GP3_OUT_MASK (0x1 << 10) +#define RT5682_GP3_OUT_L (0x0 << 10) +#define RT5682_GP3_OUT_H (0x1 << 10) +#define RT5682_GP4_PF_MASK (0x1 << 9) +#define RT5682_GP4_PF_IN (0x0 << 9) +#define RT5682_GP4_PF_OUT (0x1 << 9) +#define RT5682_GP4_OUT_MASK (0x1 << 8) +#define RT5682_GP4_OUT_L (0x0 << 8) +#define RT5682_GP4_OUT_H (0x1 << 8) +#define RT5682_GP5_PF_MASK (0x1 << 7) +#define RT5682_GP5_PF_IN (0x0 << 7) +#define RT5682_GP5_PF_OUT (0x1 << 7) +#define RT5682_GP5_OUT_MASK (0x1 << 6) +#define RT5682_GP5_OUT_L (0x0 << 6) +#define RT5682_GP5_OUT_H (0x1 << 6) +#define RT5682_GP6_PF_MASK (0x1 << 5) +#define RT5682_GP6_PF_IN (0x0 << 5) +#define RT5682_GP6_PF_OUT (0x1 << 5) +#define RT5682_GP6_OUT_MASK (0x1 << 4) +#define RT5682_GP6_OUT_L (0x0 << 4) +#define RT5682_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5682_GP6_STA (0x1 << 6) +#define RT5682_GP5_STA (0x1 << 5) +#define RT5682_GP4_STA (0x1 << 4) +#define RT5682_GP3_STA (0x1 << 3) +#define RT5682_GP2_STA (0x1 << 2) +#define RT5682_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682_SV_MASK (0x1 << 15) +#define RT5682_SV_SFT 15 +#define RT5682_SV_DIS (0x0 << 15) +#define RT5682_SV_EN (0x1 << 15) +#define RT5682_ZCD_MASK (0x1 << 10) +#define RT5682_ZCD_SFT 10 +#define RT5682_ZCD_PD (0x0 << 10) +#define RT5682_ZCD_PU (0x1 << 10) +#define RT5682_SV_DLY_MASK (0xf) +#define RT5682_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5682_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5682_ZCD_BST1_CBJ_SFT 7 +#define RT5682_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5682_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5682_ZCD_RECMIX_MASK (0x1) +#define RT5682_ZCD_RECMIX_SFT 0 +#define RT5682_ZCD_RECMIX_DIS (0x0) +#define RT5682_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682_4BTN_IL_MASK (0x1 << 15) +#define RT5682_4BTN_IL_EN (0x1 << 15) +#define RT5682_4BTN_IL_DIS (0x0 << 15) +#define RT5682_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682_4BTN_IL_NOR (0x1 << 14) +#define RT5682_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5682_JDH_RS_MASK (0x1 << 4) +#define RT5682_JDH_NO_PLUG (0x1 << 4) +#define RT5682_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682_CKXEN_DAC1_SFT 13 +#define RT5682_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682_CKXEN_ADC1_SFT 13 +#define RT5682_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682_AD2DA_LB_MASK (0x1 << 10) +#define RT5682_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682_NG2_EN_MASK (0x1 << 15) +#define RT5682_NG2_EN (0x1 << 15) +#define RT5682_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5682_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5682_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5682_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5682_SAR_BUTDET_RST (0x0 << 13) +#define RT5682_SAR_POW_MASK (0x1 << 12) +#define RT5682_SAR_POW_EN (0x1 << 12) +#define RT5682_SAR_POW_DIS (0x0 << 12) +#define RT5682_SAR_RST_MASK (0x1 << 11) +#define RT5682_SAR_RST_NORMAL (0x1 << 11) +#define RT5682_SAR_RST (0x0 << 11) +#define RT5682_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682_SAR_BYPASS_EN (0x1 << 10) +#define RT5682_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5682_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5682_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5682_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5682_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5682_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5682_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5682_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682_SAR_SOUR_MASK (0x3f) +#define RT5682_SAR_SOUR_BTN (0x3f) +#define RT5682_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5682_SCLK_S_MCLK, + RT5682_SCLK_S_PLL1, + RT5682_SCLK_S_PLL2, + RT5682_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682_PLL1_S_MCLK, + RT5682_PLL1_S_BCLK1, + RT5682_PLL1_S_RCCLK, +}; + +enum { + RT5682_AIF1, + RT5682_AIF2, + RT5682_AIFS +}; + +/* filter mask */ +enum { + RT5682_DA_STEREO1_FILTER = 0x1, + RT5682_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682_CLK_SEL_SYS, + RT5682_CLK_SEL_I2S1_ASRC, + RT5682_CLK_SEL_I2S2_ASRC, +}; + +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682_H__ */ diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c index 09451cd44f9b..85524acf3e9c 100644 --- a/sound/soc/codecs/dio2125.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -21,9 +21,9 @@ #include <linux/module.h> #include <sound/soc.h> -#define DRV_NAME "dio2125" +#define DRV_NAME "simple-amplifier" -struct dio2125 { +struct simple_amp { struct gpio_desc *gpiod_enable; }; @@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); - struct dio2125 *priv = snd_soc_component_get_drvdata(c); + struct simple_amp *priv = snd_soc_component_get_drvdata(c); int val; switch (event) { @@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, @@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTR"), }; -static const struct snd_soc_dapm_route dio2125_dapm_routes[] = { +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -static const struct snd_soc_component_driver dio2125_component_driver = { - .dapm_widgets = dio2125_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets), - .dapm_routes = dio2125_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes), +static const struct snd_soc_component_driver simple_amp_component_driver = { + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), }; -static int dio2125_probe(struct platform_device *pdev) +static int simple_amp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct dio2125 *priv; + struct simple_amp *priv; int err; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -93,28 +93,30 @@ static int dio2125_probe(struct platform_device *pdev) return err; } - return devm_snd_soc_register_component(dev, &dio2125_component_driver, + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, NULL, 0); } #ifdef CONFIG_OF -static const struct of_device_id dio2125_ids[] = { +static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, + { .compatible = "simple-audio-amplifier", }, { } }; -MODULE_DEVICE_TABLE(of, dio2125_ids); +MODULE_DEVICE_TABLE(of, simple_amp_ids); #endif -static struct platform_driver dio2125_driver = { +static struct platform_driver simple_amp_driver = { .driver = { .name = DRV_NAME, - .of_match_table = of_match_ptr(dio2125_ids), + .of_match_table = of_match_ptr(simple_amp_ids), }, - .probe = dio2125_probe, + .probe = simple_amp_probe, }; -module_platform_driver(dio2125_driver); +module_platform_driver(simple_amp_driver); -MODULE_DESCRIPTION("ASoC DIO2125 output driver"); +MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver"); MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 52f34c94ec25..ca2dfe12344e 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -7,6 +7,9 @@ * TAS5721 support: * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com> * + * TAS5707 support: + * Copyright (C) 2018 Jerome Brunet, Baylibre SAS <jbrunet@baylibre.com> + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = { .vol_reg_size = 1, }; +static const struct regmap_range tas5707_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), + regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG), +}; + +static const struct regmap_access_table tas5707_volatile_regs = { + .yes_ranges = tas5707_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas5707_volatile_regs_range), + +}; + +static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1); + +static const char * const tas5707_volume_slew_step_txt[] = { + "256", "512", "1024", "2048", +}; + +static const unsigned int tas5707_volume_slew_step_values[] = { + 3, 0, 1, 2, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum, + TAS571X_VOL_CFG_REG, 0, 0x3, + tas5707_volume_slew_step_txt, + tas5707_volume_slew_step_values); + +static const struct snd_kcontrol_new tas5707_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), + + SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum), + + BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG), + BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG), + BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG), + BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG), + BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG), + BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG), + BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG), + + BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG), + BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG), + BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG), + BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG), + BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG), + BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG), + BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG), +}; + +static const struct reg_default tas5707_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x70}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5707_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5707_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5707_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas5707_volatile_regs, +}; + +static const struct tas571x_chip tas5707_chip = { + .supply_names = tas5711_supply_names, + .num_supply_names = ARRAY_SIZE(tas5711_supply_names), + .controls = tas5707_controls, + .num_controls = ARRAY_SIZE(tas5707_controls), + .regmap_config = &tas5707_regmap_config, + .vol_reg_size = 1, +}; + static const char *const tas5717_supply_names[] = { "AVDD", "DVDD", @@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client) } static const struct of_device_id tas571x_of_match[] = { + { .compatible = "ti,tas5707", .data = &tas5707_chip, }, { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, @@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = { MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { + { "tas5707", (kernel_ulong_t) &tas5707_chip }, { "tas5711", (kernel_ulong_t) &tas5711_chip }, { "tas5717", (kernel_ulong_t) &tas5717_chip }, { "tas5719", (kernel_ulong_t) &tas5717_chip }, diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index c45677bc26ad..bd23e89cfe79 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -53,6 +53,22 @@ #define TAS571X_PWM_MUX_REG 0x25 /* 20-byte biquad registers */ +#define TAS5707_CH1_BQ0_REG 0x29 +#define TAS5707_CH1_BQ1_REG 0x2a +#define TAS5707_CH1_BQ2_REG 0x2b +#define TAS5707_CH1_BQ3_REG 0x2c +#define TAS5707_CH1_BQ4_REG 0x2d +#define TAS5707_CH1_BQ5_REG 0x2e +#define TAS5707_CH1_BQ6_REG 0x2f + +#define TAS5707_CH2_BQ0_REG 0x30 +#define TAS5707_CH2_BQ1_REG 0x31 +#define TAS5707_CH2_BQ2_REG 0x32 +#define TAS5707_CH2_BQ3_REG 0x33 +#define TAS5707_CH2_BQ4_REG 0x34 +#define TAS5707_CH2_BQ5_REG 0x35 +#define TAS5707_CH2_BQ6_REG 0x36 + #define TAS5717_CH1_BQ0_REG 0x26 #define TAS5717_CH1_BQ1_REG 0x27 #define TAS5717_CH1_BQ2_REG 0x28 diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d18ff17719cc..7396a6e5277e 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -625,25 +625,34 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Volumes */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR, FB_HPVOLL, 0x7F, 0, hpvol_scale), - SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR, FB_SPKVOLL, 0x7F, 0, spkvol_scale), - SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR, FB_DACVOLL, 0xFF, 0, dacvol_scale), - SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR, FB_ADCVOLL, 0xFF, 0, adcvol_scale), - SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR, FB_INVOLL, 0x3F, 0, invol_scale), /* INSEL */ - SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR, FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, 0, mic_boost_scale), /* Input Channel Map */ SOC_ENUM("Input Channel Map", ch_map_select_enum), + /* Mic Bias */ + SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + + /* Headphone Auto Switching */ + SOC_SINGLE("Headphone Auto Switching Switch", + R_CTL, FB_CTL_HPSWEN, 1, 0), + SOC_SINGLE("Headphone Detect Polarity Toggle Switch", + R_CTL, FB_CTL_HPSWPOL, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), @@ -733,9 +742,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0), SOC_SINGLE("Comp Switch", R_CLECTL, FB_CLECTL_COMP_EN, 1, 0), - SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("CLE Make-Up Gain Volume", R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("Comp Thresh Playback Volume", + SOC_SINGLE_TLV("Comp Thresh Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), @@ -766,9 +775,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC1 Phase Invert Switch", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), @@ -778,9 +787,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC2 Phase Invert Switch", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), @@ -790,9 +799,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC3 Phase Invert Switch", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index 814c8f3c4a68..6b3a21081635 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -34,6 +34,7 @@ enum { #define R_DACSR 0x19 #define R_PWRM1 0x1A #define R_PWRM2 0x1B +#define R_CTL 0x1C #define R_CONFIG0 0x1F #define R_CONFIG1 0x20 #define R_DMICCTL 0x24 @@ -1110,6 +1111,13 @@ enum { #define RV_PWRM2_VREF_DISABLE \ RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) +/****************************** + * R_CTL (0x1C) * + ******************************/ + +/* Fiel Offsets */ +#define FB_CTL_HPSWEN 7 +#define FB_CTL_HPSWPOL 6 /****************************** * R_CONFIG0 (0x1F) * diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bfd1abd72253..94675da514c8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component, case TWL6040_REG_HFRCTL: return priv->dl2_unmuted; default: - return 1; + return true; } } diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 3663b9fd4d65..deff65161504 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { static int wm2200_probe(struct snd_soc_component *component) { struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component); - int ret; wm2200->component = component; - ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2); - if (ret != 0) - return ret; - - return ret; + return 0; } static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a01a0c0e01eb..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 00c735c585d9..b0789a03d699 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), +WM_ADSP_FW_CONTROL("DSP4", 3), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7b8b6ef2f632..6cb3c153ba19 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg) case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: case WM8903_DC_SERVO_READBACK_4: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f13ef334c0d7..1965635ec07c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; @@ -2023,8 +2024,9 @@ static void wm8904_handle_pdata(struct snd_soc_component *component) wm8904_get_drc_enum, wm8904_put_drc_enum); /* We need an array of texts for the enum API */ - wm8904->drc_texts = kmalloc(sizeof(char *) - * pdata->num_drc_cfgs, GFP_KERNEL); + wm8904->drc_texts = kmalloc_array(pdata->num_drc_cfgs, + sizeof(char *), + GFP_KERNEL); if (!wm8904->drc_texts) return; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ba44e3d6c1e0..cd204f79647d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d495220fa25..108e8bf42a34 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -932,8 +932,9 @@ void wm8958_dsp2_init(struct snd_soc_component *component) }; /* We need an array of texts for the enum API */ - wm8994->mbc_texts = kmalloc(sizeof(char *) - * pdata->num_mbc_cfgs, GFP_KERNEL); + wm8994->mbc_texts = kmalloc_array(pdata->num_mbc_cfgs, + sizeof(char *), + GFP_KERNEL); if (!wm8994->mbc_texts) return; @@ -957,8 +958,9 @@ void wm8958_dsp2_init(struct snd_soc_component *component) }; /* We need an array of texts for the enum API */ - wm8994->vss_texts = kmalloc(sizeof(char *) - * pdata->num_vss_cfgs, GFP_KERNEL); + wm8994->vss_texts = kmalloc_array(pdata->num_vss_cfgs, + sizeof(char *), + GFP_KERNEL); if (!wm8994->vss_texts) return; @@ -983,8 +985,9 @@ void wm8958_dsp2_init(struct snd_soc_component *component) }; /* We need an array of texts for the enum API */ - wm8994->vss_hpf_texts = kmalloc(sizeof(char *) - * pdata->num_vss_hpf_cfgs, GFP_KERNEL); + wm8994->vss_hpf_texts = kmalloc_array(pdata->num_vss_hpf_cfgs, + sizeof(char *), + GFP_KERNEL); if (!wm8994->vss_hpf_texts) return; @@ -1010,8 +1013,9 @@ void wm8958_dsp2_init(struct snd_soc_component *component) }; /* We need an array of texts for the enum API */ - wm8994->enh_eq_texts = kmalloc(sizeof(char *) - * pdata->num_enh_eq_cfgs, GFP_KERNEL); + wm8994->enh_eq_texts = kmalloc_array(pdata->num_enh_eq_cfgs, + sizeof(char *), + GFP_KERNEL); if (!wm8994->enh_eq_texts) return; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c30f5aa392c6..8dc1f3d6a988 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } + /* fall through */ default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index f70f563d59f3..68b4cadc308f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a11e9d6bf950..efd8910b1ff7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 411b9eee88c2..457bc437ce54 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6e9e32a07259..14f1b0c0d286 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } + break; default: return -EINVAL; @@ -3298,8 +3299,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) }; /* We need an array of texts for the enum API */ - wm8994->drc_texts = devm_kzalloc(wm8994->hubs.component->dev, - sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); + wm8994->drc_texts = devm_kcalloc(wm8994->hubs.component->dev, + pdata->num_drc_cfgs, sizeof(char *), GFP_KERNEL); if (!wm8994->drc_texts) return; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 60e227832331..68c99fe37097 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d9d206046f8c..91711f8958c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg) case WM8996_RIGHT_PDM_SPEAKER: case WM8996_PDM_SPEAKER_MUTE_SEQUENCE: case WM8996_PDM_SPEAKER_VOLUME: - return 1; + return true; default: - return 0; + return false; } } @@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) case WM8996_MIC_DETECT_3: case WM8996_HEADPHONE_DETECT_1: case WM8996_HEADPHONE_DETECT_2: - return 1; + return true; default: - return 0; + return false; } } @@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + /* fall through */ case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5a0ea7b3c149..399255d1f78a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08dc82770273..1c12c78dbcce 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -418,7 +418,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = { { .id = SND_AUDIOCODEC_BESPOKE, .desc = { - .max_ch = 1, + .max_ch = 8, .sample_rates = { 16000 }, .num_sample_rates = 1, .formats = SNDRV_PCM_FMTBIT_S16_LE, @@ -627,22 +627,21 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, if (!root) goto err; - if (!debugfs_create_bool("booted", S_IRUGO, root, &dsp->booted)) + if (!debugfs_create_bool("booted", 0444, root, &dsp->booted)) goto err; - if (!debugfs_create_bool("running", S_IRUGO, root, &dsp->running)) + if (!debugfs_create_bool("running", 0444, root, &dsp->running)) goto err; - if (!debugfs_create_x32("fw_id", S_IRUGO, root, &dsp->fw_id)) + if (!debugfs_create_x32("fw_id", 0444, root, &dsp->fw_id)) goto err; - if (!debugfs_create_x32("fw_version", S_IRUGO, root, - &dsp->fw_id_version)) + if (!debugfs_create_x32("fw_version", 0444, root, &dsp->fw_id_version)) goto err; for (i = 0; i < ARRAY_SIZE(wm_adsp_debugfs_fops); ++i) { if (!debugfs_create_file(wm_adsp_debugfs_fops[i].name, - S_IRUGO, root, dsp, + 0444, root, dsp, &wm_adsp_debugfs_fops[i].fops)) goto err; } @@ -685,8 +684,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) } #endif -static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -696,9 +695,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, return 0; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_get); -static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -722,8 +722,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_put); -static const struct soc_enum wm_adsp_fw_enum[] = { +const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), @@ -732,24 +733,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; - -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6], - wm_adsp_fw_get, wm_adsp_fw_put), -}; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -1344,6 +1328,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; int skip = 0; + if (dsp->component->name_prefix) + avail -= strlen(dsp->component->name_prefix) + 1; + if (subname_len > avail) skip = subname_len - avail; @@ -1605,6 +1592,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp, if (ret) return -EINVAL; break; + case WMFW_CTL_TYPE_HOST_BUFFER: + ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk, + WMFW_CTL_FLAG_SYS | + WMFW_CTL_FLAG_VOLATILE | + WMFW_CTL_FLAG_READABLE, + 0); + if (ret) + return -EINVAL; + break; default: adsp_err(dsp, "Unknown control type: %d\n", coeff_blk.ctl_type); @@ -1872,9 +1868,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, } static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, + const struct wm_adsp_region *mem, unsigned int pos, unsigned int len) { void *alg; + unsigned int reg; int ret; __be32 val; @@ -1889,7 +1887,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } /* Read the terminator first to validate the length */ - ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val)); + reg = wm_adsp_region_to_reg(mem, pos + len); + + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list end: %d\n", ret); @@ -1898,13 +1898,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, if (be32_to_cpu(val) != 0xbedead) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", - pos + len, be32_to_cpu(val)); + reg, be32_to_cpu(val)); - alg = kzalloc(len * 2, GFP_KERNEL | GFP_DMA); + /* Convert length from DSP words to bytes */ + len *= sizeof(u32); + + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); - ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2); + reg = wm_adsp_region_to_reg(mem, pos); + + ret = regmap_raw_read(dsp->regmap, reg, alg, len); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list: %d\n", ret); kfree(alg); @@ -2003,10 +2008,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp1_id) / 2; - len = (sizeof(*adsp1_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp1_id) / sizeof(u32); + len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32); - adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp1_alg)) return PTR_ERR(adsp1_alg); @@ -2114,10 +2120,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp2_id) / 2; - len = (sizeof(*adsp2_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp2_id) / sizeof(u32); + len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32); - adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp2_alg)) return PTR_ERR(adsp2_alg); @@ -2868,9 +2875,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp dsp->component = component; - return snd_soc_add_component_controls(component, - &wm_adsp_fw_controls[dsp->num - 1], - 1); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_component_probe); @@ -3193,7 +3198,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; struct wm_adsp *dsp = buf->dsp; @@ -3232,6 +3237,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) return 0; } +static struct wm_coeff_ctl * +wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + + list_for_each_entry(ctl, &dsp->ctl_list, list) { + if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER) + continue; + + if (!ctl->enabled) + continue; + + return ctl; + } + + return NULL; +} + +static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + unsigned int reg; + u32 val; + int i, ret; + + ctl = wm_adsp_find_host_buffer_ctrl(buf); + if (!ctl) + return wm_adsp_legacy_host_buf_addr(buf); + + ret = wm_coeff_base_reg(ctl, ®); + if (ret) + return ret; + + for (i = 0; i < 5; ++i) { + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); + if (ret < 0) + return ret; + + if (val) + break; + + usleep_range(1000, 2000); + } + + if (!val) + return -EIO; + + buf->host_buf_ptr = be32_to_cpu(val); + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + + return 0; +} + static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) { const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index bc6d359f0533..8d58cb9d9bb9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -121,7 +121,11 @@ struct wm_adsp { .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +#define WM_ADSP_FW_CONTROL(dspname, num) \ + SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \ + wm_adsp_fw_get, wm_adsp_fw_put) + +extern const struct soc_enum wm_adsp_fw_enum[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); @@ -144,6 +148,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index ec78b9da020f..0c3f50acb8b1 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -29,6 +29,7 @@ /* Non-ALSA coefficient types start at 0x1000 */ #define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */ #define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */ +#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */ struct wmfw_header { char magic[4]; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 807040bb3921..a3206e65e5e5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * rate is lowered. */ inv_fs = true; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1f96c9dbe9c4..f70db8412c7c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) handled_mask |= XUNDRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) @@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) handled_mask |= ROVRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) @@ -1868,8 +1860,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->num_serializer = pdata->num_serializer; #ifdef CONFIG_PM_SLEEP - mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, - sizeof(u32) * mcasp->num_serializer, + mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev, + mcasp->num_serializer, sizeof(u32), GFP_KERNEL); if (!mcasp->context.xrsr_regs) { ret = -ENOMEM; @@ -2004,13 +1996,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev) * bytes. */ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = - devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 1), + devm_kcalloc(mcasp->dev, + 32 + mcasp->num_serializer - 1, + sizeof(unsigned int), GFP_KERNEL); mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = - devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 1), + devm_kcalloc(mcasp->dev, + 32 + mcasp->num_serializer - 1, + sizeof(unsigned int), GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750aa3637..44433b20435c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/clk.h> #include <linux/i2c.h> @@ -199,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); - snd_mask_set(mask, (__force int)priv->asrc_format); + snd_mask_set_format(mask, priv->asrc_format); return 0; } diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index adfb8135d739..528e8b108422 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/clk.h> #include <linux/delay.h> diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index d558dd5499a5..c60075112570 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -1,13 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_asrc.h - Freescale ASRC ALSA SoC header file * * Copyright (C) 2014 Freescale Semiconductor, Inc. * * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_ASRC_H diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 565e16d8fe85..1033ac6631b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Platform (DMA) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/dma-mapping.h> #include <linux/module.h> diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 8f43110373b8..c1d1d06783e5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; + /* fall through */ case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87b61bf..740b90df44bb 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1bacfa24ba7f..1255dfe19eef 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -142,7 +142,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) if (!ssi_dbg->dbg_dir) return -ENOMEM; - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, + ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); if (!ssi_dbg->dbg_stats) { diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7592b0406370..7f0fa4b52223 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -1,14 +1,10 @@ -/** - * Freescale ALSA SoC Machine driver utility - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ALSA SoC Machine driver utility +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2010 Freescale Semiconductor, Inc. #include <linux/module.h> #include <linux/of_address.h> diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index 1687b66ef18e..c5dc2a14b492 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -1,13 +1,10 @@ -/** +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Freescale ALSA SoC Machine driver utility * * Author: Timur Tabi <timur@freescale.com> * * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_UTILS_H diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..c29200cf755a 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -1,14 +1,7 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. #include <linux/module.h> #include <linux/of.h> diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1b6164249341..2094d2c8919f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -1,15 +1,12 @@ -/* - * ASoC audio graph sound card support - * - * Copyright (C) 2016 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph sound card support +// +// Copyright (C) 2016 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/gpio.h> @@ -21,7 +18,6 @@ #include <linux/of_graph.h> #include <linux/platform_device.h> #include <linux/string.h> -#include <sound/jack.h> #include <sound/simple_card_utils.h> struct graph_card_data { @@ -32,6 +28,8 @@ struct graph_card_data { unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct gpio_desc *pa_gpio; }; @@ -278,6 +276,22 @@ static int asoc_graph_get_dais_count(struct device *dev) return count; } +static int asoc_graph_soc_card_probe(struct snd_soc_card *card) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; @@ -296,8 +310,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (num == 0) return -EINVAL; - dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL); - dai_link = devm_kzalloc(dev, sizeof(*dai_link) * num, GFP_KERNEL); + dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; @@ -319,6 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->num_links = num; card->dapm_widgets = asoc_graph_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index a967aa143d51..92882e392d6c 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -1,17 +1,14 @@ -/* - * ASoC audio graph SCU sound card support - * - * Copyright (C) 2017 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on - * ${LINUX}/sound/soc/generic/simple-scu-card.c - * ${LINUX}/sound/soc/generic/audio-graph-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph SCU sound card support +// +// Copyright (C) 2017 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on +// ${LINUX}/sound/soc/generic/simple-scu-card.c +// ${LINUX}/sound/soc/generic/audio-graph-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/gpio.h> @@ -348,8 +345,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (num == 0) return -EINVAL; - dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL); - dai_link = devm_kzalloc(dev, sizeof(*dai_link) * num, GFP_KERNEL); + dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3751a07de6aa..d3f3f0fec74c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,16 +1,17 @@ -/* - * simple-card-utils.c - * - * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// simple-card-utils.c +// +// Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_gpio.h> #include <linux/of_graph.h> +#include <sound/jack.h> #include <sound/simple_card_utils.h> void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, @@ -419,6 +420,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) +{ + struct device *dev = card->dev; + enum of_gpio_flags flags; + char prop[128]; + char *pin_name; + char *gpio_name; + int mask; + int det; + + if (!prefix) + prefix = ""; + + sjack->gpio.gpio = -ENOENT; + + if (is_hp) { + snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + pin_name = "Headphones"; + gpio_name = "Headphone detection"; + mask = SND_JACK_HEADPHONE; + } else { + snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + pin_name = "Mic Jack"; + gpio_name = "Mic detection"; + mask = SND_JACK_MICROPHONE; + } + + det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); + if (det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (gpio_is_valid(det)) { + sjack->pin.pin = pin_name; + sjack->pin.mask = mask; + + sjack->gpio.name = gpio_name; + sjack->gpio.report = mask; + sjack->gpio.gpio = det; + sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.debounce_time = 150; + + snd_soc_card_jack_new(card, pin_name, mask, + &sjack->jack, + &sjack->pin, 1); + + snd_soc_jack_add_gpios(&sjack->jack, 1, + &sjack->gpio); + } + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4a516c428b3d..64bf3560c1d1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -1,32 +1,20 @@ -/* - * ASoC simple sound card support - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple sound card support +// +// Copyright (C) 2012 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk.h> #include <linux/device.h> -#include <linux/gpio.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> -#include <sound/jack.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> -struct asoc_simple_jack { - struct snd_soc_jack jack; - struct snd_soc_jack_pin pin; - struct snd_soc_jack_gpio gpio; -}; - struct simple_card_data { struct snd_soc_card snd_card; struct simple_dai_props { @@ -49,61 +37,6 @@ struct simple_card_data { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -#define asoc_simple_card_init_hp(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 0, prefix) -static int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) -{ - struct device *dev = card->dev; - enum of_gpio_flags flags; - char prop[128]; - char *pin_name; - char *gpio_name; - int mask; - int det; - - sjack->gpio.gpio = -ENOENT; - - if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); - pin_name = "Headphones"; - gpio_name = "Headphone detection"; - mask = SND_JACK_HEADPHONE; - } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); - pin_name = "Mic Jack"; - gpio_name = "Mic detection"; - mask = SND_JACK_MICROPHONE; - } - - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; - - if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; - - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; - - snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); - - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); - } - - return 0; -} - static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -213,14 +146,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); - if (ret < 0) - return ret; - return 0; } @@ -340,8 +265,8 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node, if (n <= 0) return -EINVAL; - card->aux_dev = devm_kzalloc(dev, - n * sizeof(*card->aux_dev), GFP_KERNEL); + card->aux_dev = devm_kcalloc(dev, + n, sizeof(*card->aux_dev), GFP_KERNEL); if (!card->aux_dev) return -ENOMEM; @@ -414,6 +339,22 @@ card_parse_end: return ret; } +static int asoc_simple_soc_card_probe(struct snd_soc_card *card) +{ + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -435,8 +376,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL); - dai_link = devm_kzalloc(dev, sizeof(*dai_link) * num, GFP_KERNEL); + dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; @@ -449,6 +390,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = num; + card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 48606c63562a..16a83bc51e0e 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -1,15 +1,12 @@ -/* - * ASoC simple SCU sound card support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple SCU sound card support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/module.h> @@ -246,8 +243,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) num = of_get_child_count(np); - dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL); - dai_link = devm_kzalloc(dev, sizeof(*dai_link) * num, GFP_KERNEL); + dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index d7fbb0a0a28b..388cefd7340a 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -509,8 +509,8 @@ static int img_i2s_in_probe(struct platform_device *pdev) pm_runtime_put(&pdev->dev); - i2s->suspend_ch_ctl = devm_kzalloc(dev, - sizeof(*i2s->suspend_ch_ctl) * i2s->max_i2s_chan, GFP_KERNEL); + i2s->suspend_ch_ctl = devm_kcalloc(dev, + i2s->max_i2s_chan, sizeof(*i2s->suspend_ch_ctl), GFP_KERNEL); if (!i2s->suspend_ch_ctl) { ret = -ENOMEM; goto err_suspend; diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 30a95bcef2db..fc2d1dac6333 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -479,8 +479,8 @@ static int img_i2s_out_probe(struct platform_device *pdev) return PTR_ERR(i2s->clk_ref); } - i2s->suspend_ch_ctl = devm_kzalloc(dev, - sizeof(*i2s->suspend_ch_ctl) * i2s->max_i2s_chan, GFP_KERNEL); + i2s->suspend_ch_ctl = devm_kcalloc(dev, + i2s->max_i2s_chan, sizeof(*i2s->suspend_ch_ctl), GFP_KERNEL); if (!i2s->suspend_ch_ctl) return -ENOMEM; diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 6a8b253c58d2..5455d6e0ab53 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id, stream->cumm_bytes += bytes; dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = ((void __iomem *)(ctx->mailbox + ctx->tstamp)) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); fw_tstamp.bytes_copied = stream->cumm_bytes; dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", fw_tstamp.bytes_copied, bytes); - addr = ((void *)(ctx->mailbox + ctx->tstamp)) + - (str_id * sizeof(fw_tstamp)); offset = offsetof(struct snd_sst_tstamp, bytes_copied); sst_shim_write(addr, offset, fw_tstamp.bytes_copied); return 0; @@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, struct snd_sst_tstamp fw_tstamp = {0,}; struct stream_info *stream; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; + + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); stream = get_stream_info(ctx, str_id); if (!stream) @@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) struct snd_sst_tstamp fw_tstamp; unsigned int str_id; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; str_id = info->str_id; stream = get_stream_info(ctx, str_id); @@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) return -EINVAL; substream = stream->pcm_substream; - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - + (str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); + return sst_calc_tstamp(ctx, info, substream, &fw_tstamp); } diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index a686eef2cf7f..27413ebae956 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count) /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __iowrite32_copy(dst, src, count / 4); } void memcpy32_fromio(void *dst, const void __iomem *src, int count) { - /* __iowrite32_copy uses 32-bit count values so divide by 4 for + /* __ioread32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __ioread32_copy(dst, src, count / 4); } /** diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..cccda87f4b34 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..87ef8b4058e5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 6ea360f33575..efcfd906c856 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 40eb979d5ac1..6f052fc8d1e2 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); @@ -586,7 +600,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { - .name = "bxt_da7219_max98357a_i2s", + .name = "bxt_da7219_max98357a", .pm = &snd_soc_pm_ops, }, }; @@ -599,4 +613,4 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>"); MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s"); +MODULE_ALIAS("platform:bxt_da7219_max98357a"); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index b68c289558a8..27308337ab12 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP5 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 33065ba294a9..d32844f94d74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_TH_1500UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), @@ -464,12 +464,38 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Chuwi Vi10 (CWI505) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S165"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), @@ -536,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 2 8 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), @@ -549,6 +588,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_DIFF_MIC | BYT_RT5640_MCLK_EN), }, + { /* Nuvison/TMax TM800W560 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 987720e203f9..f8a68bdb3885 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,8 +26,12 @@ #include <linux/clk.h> #include <linux/device.h> #include <linux/dmi.h> +#include <linux/input.h> +#include <linux/gpio/consumer.h> +#include <linux/gpio/machine.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> +#include <asm/intel-family.h> #include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -42,8 +46,6 @@ enum { BYT_RT5651_IN1_MAP, BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, - BYT_RT5651_IN1_HS_IN3_MAP, - BYT_RT5651_IN2_HS_IN3_MAP, }; enum { @@ -76,21 +78,26 @@ enum { #define BYT_RT5651_SSP2_AIF2 BIT(19) /* default is using AIF1 */ #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_HP_LR_SWAPPED BIT(22) +#define BYT_RT5651_MONO_SPEAKER BIT(23) + +#define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ + BYT_RT5651_JD1_1 | \ + BYT_RT5651_OVCD_TH_2000UA | \ + BYT_RT5651_OVCD_SF_0P75) /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */ #define MAX_NO_PROPS 5 struct byt_rt5651_private { struct clk *mclk; + struct gpio_desc *ext_amp_gpio; struct snd_soc_jack jack; }; /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ -static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | - BYT_RT5651_IN2_HS_IN3_MAP; +static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP; static void log_quirks(struct device *dev) { @@ -100,10 +107,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) dev_info(dev, "quirk IN2_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) - dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) - dev_info(dev, "quirk IN2_HS_IN3_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) + dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -124,6 +129,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled\n"); if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); } #define BYT_CODEC_DAI1 "rt5651-aif1" @@ -211,6 +218,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->ext_amp_gpio, 1); + else + gpiod_set_value_cansleep(priv->ext_amp_gpio, 0); + + return 0; +} + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -220,7 +241,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - + SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0, + rt5651_ext_amp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -228,6 +251,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Speaker", NULL, "Ext Amp Power"}, {"Line In", NULL, "Platform Clock"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ @@ -241,38 +265,26 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { - {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN2P", NULL, "Headset Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Headset Mic"}, - {"IN2P", NULL, "Internal Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Internal Mic"}, {"IN2P", NULL, "Internal Mic"}, {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { +static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, {"IN2P", NULL, "Internal Mic"}, {"IN3P", NULL, "Headset Mic"}, }; @@ -357,46 +369,72 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id) static const struct dmi_system_id byt_rt5651_quirk_table[] = { { + /* Chuwi Hi8 Pro (CWI513) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), }, - .driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { + /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "ADI"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), + }, + { + /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6 + * (these all use the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all of itWORKS.G.WI71C.JGBMRBA, + * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */ + DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), }, { + /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, { - /* Chuwi Vi8 Plus (CWI519) */ + /* Minnowboard Max B3 */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), - DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), + DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_MAP), + }, + { + /* Minnowboard Turbot */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ADI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), }, .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | - BYT_RT5651_IN2_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { /* VIOS LTH17 */ @@ -405,11 +443,24 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | BYT_RT5651_JD1_1 | BYT_RT5651_OVCD_TH_2000UA | BYT_RT5651_OVCD_SF_1P0 | - BYT_RT5651_IN1_IN2_MAP), + BYT_RT5651_MCLK_EN), + }, + { + /* Yours Y8W81 (and others using the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all devs with a W86C mainboard */ + DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), }, {} }; @@ -418,15 +469,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) +static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) { struct property_entry props[MAX_NO_PROPS] = {}; - struct device *i2c_dev; - int ret, cnt = 0; - - i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name); - if (!i2c_dev) - return -EPROBE_DEFER; + int cnt = 0; props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -440,10 +486,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en"); - ret = device_add_properties(i2c_dev, props); - put_device(i2c_dev); - - return ret; + return device_add_properties(i2c_dev, props); } static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) @@ -475,14 +518,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); break; - case BYT_RT5651_IN1_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in1_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); - break; - case BYT_RT5651_IN2_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in2_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -546,13 +581,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET, &priv->jack, - bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); + SND_JACK_HEADSET | SND_JACK_BTN_0, + &priv->jack, bytcr_jack_pins, + ARRAY_SIZE(bytcr_jack_pins)); if (ret) { dev_err(runtime->dev, "jack creation failed %d\n", ret); return ret; } + snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, + KEY_PLAYPAUSE); + ret = snd_soc_component_set_jack(codec, &priv->jack, NULL); if (ret) return ret; @@ -691,6 +730,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { }; /* SoC card */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; +static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ +static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ + +static int byt_rt5651_suspend(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "disabling jack detect before suspend\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static int byt_rt5651_resume(struct snd_soc_card *card) +{ + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "re-enabling jack detect after resume\n"); + snd_soc_component_set_jack(component, &priv->jack, NULL); + break; + } + } + + return 0; +} + static struct snd_soc_card byt_rt5651_card = { .name = "bytcr-rt5651", .owner = THIS_MODULE, @@ -701,23 +782,86 @@ static struct snd_soc_card byt_rt5651_card = { .dapm_routes = byt_rt5651_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map), .fully_routed = true, + .suspend_pre = byt_rt5651_suspend, + .resume_post = byt_rt5651_resume, }; -static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ -static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */ +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + {} +}; + +static const struct x86_cpu_id cherrytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */ + {} +}; + +static const struct acpi_gpio_params first_gpio = { 0, 0, false }; +static const struct acpi_gpio_params second_gpio = { 1, 0, false }; + +static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { + { "ext-amp-enable-gpios", &first_gpio, 1 }, + { }, +}; -static bool is_valleyview(void) +static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { + { "ext-amp-enable-gpios", &second_gpio, 1 }, + { }, +}; + +/* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other + * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the + * GpioIo one for the ext-amp-enable-gpio and both count for the index in + * acpi_gpio_params index. So we have 2 different mappings and the code + * below figures out which one to use. + */ +struct byt_rt5651_acpi_resource_data { + int gpio_count; + int gpio_int_idx; +}; + +static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) { - static const struct x86_cpu_id cpu_ids[] = { - { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ - {} - }; - - if (!x86_match_cpu(cpu_ids)) - return false; - return true; + struct byt_rt5651_acpi_resource_data *data = arg; + + if (ares->type != ACPI_RESOURCE_TYPE_GPIO) + return 0; + + if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) + data->gpio_int_idx = data->gpio_count; + + data->gpio_count++; + return 0; +} + +static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) +{ + struct byt_rt5651_acpi_resource_data data = { 0, -1 }; + LIST_HEAD(resources); + int ret; + + ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, + snd_byt_rt5651_acpi_resource, &data); + if (ret < 0) { + dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); + return; + } + + /* All info we need is gathered during the walk */ + acpi_dev_free_resource_list(&resources); + + switch (data.gpio_int_idx) { + case 0: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second); + break; + case 1: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first); + break; + default: + dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", + data.gpio_int_idx); + } } struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ @@ -727,13 +871,12 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = - { "dmic", "in1", "in2", "in12", "in1", "in2" }; - const char * const hsmic_name[] = - { "in2", "in2", "in1", "in3", "in3", "in3" }; + const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; + struct device *codec_dev; const char *i2c_name = NULL; + const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -767,11 +910,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) "%s%s", "i2c-", i2c_name); byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name; + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + byt_rt5651_codec_name); + if (!codec_dev) + return -EPROBE_DEFER; + /* * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) */ - if (is_valleyview()) { + if (x86_match_cpu(baytrail_cpu_ids)) { struct sst_platform_info *p_info = mach->pdata; const struct sst_res_info *res_info = p_info->res_info; @@ -830,9 +978,37 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) dmi_check_system(byt_rt5651_quirk_table); /* Must be called before register_card, also see declaration comment. */ - ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name); - if (ret_val) + ret_val = byt_rt5651_add_codec_device_props(codec_dev); + if (ret_val) { + put_device(codec_dev); return ret_val; + } + + /* Cherry Trail devices use an external amplifier enable gpio */ + if (x86_match_cpu(cherrytrail_cpu_ids)) { + snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev); + priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( + &pdev->dev, "ext-amp-enable", 0, + codec_dev->fwnode, + GPIOD_OUT_LOW, "speaker-amp"); + if (IS_ERR(priv->ext_amp_gpio)) { + ret_val = PTR_ERR(priv->ext_amp_gpio); + switch (ret_val) { + case -ENOENT: + priv->ext_amp_gpio = NULL; + break; + default: + dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", + ret_val); + /* fall through */ + case -EPROBE_DEFER: + put_device(codec_dev); + return ret_val; + } + } + } + + put_device(codec_dev); log_quirks(&pdev->dev); @@ -876,10 +1052,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } + if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) + hp_swapped = "-hp-swapped"; + else + hp_swapped = ""; + snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-intmic-%s-hsmic", - intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], - hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-spk-%s-mic%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? + "mono" : "stereo", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 000000000000..c4b94e2617c5 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 94294c27d1db..38f6ab74709d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .trigger = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3a61252fe450..21a6490746a6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } /* * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 92f5fb2ae0a3..a892b37eab7c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) @@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ff6646cfa21..d31482b8c9bb 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index b0610bba3cfa..e877bb60beb1 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 38a1495c29cf..0e1818dd4cc6 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 7379d8830c39..915a34cdc8ac 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o -snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o +snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ + soc-acpi-intel-hsw-bdw-match.o \ + soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ + soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ + soc-acpi-intel-cnl-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c new file mode 100644 index 000000000000..f39386e540d3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -0,0 +1,59 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static struct snd_soc_acpi_codecs bxt_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { + { + .id = "INT343A", + .drv_name = "bxt_alc298s_i2s", + .fw_filename = "intel/dsp_fw_bxtn.bin", + }, + { + .id = "DLGS7219", + .drv_name = "bxt_da7219_max98357a", + .fw_filename = "intel/dsp_fw_bxtn.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &bxt_codecs, + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "104C5122", + .drv_name = "bxt-pcm512x", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-pcm512x.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "1AEC8804", + .drv_name = "bxt-wm8804", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-wm8804.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "INT34C3", + .drv_name = "bxt_tdf8532", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index bfe1ca68a542..4daa8a4f0c0c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5670.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -98,8 +98,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", .machine_quirk = byt_quirk, - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -107,8 +107,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -116,8 +116,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -125,8 +125,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5651.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -134,8 +134,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -143,8 +143,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ @@ -153,8 +153,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -162,8 +162,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* use CHT driver to Baytrail Chromebooks */ @@ -172,8 +172,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-max98090.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index ad1eb2d644be..91bb99b69601 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -68,8 +68,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -77,8 +77,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -86,8 +86,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -95,8 +95,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -104,8 +104,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -113,8 +113,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-max98090.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -122,8 +122,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-nau8824", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-nau8824.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -131,8 +131,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -140,8 +140,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -149,8 +149,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-es8316.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-es8316.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ @@ -160,8 +160,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", .machine_quirk = cht_quirk, - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -169,8 +169,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5651, use Baytrail machine driver */ @@ -179,8 +179,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5651.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c new file mode 100644 index 000000000000..ec8e28e7b937 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -0,0 +1,32 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata cnl_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { + { + .id = "INT34C2", + .drv_name = "cnl_rt274", + .fw_filename = "intel/dsp_fw_cnl.bin", + .pdata = &cnl_pdata, + .sof_fw_filename = "intel/sof-cnl.ri", + .sof_tplg_filename = "intel/sof-cnl-rt274.tplg", + .asoc_plat_name = "0000:00:1f.3", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c new file mode 100644 index 000000000000..305875af71ca --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static struct snd_soc_acpi_codecs glk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { + { + .id = "INT343A", + .drv_name = "glk_alc298s_i2s", + .fw_filename = "intel/dsp_fw_glk.bin", + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-alc298.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "DLGS7219", + .drv_name = "glk_da7219_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index e0e8c8c27528..494a0ea9b029 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST1.bin", - .sof_fw_filename = "intel/reef-hsw.ri", - .sof_tplg_filename = "intel/reef-hsw.tplg", + .sof_fw_filename = "intel/sof-hsw.ri", + .sof_tplg_filename = "intel/sof-hsw.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} @@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .id = "INT343A", .drv_name = "broadwell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt286.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5640.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5640.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c new file mode 100644 index 000000000000..0ee173ca437d --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -0,0 +1,91 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs kbl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { + .num_codecs = 1, + .codecs = {"10EC5663"} +}; + +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { + .num_codecs = 2, + .codecs = {"10EC5663", "10EC5514"} +}; + +static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { + { + .id = "INT343A", + .drv_name = "kbl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "INT343B", + .drv_name = "kbl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "kbl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_r5514_5663_max", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_5663_5514_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_rt5663_m98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_poppy_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "10EC5663", + .drv_name = "kbl_rt5663", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98357_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c new file mode 100644 index 000000000000..0c9c0edd35b3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs skl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { + { + .id = "INT343A", + .drv_name = "skl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_release.bin", + }, + { + .id = "INT343B", + .drv_name = "skl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "skl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 657afc02f1c4..11041aedea31 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp) } EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel); -int sst_dma_new(struct sst_dsp *sst) +static int sst_dma_new(struct sst_dsp *sst) { struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; @@ -320,9 +320,8 @@ err_dma_dev: devm_kfree(sst->dev, dma); return ret; } -EXPORT_SYMBOL(sst_dma_new); -void sst_dma_free(struct sst_dma *dma) +static void sst_dma_free(struct sst_dma *dma) { if (dma == NULL) @@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma) dw_remove(dma->chip); } -EXPORT_SYMBOL(sst_dma_free); /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 62f3a8e0ec87..dcff13802c00 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -121,8 +121,8 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc) { int i; - ipc->msg = kzalloc(sizeof(struct ipc_message) * - IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + ipc->msg = kcalloc(IPC_EMPTY_LIST_SIZE, sizeof(struct ipc_message), + GFP_KERNEL); if (ipc->msg == NULL) return -ENOMEM; diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index b2bec36d074c..a28220e67cdf 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module_template template; int count, ret; void __iomem *ram; + int type = le16_to_cpu(module->type); + int entry_point = le32_to_cpu(module->entry_point); /* TODO: allowed module types need to be configurable */ - if (module->type != SST_HSW_MODULE_BASE_FW - && module->type != SST_HSW_MODULE_PCM_SYSTEM - && module->type != SST_HSW_MODULE_PCM - && module->type != SST_HSW_MODULE_PCM_REFERENCE - && module->type != SST_HSW_MODULE_PCM_CAPTURE - && module->type != SST_HSW_MODULE_WAVES - && module->type != SST_HSW_MODULE_LPAL) + if (type != SST_HSW_MODULE_BASE_FW && + type != SST_HSW_MODULE_PCM_SYSTEM && + type != SST_HSW_MODULE_PCM && + type != SST_HSW_MODULE_PCM_REFERENCE && + type != SST_HSW_MODULE_PCM_CAPTURE && + type != SST_HSW_MODULE_WAVES && + type != SST_HSW_MODULE_LPAL) return 0; dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", module->signature, module->mod_size, - module->blocks, module->type); - dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + module->blocks, type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point); dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", module->info.persistent_size, module->info.scratch_size); memset(&template, 0, sizeof(template)); - template.id = module->type; - template.entry = module->entry_point - 4; - template.persistent_size = module->info.persistent_size; - template.scratch_size = module->info.scratch_size; + template.id = type; + template.entry = entry_point - 4; + template.persistent_size = le32_to_cpu(module->info.persistent_size); + template.scratch_size = le32_to_cpu(module->info.scratch_size); mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)module + sizeof(*module); - for (count = 0; count < module->blocks; count++) { + for (count = 0; count < le32_to_cpu(module->blocks); count++) { - if (block->size <= 0) { + if (le32_to_cpu(block->size) <= 0) { dev_err(dsp->dev, "error: block %d size invalid\n", count); sst_module_free(mod); return -EINVAL; } - switch (block->type) { + switch (le32_to_cpu(block->type)) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - mod->offset = - block->ram_offset + dsp->addr.iram_offset; + mod->offset = le32_to_cpu(block->ram_offset) + + dsp->addr.iram_offset; mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: case SST_HSW_REGS: ram = dsp->addr.lpe; - mod->offset = block->ram_offset; + mod->offset = le32_to_cpu(block->ram_offset); mod->type = SST_MEM_DRAM; break; default: @@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - mod->size = block->size; + mod->size = le32_to_cpu(block->size); mod->data = (void *)block + sizeof(*block); mod->data_offset = mod->data - fw->dma_buf; @@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return ret; } - block = (void *)block + sizeof(*block) + block->size; + block = (void *)block + sizeof(*block) + + le32_to_cpu(block->size); } mod->state = SST_MODULE_STATE_LOADED; @@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* verify FW */ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || - (sst_fw->size != header->file_size + sizeof(*header))) { + (sst_fw->size != + le32_to_cpu(header->file_size) + sizeof(*header))) { dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); return -EINVAL; } @@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* parse each module */ module = (void *)sst_fw->dma_buf + sizeof(*header); - for (count = 0; count < header->modules; count++) { + for (count = 0; count < le32_to_cpu(header->modules); count++) { /* module */ ret = hsw_parse_module(dsp, sst_fw, module); @@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) dev_err(dsp->dev, "error: invalid module %d\n", count); return ret; } - module = (void *)module + sizeof(*module) + module->mod_size; + module = (void *)module + sizeof(*module) + + le32_to_cpu(module->mod_size); } return 0; diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index d5f9c30eba32..8bfb8b0fa3d5 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -33,8 +33,7 @@ static int skl_alloc_dma_buf(struct device *dev, struct snd_dma_buffer *dmab, size_t size) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -44,8 +43,7 @@ static int skl_alloc_dma_buf(struct device *dev, static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -89,8 +87,7 @@ void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) static int skl_dsp_setup_spib(struct device *dev, unsigned int size, int stream_tag, int enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream = snd_hdac_get_stream(bus, SNDRV_PCM_STREAM_PLAYBACK, stream_tag); struct hdac_ext_stream *estream; @@ -100,10 +97,10 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, estream = stream_to_hdac_ext_stream(stream); /* enable/disable SPIB for this hdac stream */ - snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + snd_hdac_ext_stream_spbcap_enable(bus, enable, stream->index); /* set the spib value */ - snd_hdac_ext_stream_set_spib(ebus, estream, size); + snd_hdac_ext_stream_set_spib(bus, estream, size); return 0; } @@ -111,8 +108,7 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, static int skl_dsp_prepare(struct device *dev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_ext_stream *estream; struct hdac_stream *stream; struct snd_pcm_substream substream; @@ -124,7 +120,7 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, memset(&substream, 0, sizeof(substream)); substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - estream = snd_hdac_ext_stream_assign(ebus, &substream, + estream = snd_hdac_ext_stream_assign(bus, &substream, HDAC_EXT_STREAM_TYPE_HOST); if (!estream) return -ENODEV; @@ -143,9 +139,8 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -163,10 +158,9 @@ static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) static int skl_dsp_cleanup(struct device *dev, struct snd_dma_buffer *dmab, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; struct hdac_ext_stream *estream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -270,8 +264,7 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) int skl_init_dsp(struct skl *skl) { void __iomem *mmio_base; - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_dsp_loader_ops loader_ops; int irq = bus->irq; const struct skl_dsp_ops *ops; @@ -279,8 +272,8 @@ int skl_init_dsp(struct skl *skl) int ret; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* read the BAR of the ADSP MMIO */ mmio_base = pci_ioremap_bar(skl->pci, 4); @@ -335,12 +328,11 @@ unmap_mmio: int skl_free_dsp(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_sst *ctx = skl->skl_sst; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); ctx->dsp_ops->cleanup(bus->dev, ctx); @@ -383,10 +375,11 @@ int skl_suspend_late_dsp(struct skl *skl) int skl_suspend_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; ret = skl_dsp_sleep(ctx->dsp); @@ -394,8 +387,8 @@ int skl_suspend_dsp(struct skl *skl) return ret; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); + snd_hdac_ext_bus_ppcap_enable(bus, false); return 0; } @@ -403,15 +396,16 @@ int skl_suspend_dsp(struct skl *skl) int skl_resume_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* check if DSP 1st boot is done */ if (skl->skl_sst->is_first_boot == true) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index b9b140275be0..01a050cf8775 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -141,7 +141,7 @@ struct nhlt_specific_cfg { struct nhlt_fmt *fmt; struct nhlt_endpoint *epnt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; struct nhlt_acpi_table *nhlt = skl->nhlt; @@ -228,7 +228,7 @@ static void skl_nhlt_trim_space(char *trim) int skl_nhlt_update_topology_bin(struct skl *skl) { struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", @@ -248,8 +248,8 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; char platform_id[32]; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..823e39103edd 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -67,16 +67,15 @@ struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream) return substream->runtime->private_data; } -static struct hdac_ext_bus *get_bus_ctx(struct snd_pcm_substream *substream) +static struct hdac_bus *get_bus_ctx(struct snd_pcm_substream *substream) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct hdac_stream *hstream = hdac_stream(stream); struct hdac_bus *bus = hstream->bus; - - return hbus_to_ebus(bus); + return bus; } -static int skl_substream_alloc_pages(struct hdac_ext_bus *ebus, +static int skl_substream_alloc_pages(struct hdac_bus *bus, struct snd_pcm_substream *substream, size_t size) { @@ -95,7 +94,7 @@ static int skl_substream_free_pages(struct hdac_bus *bus, return snd_pcm_lib_free_pages(substream); } -static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, +static void skl_set_pcm_constrains(struct hdac_bus *bus, struct snd_pcm_runtime *runtime) { snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -105,9 +104,9 @@ static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, 20, 178000000); } -static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *ebus) +static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_bus *bus) { - if ((ebus_to_hbus(ebus))->ppcap) + if (bus->ppcap) return HDAC_EXT_STREAM_TYPE_HOST; else return HDAC_EXT_STREAM_TYPE_COUPLED; @@ -123,9 +122,9 @@ static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *e static void skl_set_suspend_active(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, bool enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_dapm_widget *w; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -140,8 +139,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream, int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -153,7 +151,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->host_bps, 0); @@ -177,8 +175,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -190,7 +187,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->link_bps, 0); @@ -201,7 +198,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) snd_hdac_ext_link_stream_setup(stream, format_val); - list_for_each_entry(link, &ebus->hlink_list, list) { + list_for_each_entry(link, &bus->hlink_list, list) { if (link->index == params->link_index) snd_hdac_ext_link_set_stream_id(link, hstream->stream_tag); @@ -215,7 +212,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) static int skl_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream; struct snd_pcm_runtime *runtime = substream->runtime; struct skl_dma_params *dma_params; @@ -224,12 +221,12 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - stream = snd_hdac_ext_stream_assign(ebus, substream, - skl_get_host_stream_type(ebus)); + stream = snd_hdac_ext_stream_assign(bus, substream, + skl_get_host_stream_type(bus)); if (stream == NULL) return -EBUSY; - skl_set_pcm_constrains(ebus, runtime); + skl_set_pcm_constrains(bus, runtime); /* * disable WALLCLOCK timestamps for capture streams @@ -301,7 +298,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct skl_pipe_params p_params = {0}; @@ -309,7 +306,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, int ret, dma_id; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - ret = skl_substream_alloc_pages(ebus, substream, + ret = skl_substream_alloc_pages(bus, substream, params_buffer_bytes(params)); if (ret < 0) return ret; @@ -343,14 +340,14 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct skl_dma_params *dma_params = NULL; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(ebus)); + snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(bus)); dma_params = snd_soc_dai_get_dma_data(dai, substream); /* @@ -380,7 +377,7 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, static int skl_pcm_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct skl *skl = get_skl_ctx(dai->dev); struct skl_module_cfg *mconfig; @@ -400,7 +397,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream, snd_hdac_stream_cleanup(hdac_stream(stream)); hdac_stream(stream)->prepared = 0; - return skl_substream_free_pages(ebus_to_hbus(ebus), substream); + return skl_substream_free_pages(bus, substream); } static int skl_be_hw_params(struct snd_pcm_substream *substream, @@ -420,8 +417,7 @@ static int skl_be_hw_params(struct snd_pcm_substream *substream, static int skl_decoupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; int start; unsigned long cookie; @@ -470,7 +466,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl *skl = get_skl_ctx(dai->dev); struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_soc_dapm_widget *w; int ret; @@ -492,9 +488,9 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, * dpib & lpib position to resume before starting the * DMA */ - snd_hdac_ext_stream_drsm_enable(ebus, true, + snd_hdac_ext_stream_drsm_enable(bus, true, hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, + snd_hdac_ext_stream_set_dpibr(bus, stream, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } @@ -528,14 +524,14 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, ret = skl_decoupled_trigger(substream, cmd); if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ - stream->dpib = readl(ebus->bus.remap_addr + + stream->dpib = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(stream)->index)); stream->lpib = snd_hdac_stream_get_pos_lpib( hdac_stream(stream)); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } break; @@ -546,11 +542,12 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } + static int skl_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_soc_dai *codec_dai = rtd->codec_dai; @@ -558,14 +555,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_link *link; int stream_tag; - link_dev = snd_hdac_ext_stream_assign(ebus, substream, + link_dev = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_LINK); if (!link_dev) return -EBUSY; snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); if (!link) return -EINVAL; @@ -610,7 +607,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); @@ -626,7 +623,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: snd_hdac_ext_link_stream_clear(link_dev); if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); break; default: @@ -638,7 +635,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, static int skl_link_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); @@ -648,7 +645,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); if (!link) return -EINVAL; @@ -1017,10 +1014,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } @@ -1041,8 +1039,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream) static int skl_coupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; struct snd_pcm_substream *s; bool start; @@ -1115,9 +1112,9 @@ static int skl_coupled_trigger(struct snd_pcm_substream *substream, static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); - if (!(ebus_to_hbus(ebus))->ppcap) + if (!bus->ppcap) return skl_coupled_trigger(substream, cmd); return 0; @@ -1127,7 +1124,7 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); unsigned int pos; /* @@ -1152,12 +1149,12 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + + pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); } else { udelay(20); - readl(ebus->bus.remap_addr + + readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); @@ -1242,11 +1239,11 @@ static void skl_pcm_free(struct snd_pcm *pcm) static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_pcm *pcm = rtd->pcm; unsigned int size; int retval = 0; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { @@ -1356,19 +1353,19 @@ static int skl_populate_modules(struct skl *skl) static int skl_platform_soc_probe(struct snd_soc_component *component) { - struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(component->dev); + struct skl *skl = bus_to_skl(bus); const struct skl_dsp_ops *ops; int ret; pm_runtime_get_sync(component->dev); - if ((ebus_to_hbus(ebus))->ppcap) { + if (bus->ppcap) { skl->component = component; /* init debugfs */ skl->debugfs = skl_debugfs_init(skl); - ret = skl_tplg_init(component, ebus); + ret = skl_tplg_init(component, bus); if (ret < 0) { dev_err(component->dev, "Failed to init topology!\n"); return ret; @@ -1425,10 +1422,10 @@ static const struct snd_soc_component_driver skl_component = { int skl_platform_register(struct device *dev) { int ret; - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); struct snd_soc_dai_driver *dais; int num_dais = ARRAY_SIZE(skl_platform_dai); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); INIT_LIST_HEAD(&skl->ppl_list); INIT_LIST_HEAD(&skl->bind_list); @@ -1464,8 +1461,8 @@ err: int skl_platform_unregister(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); struct skl_module_deferred_bind *modules, *tmp; if (!list_empty(&skl->bind_list)) { diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index d2b1d60fec02..5bc0d38da7e3 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp *ctx) /* Code loader helper APIs */ static void skl_cldma_setup_bdle(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc) + __le32 **bdlp, int size, int with_ioc) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; ctx->cl_dev.frags = 0; while (size > 0) { @@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx) int skl_cldma_prepare(struct sst_dsp *ctx) { int ret; - u32 *bdl; + __le32 *bdl; ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; @@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx) ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); return ret; } - bdl = (u32 *)ctx->cl_dev.dmab_bdl.area; + bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area; /* Allocate BDLs */ ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h index 5b730a1a0ae4..ec736921a083 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.h +++ b/sound/soc/intel/skylake/skl-sst-cldma.h @@ -203,7 +203,7 @@ struct sst_dsp; struct skl_cl_dev_ops { void (*cl_setup_bdle)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc); + __le32 **bdlp, int size, int with_ioc); void (*cl_setup_controller)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_bdl, unsigned int max_size, u32 page_count); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2c5129782959..2620d77729c5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; @@ -934,7 +937,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl, struct soc_bytes_ext *sb = (void *) k->private_value; struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private; struct skl_kpb_params *uuid_params, *params; - struct hdac_bus *bus = ebus_to_hbus(skl_to_ebus(skl)); + struct hdac_bus *bus = skl_to_bus(skl); int i, size, module_id; if (bc->set_params == SKL_PARAM_BIND && bc->max) { @@ -2428,8 +2431,10 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U8_DYN_IN_PIN: if (!mconfig->m_in_pin) - mconfig->m_in_pin = devm_kzalloc(dev, MAX_IN_QUEUE * - sizeof(*mconfig->m_in_pin), GFP_KERNEL); + mconfig->m_in_pin = + devm_kcalloc(dev, MAX_IN_QUEUE, + sizeof(*mconfig->m_in_pin), + GFP_KERNEL); if (!mconfig->m_in_pin) return -ENOMEM; @@ -2439,8 +2444,10 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U8_DYN_OUT_PIN: if (!mconfig->m_out_pin) - mconfig->m_out_pin = devm_kzalloc(dev, MAX_IN_QUEUE * - sizeof(*mconfig->m_in_pin), GFP_KERNEL); + mconfig->m_out_pin = + devm_kcalloc(dev, MAX_IN_QUEUE, + sizeof(*mconfig->m_in_pin), + GFP_KERNEL); if (!mconfig->m_out_pin) return -ENOMEM; @@ -2852,14 +2859,14 @@ static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w, mconfig->time_slot = dfw->time_slot; mconfig->formats_config.caps_size = dfw->caps.caps_size; - mconfig->m_in_pin = devm_kzalloc(dev, - MAX_IN_QUEUE * sizeof(*mconfig->m_in_pin), + mconfig->m_in_pin = devm_kcalloc(dev, + MAX_IN_QUEUE, sizeof(*mconfig->m_in_pin), GFP_KERNEL); if (!mconfig->m_in_pin) return -ENOMEM; - mconfig->m_out_pin = devm_kzalloc(dev, - MAX_OUT_QUEUE * sizeof(*mconfig->m_out_pin), + mconfig->m_out_pin = devm_kcalloc(dev, + MAX_OUT_QUEUE, sizeof(*mconfig->m_out_pin), GFP_KERNEL); if (!mconfig->m_out_pin) return -ENOMEM; @@ -3020,14 +3027,13 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { int ret; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; if (!tplg_w->priv.size) @@ -3127,14 +3133,14 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { struct soc_bytes_ext *sb; struct snd_soc_tplg_bytes_control *tplg_bc; struct snd_soc_tplg_enum_control *tplg_ec; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); struct soc_enum *se; switch (hdr->ops.info) { @@ -3615,12 +3621,11 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); /* proceed only if we have private data defined */ if (manifest->priv.size == 0) @@ -3709,12 +3714,11 @@ static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe) /* * SKL topology init routine */ -int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus) +int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) { int ret; const struct firmware *fw; - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_pipeline *ppl; ret = request_firmware(&fw, skl->tplg_name, bus->dev); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..82282cac9751 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -458,9 +458,9 @@ enum skl_channel { static inline struct skl *get_skl_ctx(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); - return ebus_to_skl(ebus); + return bus_to_skl(bus); } int skl_tplg_be_update_params(struct snd_soc_dai *dai, @@ -470,7 +470,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps, void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, struct skl_pipe_params *params, int stream); int skl_tplg_init(struct snd_soc_component *component, - struct hdac_ext_bus *ebus); + struct hdac_bus *ebus); struct skl_module_cfg *skl_tplg_fe_get_cpr_module( struct snd_soc_dai *dai, int stream); int skl_tplg_update_pipe_params(struct device *dev, @@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f0d9793f872a..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -29,6 +29,7 @@ #include <linux/delay.h> #include <sound/pcm.h> #include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> #include <sound/hda_register.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> @@ -36,8 +37,6 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" -static struct skl_machine_pdata skl_dmic_data; - /* * initialize the PCI registers */ @@ -54,7 +53,7 @@ static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg, static void skl_init_pci(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = skl_to_bus(skl); /* * Clear bits 0-2 of PCI register TCSEL (at offset 0x44) @@ -63,7 +62,7 @@ static void skl_init_pci(struct skl *skl) * codecs. * The PCI register TCSEL is defined in the Intel manuals. */ - dev_dbg(ebus_to_hbus(ebus)->dev, "Clearing TCSEL\n"); + dev_dbg(bus->dev, "Clearing TCSEL\n"); skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); } @@ -103,8 +102,7 @@ static void skl_enable_miscbdcge(struct device *dev, bool enable) static void skl_clock_power_gating(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u32 val; /* Update PDCGE bit of CGCTL register */ @@ -127,7 +125,6 @@ static void skl_clock_power_gating(struct device *dev, bool enable) */ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink; int ret; @@ -135,7 +132,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) ret = snd_hdac_bus_init_chip(bus, full_reset); /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &ebus->hlink_list, list) + list_for_each_entry(hlink, &bus->hlink_list, list) bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); @@ -146,8 +143,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) void skl_update_d0i3c(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u8 reg; int timeout = 50; @@ -197,8 +193,7 @@ static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr) static irqreturn_t skl_interrupt(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; if (!pm_runtime_active(bus->dev)) @@ -227,8 +222,7 @@ static irqreturn_t skl_interrupt(int irq, void *dev_id) static irqreturn_t skl_threaded_handler(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; status = snd_hdac_chip_readl(bus, INTSTS); @@ -238,16 +232,15 @@ static irqreturn_t skl_threaded_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) +static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); int ret; ret = request_threaded_irq(skl->pci->irq, skl_interrupt, skl_threaded_handler, IRQF_SHARED, - KBUILD_MODNAME, ebus); + KBUILD_MODNAME, bus); if (ret) { dev_err(bus->dev, "unable to grab IRQ %d, disabling device\n", @@ -264,21 +257,20 @@ static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) static int skl_suspend_late(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); return skl_suspend_late_dsp(skl); } #ifdef CONFIG_PM -static int _skl_suspend(struct hdac_ext_bus *ebus) +static int _skl_suspend(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = to_pci_dev(bus->dev); int ret; - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); ret = skl_suspend_dsp(skl); if (ret < 0) @@ -295,10 +287,9 @@ static int _skl_suspend(struct hdac_ext_bus *ebus) return 0; } -static int _skl_resume(struct hdac_ext_bus *ebus) +static int _skl_resume(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl_init_pci(skl); skl_init_chip(bus, true); @@ -314,9 +305,8 @@ static int _skl_resume(struct hdac_ext_bus *ebus) static int skl_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); int ret = 0; /* @@ -325,15 +315,15 @@ static int skl_suspend(struct device *dev) */ if (skl->supend_active) { /* turn off the links and stop the CORB/RIRB DMA if it is On */ - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); - if (ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); enable_irq_wake(bus->irq); pci_save_state(pci); } else { - ret = _skl_suspend(ebus); + ret = _skl_suspend(bus); if (ret < 0) return ret; skl->skl_sst->fw_loaded = false; @@ -352,9 +342,8 @@ static int skl_suspend(struct device *dev) static int skl_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct hdac_ext_link *hlink = NULL; int ret; @@ -374,32 +363,32 @@ static int skl_resume(struct device *dev) */ if (skl->supend_active) { pci_restore_state(pci); - snd_hdac_ext_bus_link_power_up_all(ebus); + snd_hdac_ext_bus_link_power_up_all(bus); disable_irq_wake(bus->irq); /* * turn On the links which are On before active suspend * and start the CORB/RIRB DMA if On before * active suspend. */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) snd_hdac_ext_bus_link_power_up(hlink); } - if (ebus->cmd_dma_state) - snd_hdac_bus_init_cmd_io(&ebus->bus); ret = 0; + if (bus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(bus); } else { - ret = _skl_resume(ebus); + ret = _skl_resume(bus); /* turn off the links which are off before suspend */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (!hlink->ref_count) snd_hdac_ext_bus_link_power_down(hlink); } - if (!ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (!bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); } return ret; @@ -410,23 +399,21 @@ static int skl_resume(struct device *dev) static int skl_runtime_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_suspend(ebus); + return _skl_suspend(bus); } static int skl_runtime_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_resume(ebus); + return _skl_resume(bus); } #endif /* CONFIG_PM */ @@ -439,20 +426,19 @@ static const struct dev_pm_ops skl_pm = { /* * destructor */ -static int skl_free(struct hdac_ext_bus *ebus) +static int skl_free(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl->init_done = 0; /* to be sure */ - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); if (bus->irq >= 0) - free_irq(bus->irq, (void *)ebus); + free_irq(bus->irq, (void *)bus); snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(ebus); - snd_hdac_link_free_all(ebus); + snd_hdac_stream_free_all(bus); + snd_hdac_link_free_all(bus); if (bus->remap_addr) iounmap(bus->remap_addr); @@ -460,11 +446,11 @@ static int skl_free(struct hdac_ext_bus *ebus) pci_release_regions(skl->pci); pci_disable_device(skl->pci); - snd_hdac_ext_bus_exit(ebus); + snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_i915_exit(&ebus->bus); + snd_hdac_i915_exit(bus); return 0; } @@ -488,8 +474,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static int skl_find_machine(struct skl *skl, void *driver_data) { + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = driver_data; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); @@ -500,17 +486,19 @@ static int skl_find_machine(struct skl *skl, void *driver_data) skl->mach = mach; skl->fw_name = mach->fw_filename; - pdata = skl->mach->pdata; + pdata = mach->pdata; - if (mach->pdata) + if (pdata) { skl->use_tplg_pcm = pdata->use_tplg_pcm; + pdata->dmic_num = skl_get_dmic_geo(skl); + } return 0; } static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; struct platform_device *pdev; int ret; @@ -544,7 +532,7 @@ static void skl_machine_device_unregister(struct skl *skl) static int skl_dmic_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct platform_device *pdev; int ret; @@ -643,12 +631,13 @@ static void skl_clock_device_unregister(struct skl *skl) /* * Probe the given codec address */ -static int probe_codec(struct hdac_ext_bus *ebus, int addr) +static int probe_codec(struct hdac_bus *bus, int addr) { - struct hdac_bus *bus = ebus_to_hbus(ebus); unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; + struct skl *skl = bus_to_skl(bus); + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -658,13 +647,16 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK\n", addr); - return snd_hdac_ext_bus_device_init(ebus, addr); + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); } /* Codec initialization */ -static void skl_codec_create(struct hdac_ext_bus *ebus) +static void skl_codec_create(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); int c, max_slots; max_slots = HDA_MAX_CODECS; @@ -672,7 +664,7 @@ static void skl_codec_create(struct hdac_ext_bus *ebus) /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { if ((bus->codec_mask & (1 << c))) { - if (probe_codec(ebus, c) < 0) { + if (probe_codec(bus, c) < 0) { /* * Some BIOSen give you wrong codec addresses * that don't exist @@ -722,8 +714,7 @@ static int skl_i915_init(struct hdac_bus *bus) static void skl_probe_work(struct work_struct *work) { struct skl *skl = container_of(work, struct skl, probe_work); - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct hdac_ext_link *hlink = NULL; int err; @@ -744,7 +735,7 @@ static void skl_probe_work(struct work_struct *work) dev_info(bus->dev, "no hda codecs found!\n"); /* create codec instances */ - skl_codec_create(ebus); + skl_codec_create(bus); /* register platform dai and controls */ err = skl_platform_register(bus->dev); @@ -773,8 +764,8 @@ static void skl_probe_work(struct work_struct *work) /* * we are done probing so decrement link counts */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); @@ -796,7 +787,7 @@ static int skl_create(struct pci_dev *pci, struct skl **rskl) { struct skl *skl; - struct hdac_ext_bus *ebus; + struct hdac_bus *bus; int err; @@ -811,23 +802,22 @@ static int skl_create(struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - ebus = &skl->ebus; - snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); - ebus->bus.use_posbuf = 1; + + bus = skl_to_bus(skl); + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); + bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - - ebus->bus.bdl_pos_adj = 0; + bus->bdl_pos_adj = 0; *rskl = skl; return 0; } -static int skl_first_init(struct hdac_ext_bus *ebus) +static int skl_first_init(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = skl->pci; int err; unsigned short gcap; @@ -848,7 +838,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) snd_hdac_bus_parse_capabilities(bus); - if (skl_acquire_irq(ebus, 0) < 0) + if (skl_acquire_irq(bus, 0) < 0) return -EBUSY; pci_set_master(pci); @@ -872,14 +862,14 @@ static int skl_first_init(struct hdac_ext_bus *ebus) if (!pb_streams && !cp_streams) return -EIO; - ebus->num_streams = cp_streams + pb_streams; + bus->num_streams = cp_streams + pb_streams; /* initialize streams */ snd_hdac_ext_stream_init_all - (ebus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); + (bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); start_idx = cp_streams; snd_hdac_ext_stream_init_all - (ebus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); + (bus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); err = snd_hdac_bus_alloc_stream_pages(bus); if (err < 0) @@ -895,7 +885,6 @@ static int skl_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct skl *skl; - struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; int err; @@ -904,10 +893,9 @@ static int skl_probe(struct pci_dev *pci, if (err < 0) return err; - ebus = &skl->ebus; - bus = ebus_to_hbus(ebus); + bus = skl_to_bus(skl); - err = skl_first_init(ebus); + err = skl_first_init(bus); if (err < 0) goto out_free; @@ -928,9 +916,7 @@ static int skl_probe(struct pci_dev *pci, skl_nhlt_update_topology_bin(skl); - pci_set_drvdata(skl->pci, ebus); - - skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); + pci_set_drvdata(skl->pci, bus); /* check if dsp is there */ if (bus->ppcap) { @@ -952,7 +938,7 @@ static int skl_probe(struct pci_dev *pci, skl->skl_sst->clock_power_gating = skl_clock_power_gating; } if (bus->mlcap) - snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_ext_bus_get_ml_capabilities(bus); snd_hdac_bus_stop_chip(bus); @@ -972,31 +958,30 @@ out_clk_free: out_nhlt_free: skl_nhlt_free(skl->nhlt); out_free: - skl_free(ebus); + skl_free(bus); return err; } static void skl_shutdown(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); struct hdac_stream *s; struct hdac_ext_stream *stream; struct skl *skl; - if (ebus == NULL) + if (!bus) return; - skl = ebus_to_skl(ebus); + skl = bus_to_skl(bus); if (!skl->init_done) return; - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); list_for_each_entry(s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } snd_hdac_bus_stop_chip(bus); @@ -1004,15 +989,15 @@ static void skl_shutdown(struct pci_dev *pci) static void skl_remove(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); release_firmware(skl->tplg); pm_runtime_get_noresume(&pci->dev); /* codec removal, invoke bus_device_remove */ - snd_hdac_ext_bus_device_remove(ebus); + snd_hdac_ext_bus_device_remove(bus); skl->debugfs = NULL; skl_platform_unregister(&pci->dev); @@ -1022,176 +1007,27 @@ static void skl_remove(struct pci_dev *pci) skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); skl_nhlt_free(skl->nhlt); - skl_free(ebus); + skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } -static struct snd_soc_acpi_codecs skl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs kbl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs bxt_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct snd_soc_acpi_codecs kbl_poppy_codecs = { - .num_codecs = 1, - .codecs = {"10EC5663"} -}; - -static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { - .num_codecs = 2, - .codecs = {"10EC5663", "10EC5514"} -}; - -static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct skl_machine_pdata cnl_pdata = { - .use_tplg_pcm = true, -}; - -static struct snd_soc_acpi_mach sst_skl_devdata[] = { - { - .id = "INT343A", - .drv_name = "skl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_release.bin", - }, - { - .id = "INT343B", - .drv_name = "skl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "skl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - {} -}; - -static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { - { - .id = "INT343A", - .drv_name = "bxt_alc298s_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - }, - { - .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &bxt_codecs, - }, - {} -}; - -static struct snd_soc_acpi_mach sst_kbl_devdata[] = { - { - .id = "INT343A", - .drv_name = "kbl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "INT343B", - .drv_name = "kbl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "kbl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_r5514_5663_max", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_5663_5514_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_rt5663_m98927", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_poppy_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "10EC5663", - .drv_name = "kbl_rt5663", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_7219_98357_codecs, - .pdata = &skl_dmic_data - }, - - {} -}; - -static struct snd_soc_acpi_mach sst_glk_devdata[] = { - { - .id = "INT343A", - .drv_name = "glk_alc298s_i2s", - .fw_filename = "intel/dsp_fw_glk.bin", - }, - {} -}; - -static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { - { - .id = "INT34C2", - .drv_name = "cnl_rt274", - .fw_filename = "intel/dsp_fw_cnl.bin", - .pdata = &cnl_pdata, - }, - {} -}; - /* PCI IDs */ static const struct pci_device_id skl_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&sst_skl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = (unsigned long)&sst_bxtp_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), - .driver_data = (unsigned long)&sst_kbl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, /* GLK */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = (unsigned long)&sst_glk_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = (unsigned long)&sst_cnl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 0d5375cbcf6e..78aa8bdcb619 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -71,7 +71,7 @@ struct skl_fw_config { }; struct skl { - struct hdac_ext_bus ebus; + struct hdac_bus hbus; struct pci_dev *pci; unsigned int init_done:1; /* delayed init status */ @@ -105,9 +105,8 @@ struct skl { struct snd_soc_acpi_mach *mach; }; -#define skl_to_ebus(s) (&(s)->ebus) -#define ebus_to_skl(sbus) \ - container_of(sbus, struct skl, sbus) +#define skl_to_bus(s) (&(s)->hbus) +#define bus_to_skl(bus) container_of(bus, struct skl, hbus) /* to pass dai dma data */ struct skl_dma_params { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 828d11c30c6a..968fba4d7533 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1347,7 +1347,8 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) afe->dev = &pdev->dev; dev = afe->dev; - afe_priv->i2s_path = devm_kzalloc(dev, afe_priv->soc->i2s_num * + afe_priv->i2s_path = devm_kcalloc(dev, + afe_priv->soc->i2s_num, sizeof(struct mt2701_i2s_path), GFP_KERNEL); if (!afe_priv->i2s_path) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include <sound/soc.h> +#include <linux/list.h> #include <linux/regmap.h> #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; - - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig new file mode 100644 index 000000000000..8af8bc358a90 --- /dev/null +++ b/sound/soc/meson/Kconfig @@ -0,0 +1,65 @@ +menu "ASoC support for Amlogic platforms" + depends on ARCH_MESON || COMPILE_TEST + +config SND_MESON_AXG_FIFO + tristate + select REGMAP_MMIO + +config SND_MESON_AXG_FRDDR + tristate "Amlogic AXG Playback FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend playback interfaces + embedded in the Amlogic AXG SoC family + +config SND_MESON_AXG_TODDR + tristate "Amlogic AXG Capture FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend capture interfaces + embedded in the Amlogic AXG SoC family + +config SND_MESON_AXG_TDM_FORMATTER + tristate + select REGMAP_MMIO + +config SND_MESON_AXG_TDM_INTERFACE + tristate + select SND_MESON_AXG_TDM_FORMATTER + +config SND_MESON_AXG_TDMIN + tristate "Amlogic AXG TDM Input Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM input formatter embedded + in the Amlogic AXG SoC family + +config SND_MESON_AXG_TDMOUT + tristate "Amlogic AXG TDM Output Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM output formatter embedded + in the Amlogic AXG SoC family + +config SND_MESON_AXG_SOUND_CARD + tristate "Amlogic AXG Sound Card Support" + select SND_MESON_AXG_TDM_INTERFACE + imply SND_MESON_AXG_FRDDR + imply SND_MESON_AXG_TODDR + imply SND_MESON_AXG_TDMIN + imply SND_MESON_AXG_TDMOUT + imply SND_MESON_AXG_SPDIFOUT + help + Select Y or M to add support for the AXG SoC sound card + +config SND_MESON_AXG_SPDIFOUT + tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 + imply SND_SOC_SPDIF + help + Select Y or M to add support for SPDIF output serializer embedded + in the Amlogic AXG SoC family + +endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile new file mode 100644 index 000000000000..c5e003b093db --- /dev/null +++ b/sound/soc/meson/Makefile @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0 OR MIT) + +snd-soc-meson-axg-fifo-objs := axg-fifo.o +snd-soc-meson-axg-frddr-objs := axg-frddr.o +snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o +snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmin-objs := axg-tdmin.o +snd-soc-meson-axg-tdmout-objs := axg-tdmout.o +snd-soc-meson-axg-sound-card-objs := axg-card.o +snd-soc-meson-axg-spdifout-objs := axg-spdifout.o + +obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o +obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o +obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o +obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o +obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o +obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o +obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c new file mode 100644 index 000000000000..2914ba0d965b --- /dev/null +++ b/sound/soc/meson/axg-card.c @@ -0,0 +1,671 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm.h" + +struct axg_card { + struct snd_soc_card card; + void **link_data; +}; + +struct axg_dai_link_tdm_mask { + u32 tx; + u32 rx; +}; + +struct axg_dai_link_tdm_data { + unsigned int mclk_fs; + unsigned int slots; + unsigned int slot_width; + u32 *tx_mask; + u32 *rx_mask; + struct axg_dai_link_tdm_mask *codec_masks; +}; + +#define PREFIX "amlogic," + +static int axg_card_reallocate_links(struct axg_card *priv, + unsigned int num_links) +{ + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} + +static int axg_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} + +static int axg_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, link->cpu_of_node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +static void axg_card_clean_references(struct axg_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + int i, j; + + if (card->dai_link) { + for (i = 0; i < card->num_links; i++) { + link = &card->dai_link[i]; + of_node_put(link->cpu_of_node); + for (j = 0; j < link->num_codecs; j++) + of_node_put(link->codecs[j].of_node); + } + } + + if (card->aux_dev) { + for (i = 0; i < card->num_aux_devs; i++) + of_node_put(card->aux_dev[i].codec_of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +static int axg_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); + if (num == -ENOENT) { + /* + * It is ok to have no auxiliary devices but for this card it + * is a strange situtation. Let's warn the about it. + */ + dev_warn(card->dev, "card has no auxiliary devices\n"); + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for (i = 0; i < card->num_aux_devs; i++, aux++) { + aux->codec_of_node = + of_parse_phandle(node, "audio-aux-devs", i); + if (!aux->codec_of_node) + return -EINVAL; + } + + return 0; +} + +static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (be->mclk_fs) { + mclk = params_rate(params) * be->mclk_fs; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + } + + return 0; +} + +static const struct snd_soc_ops axg_card_tdm_be_ops = { + .hw_params = axg_card_tdm_be_hw_params, +}; + +static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + int ret, i; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_tdm_slot(codec_dai, + be->codec_masks[i].tx, + be->codec_masks[i].rx, + be->slots, be->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(codec_dai->dev, + "setting tdm link slots failed\n"); + return ret; + } + } + + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + int ret; + + /* The loopback rx_mask is the pad tx_mask */ + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_add_tdm_loopback(struct snd_soc_card *card, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *lb; + int ret; + + /* extend links */ + ret = axg_card_reallocate_links(priv, card->num_links + 1); + if (ret) + return ret; + + lb = &card->dai_link[*index + 1]; + + lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + if (!lb->name) + return -ENOMEM; + + lb->stream_name = lb->name; + lb->cpu_of_node = pad->cpu_of_node; + lb->cpu_dai_name = "TDM Loopback"; + lb->codec_name = "snd-soc-dummy"; + lb->codec_dai_name = "snd-soc-dummy-dai"; + lb->dpcm_capture = 1; + lb->no_pcm = 1; + lb->ops = &axg_card_tdm_be_ops; + lb->init = axg_card_tdm_dai_lb_init; + + /* Provide the same link data to the loopback */ + priv->link_data[*index + 1] = priv->link_data[*index]; + + /* + * axg_card_clean_references() will iterate over this link, + * make sure the node count is balanced + */ + of_node_get(lb->cpu_of_node); + + /* Let add_links continue where it should */ + *index += 1; + + return 0; +} + +static unsigned int axg_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} + +static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + char propname[32]; + u32 tx, rx; + int i; + + be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->tx_mask), GFP_KERNEL); + be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->rx_mask), GFP_KERNEL); + if (!be->tx_mask || !be->rx_mask) + return -ENOMEM; + + for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]); + tx = max(tx, be->tx_mask[i]); + } + + /* Disable playback is the interface has no tx slots */ + if (!tx) + link->dpcm_playback = 0; + + for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]); + rx = max(rx, be->rx_mask[i]); + } + + /* Disable capture is the interface has no rx slots */ + if (!rx) + link->dpcm_capture = 0; + + /* ... but the interface should at least have one of them */ + if (!tx && !rx) { + dev_err(card->dev, "tdm link has no cpu slots\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-num", &be->slots); + if (!be->slots) { + /* + * If the slot number is not provided, set it such as it + * accommodates the largest mask + */ + be->slots = fls(max(tx, rx)); + } else if (be->slots < fls(max(tx, rx)) || be->slots > 32) { + /* + * Error if the slots can't accommodate the largest mask or + * if it is just too big + */ + dev_err(card->dev, "bad slot number\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width); + + return 0; +} + +static int axg_card_parse_codecs_masks(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + struct axg_dai_link_tdm_mask *codec_mask; + struct device_node *np; + + codec_mask = devm_kcalloc(card->dev, link->num_codecs, + sizeof(*codec_mask), GFP_KERNEL); + if (!codec_mask) + return -ENOMEM; + + be->codec_masks = codec_mask; + + for_each_child_of_node(node, np) { + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", + &codec_mask->rx); + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", + &codec_mask->tx); + + codec_mask++; + } + + return 0; +} + +static int axg_card_parse_tdm(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct axg_dai_link_tdm_data *be; + int ret; + + /* Allocate tdm link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup tdm link */ + link->ops = &axg_card_tdm_be_ops; + link->init = axg_card_tdm_dai_init; + link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + ret = axg_card_parse_cpu_tdm_slots(card, link, node, be); + if (ret) { + dev_err(card->dev, "error parsing tdm link slots\n"); + return ret; + } + + ret = axg_card_parse_codecs_masks(card, link, node, be); + if (ret) + return ret; + + /* Add loopback if the pad dai has playback */ + if (link->dpcm_playback) { + ret = axg_card_add_tdm_loopback(card, index); + if (ret) + return ret; + } + + return 0; +} + +static int axg_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = axg_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = axg_card_set_link_name(card, link, "be"); + if (ret) + dev_err(card->dev, "error setting %s link name\n", np->name); + + return ret; +} + +static int axg_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + bool is_playback) +{ + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return axg_card_set_link_name(card, link, "fe"); +} + +static int axg_card_cpu_is_capture_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-toddr"); +} + +static int axg_card_cpu_is_playback_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-frddr"); +} + +static int axg_card_cpu_is_tdm_iface(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); +} + +static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + int ret; + + ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); + if (ret) + return ret; + + if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, true); + else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, false); + else + ret = axg_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node)) + ret = axg_card_parse_tdm(card, np, index); + + return ret; +} + +static int axg_card_add_links(struct snd_soc_card *card) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = axg_card_reallocate_links(priv, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = axg_card_add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int axg_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static const struct of_device_id axg_card_of_match[] = { + { .compatible = "amlogic,axg-sound-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_card_of_match); + +static int axg_card_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_card *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + + ret = snd_soc_of_parse_card_name(&priv->card, "model"); + if (ret < 0) + return ret; + + ret = axg_card_parse_of_optional(&priv->card, "audio-routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = axg_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = axg_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + axg_card_clean_references(priv); + return ret; +} + +static int axg_card_remove(struct platform_device *pdev) +{ + struct axg_card *priv = platform_get_drvdata(pdev); + + axg_card_clean_references(priv); + + return 0; +} + +static struct platform_driver axg_card_pdrv = { + .probe = axg_card_probe, + .remove = axg_card_remove, + .driver = { + .name = "axg-sound-card", + .of_match_table = axg_card_of_match, + }, +}; +module_platform_driver(axg_card_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c new file mode 100644 index 000000000000..30262550e37b --- /dev/null +++ b/sound/soc/meson/axg-fifo.c @@ -0,0 +1,341 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/reset.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +/* + * This file implements the platform operations common to the playback and + * capture frontend DAI. The logic behind this two types of fifo is very + * similar but some difference exist. + * These differences the respective DAI drivers + */ + +static struct snd_pcm_hardware axg_fifo_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = AXG_FIFO_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .period_bytes_min = AXG_FIFO_MIN_DEPTH, + .period_bytes_max = UINT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return rtd->cpu_dai; +} + +static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return snd_soc_dai_get_drvdata(dai); +} + +static struct device *axg_fifo_dev(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return dai->dev; +} + +static void __dma_enable(struct axg_fifo *fifo, bool enable) +{ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN, + enable ? CTRL0_DMA_EN : 0); +} + +static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + __dma_enable(fifo, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + __dma_enable(fifo, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct snd_pcm_runtime *runtime = ss->runtime; + unsigned int addr; + + regmap_read(fifo->map, FIFO_STATUS2, &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = ss->runtime; + struct axg_fifo *fifo = axg_fifo_data(ss); + dma_addr_t end_ptr; + unsigned int burst_num; + int ret; + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup dma memory pointers */ + end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr); + regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr); + + /* Setup interrupt periodicity */ + burst_num = params_period_bytes(params) / AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_INT_ADDR, burst_num); + + /* Enable block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + + return 0; +} + +static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + /* Disable the block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + + return snd_pcm_lib_free_pages(ss); +} + +static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) +{ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + CTRL1_INT_CLR(mask)); + + /* Clear must also be cleared */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + 0); +} + +static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) +{ + struct snd_pcm_substream *ss = dev_id; + struct axg_fifo *fifo = axg_fifo_data(ss); + unsigned int status; + + regmap_read(fifo->map, FIFO_STATUS1, &status); + + status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + if (status & FIFO_INT_COUNT_REPEAT) + snd_pcm_period_elapsed(ss); + else + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); + + /* Ack irqs */ + axg_fifo_ack_irq(fifo, status); + + return IRQ_RETVAL(status); +} + +static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct device *dev = axg_fifo_dev(ss); + int ret; + + snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw); + + /* + * Make sure the buffer and period size are multiple of the FIFO + * minimum depth size + */ + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, + dev_name(dev), ss); + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Setup status2 so it reports the memory pointer */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_STATUS2_SEL_MASK, + CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + + /* Make sure the dma is initially disabled */ + __dma_enable(fifo, false); + + /* Disable irqs until params are ready */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_MASK), 0); + + /* Clear any pending interrupt */ + axg_fifo_ack_irq(fifo, FIFO_INT_MASK); + + /* Take memory arbitror out of reset */ + ret = reset_control_deassert(fifo->arb); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +static int axg_fifo_pcm_close(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + int ret; + + /* Put the memory arbitror back in reset */ + ret = reset_control_assert(fifo->arb); + + /* Disable fifo ip and register access */ + clk_disable_unprepare(fifo->pclk); + + /* remove IRQ */ + free_irq(fifo->irq, ss); + + return ret; +} + +const struct snd_pcm_ops axg_fifo_pcm_ops = { + .open = axg_fifo_pcm_open, + .close = axg_fifo_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = axg_fifo_pcm_hw_params, + .hw_free = axg_fifo_pcm_hw_free, + .pointer = axg_fifo_pcm_pointer, + .trigger = axg_fifo_pcm_trigger, +}; +EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops); + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = axg_fifo_hw.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream, + SNDRV_DMA_TYPE_DEV, card->dev, + size, size); +} +EXPORT_SYMBOL_GPL(axg_fifo_pcm_new); + +static const struct regmap_config axg_fifo_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = FIFO_STATUS2, +}; + +int axg_fifo_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_fifo_match_data *data; + struct axg_fifo *fifo; + struct resource *res; + void __iomem *regs; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + platform_set_drvdata(pdev, fifo); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg); + if (IS_ERR(fifo->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(fifo->map)); + return PTR_ERR(fifo->map); + } + + fifo->pclk = devm_clk_get(dev, NULL); + if (IS_ERR(fifo->pclk)) { + if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %ld\n", + PTR_ERR(fifo->pclk)); + return PTR_ERR(fifo->pclk); + } + + fifo->arb = devm_reset_control_get_exclusive(dev, NULL); + if (IS_ERR(fifo->arb)) { + if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) + dev_err(dev, "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); + return PTR_ERR(fifo->arb); + } + + fifo->irq = of_irq_get(dev->of_node, 0); + if (fifo->irq <= 0) { + dev_err(dev, "failed to get irq: %d\n", fifo->irq); + return fifo->irq; + } + + return devm_snd_soc_register_component(dev, data->component_drv, + data->dai_drv, 1); +} +EXPORT_SYMBOL_GPL(axg_fifo_probe); + +MODULE_DESCRIPTION("Amlogic AXG fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h new file mode 100644 index 000000000000..cb6c4013ca33 --- /dev/null +++ b/sound/soc/meson/axg-fifo.h @@ -0,0 +1,80 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_FIFO_H +#define _MESON_AXG_FIFO_H + +struct clk; +struct platform_device; +struct regmap; +struct reset_control; + +struct snd_soc_component_driver; +struct snd_soc_dai; +struct snd_soc_dai_driver; +struct snd_pcm_ops; +struct snd_soc_pcm_runtime; + +#define AXG_FIFO_CH_MAX 128 +#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AXG_FIFO_BURST 8 +#define AXG_FIFO_MIN_CNT 64 +#define AXG_FIFO_MIN_DEPTH (AXG_FIFO_BURST * AXG_FIFO_MIN_CNT) + +#define FIFO_INT_ADDR_FINISH BIT(0) +#define FIFO_INT_ADDR_INT BIT(1) +#define FIFO_INT_COUNT_REPEAT BIT(2) +#define FIFO_INT_COUNT_ONCE BIT(3) +#define FIFO_INT_FIFO_ZERO BIT(4) +#define FIFO_INT_FIFO_DEPTH BIT(5) +#define FIFO_INT_MASK GENMASK(7, 0) + +#define FIFO_CTRL0 0x00 +#define CTRL0_DMA_EN BIT(31) +#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_SEL_MASK GENMASK(2, 0) +#define CTRL0_SEL_SHIFT 0 +#define FIFO_CTRL1 0x04 +#define CTRL1_INT_CLR(x) ((x) << 0) +#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) +#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define STATUS2_SEL_DDR_READ 0 +#define CTRL1_THRESHOLD_MASK GENMASK(23, 16) +#define CTRL1_THRESHOLD(x) ((x) << 16) +#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) +#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define FIFO_START_ADDR 0x08 +#define FIFO_FINISH_ADDR 0x0c +#define FIFO_INT_ADDR 0x10 +#define FIFO_STATUS1 0x14 +#define STATUS1_INT_STS(x) ((x) << 0) +#define FIFO_STATUS2 0x18 + +struct axg_fifo { + struct regmap *map; + struct clk *pclk; + struct reset_control *arb; + int irq; +}; + +struct axg_fifo_match_data { + const struct snd_soc_component_driver *component_drv; + struct snd_soc_dai_driver *dai_drv; +}; + +extern const struct snd_pcm_ops axg_fifo_pcm_ops; + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type); +int axg_fifo_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_FIFO_H */ diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c new file mode 100644 index 000000000000..a6f6f6a2eca8 --- /dev/null +++ b/sound/soc/meson/axg-frddr.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +/* This driver implements the frontend playback DAI of AXG based SoCs */ + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +#define CTRL0_FRDDR_PP_MODE BIT(30) + +static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_depth, fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); + + /* + * TODO: We could adapt the fifo depth and the fifo threshold + * depending on the expected memory throughput and lantencies + * For now, we'll just use the same values as the vendor kernel + * Depth and threshold are zero based. + */ + fifo_depth = AXG_FIFO_MIN_CNT - 1; + fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK, + CTRL1_FRDDR_DEPTH(fifo_depth) | + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK); +} + +static const struct snd_soc_dai_ops axg_frddr_ops = { + .startup = axg_frddr_dai_startup, + .shutdown = axg_frddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_frddr_dai_drv = { + .name = "FRDDR", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_frddr_ops, + .pcm_new = axg_frddr_pcm_new, +}; + +static const char * const axg_frddr_sel_texts[] = { + "OUT 0", "OUT 1", "OUT 2", "OUT 3" +}; + +static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT, + axg_frddr_sel_texts); + +static const struct snd_kcontrol_new axg_frddr_out_demux = + SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = { + SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0, + &axg_frddr_out_demux), + SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = { + { "SINK SEL", NULL, "Playback" }, + { "OUT 0", "OUT 0", "SINK SEL" }, + { "OUT 1", "OUT 1", "SINK SEL" }, + { "OUT 2", "OUT 2", "SINK SEL" }, + { "OUT 3", "OUT 3", "SINK SEL" }, +}; + +static const struct snd_soc_component_driver axg_frddr_component_drv = { + .dapm_widgets = axg_frddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_frddr_dapm_widgets), + .dapm_routes = axg_frddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_frddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_frddr_match_data = { + .component_drv = &axg_frddr_component_drv, + .dai_drv = &axg_frddr_dai_drv +}; + +static const struct of_device_id axg_frddr_of_match[] = { + { + .compatible = "amlogic,axg-frddr", + .data = &axg_frddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_frddr_of_match); + +static struct platform_driver axg_frddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-frddr", + .of_match_table = axg_frddr_of_match, + }, +}; +module_platform_driver(axg_frddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG playback fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c new file mode 100644 index 000000000000..9dea528053ad --- /dev/null +++ b/sound/soc/meson/axg-spdifout.c @@ -0,0 +1,456 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm_params.h> +#include <sound/pcm_iec958.h> + +/* + * NOTE: + * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite + * of what the documentation says. Manual control on V, U and C bits is + * applied when the related sel bits are cleared + */ + +#define SPDIFOUT_STAT 0x00 +#define SPDIFOUT_GAIN0 0x04 +#define SPDIFOUT_GAIN1 0x08 +#define SPDIFOUT_CTRL0 0x0c +#define SPDIFOUT_CTRL0_EN BIT(31) +#define SPDIFOUT_CTRL0_RST_OUT BIT(29) +#define SPDIFOUT_CTRL0_RST_IN BIT(28) +#define SPDIFOUT_CTRL0_USEL BIT(26) +#define SPDIFOUT_CTRL0_USET BIT(25) +#define SPDIFOUT_CTRL0_CHSTS_SEL BIT(24) +#define SPDIFOUT_CTRL0_DATA_SEL BIT(20) +#define SPDIFOUT_CTRL0_MSB_FIRST BIT(19) +#define SPDIFOUT_CTRL0_VSEL BIT(18) +#define SPDIFOUT_CTRL0_VSET BIT(17) +#define SPDIFOUT_CTRL0_MASK_MASK GENMASK(11, 4) +#define SPDIFOUT_CTRL0_MASK(x) ((x) << 4) +#define SPDIFOUT_CTRL1 0x10 +#define SPDIFOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define SPDIFOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define SPDIFOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define SPDIFOUT_CTRL1_TYPE(x) ((x) << 4) +#define SPDIFOUT_PREAMB 0x14 +#define SPDIFOUT_SWAP 0x18 +#define SPDIFOUT_CHSTS0 0x1c +#define SPDIFOUT_CHSTS1 0x20 +#define SPDIFOUT_CHSTS2 0x24 +#define SPDIFOUT_CHSTS3 0x28 +#define SPDIFOUT_CHSTS4 0x2c +#define SPDIFOUT_CHSTS5 0x30 +#define SPDIFOUT_CHSTS6 0x34 +#define SPDIFOUT_CHSTS7 0x38 +#define SPDIFOUT_CHSTS8 0x3c +#define SPDIFOUT_CHSTS9 0x40 +#define SPDIFOUT_CHSTSA 0x44 +#define SPDIFOUT_CHSTSB 0x48 +#define SPDIFOUT_MUTE_VAL 0x4c + +struct axg_spdifout { + struct regmap *map; + struct clk *mclk; + struct clk *pclk; +}; + +static void axg_spdifout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN, + 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT); + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_IN, SPDIFOUT_CTRL0_RST_IN); + + /* Enable spdifout */ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, + SPDIFOUT_CTRL0_EN); +} + +static void axg_spdifout_disable(struct regmap *map) +{ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0); +} + +static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_spdifout_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_spdifout_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + /* Use spdif valid bit to perform digital mute */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET, + mute ? SPDIFOUT_CTRL0_VSET : 0); + + return 0; +} + +static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + /* Set the samples spdifout will pull from the FIFO */ + switch (params_channels(params)) { + case 1: + val = SPDIFOUT_CTRL0_MASK(0x1); + break; + case 2: + val = SPDIFOUT_CTRL0_MASK(0x3); + break; + default: + dev_err(dai->dev, "too many channels for spdif dai: %u\n", + params_channels(params)); + return -EINVAL; + } + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MASK_MASK, val); + + /* FIFO data are arranged in chunks of 64bits */ + switch (params_physical_width(params)) { + case 8: + /* 8 samples of 8 bits */ + val = SPDIFOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(4); + break; + default: + dev_err(dai->dev, "Unsupported physical width: %u\n", + params_physical_width(params)); + return -EINVAL; + } + + /* Position of the MSB in FIFO samples */ + val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL1, + SPDIFOUT_CTRL1_MSB_POS_MASK | + SPDIFOUT_CTRL1_TYPE_MASK, val); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + return 0; +} + +static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int offset; + int ret; + u8 cs[4]; + u32 val; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24; + + /* Setup channel status A bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS0, val); + + /* Clear channel status A bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + /* Setup channel status B bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS6, val); + + /* Clear channel status B bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + return 0; +} + +static int axg_spdifout_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + int ret; + + /* 2 * 32bits per subframe * 2 channels = 128 */ + ret = clk_set_rate(priv->mclk, rate * 128); + if (ret) { + dev_err(dai->dev, "failed to set spdif clock\n"); + return ret; + } + + ret = axg_spdifout_sample_fmt(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup sample format\n"); + return ret; + } + + ret = axg_spdifout_set_chsts(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup channel status words\n"); + return ret; + } + + return 0; +} + +static int axg_spdifout_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + /* Clock the spdif output block */ + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "failed to enable pclk\n"); + return ret; + } + + /* Make sure the block is initially stopped */ + axg_spdifout_disable(priv->map); + + /* Insert data from bit 27 lsb first */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + /* Manual control of V, C and U, U = 0 */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL | + SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET, + 0); + + /* Static SWAP configuration ATM */ + regmap_write(priv->map, SPDIFOUT_SWAP, 0x10); + + return 0; +} + +static void axg_spdifout_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->pclk); +} + +static const struct snd_soc_dai_ops axg_spdifout_ops = { + .trigger = axg_spdifout_trigger, + .digital_mute = axg_spdifout_digital_mute, + .hw_params = axg_spdifout_hw_params, + .startup = axg_spdifout_startup, + .shutdown = axg_spdifout_shutdown, +}; + +static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = { + { + .name = "SPDIF Output", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &axg_spdifout_ops, + }, +}; + +static const char * const spdifout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24, + spdifout_sel_texts); + +static const struct snd_kcontrol_new axg_spdifout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum); + +static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux), +}; + +static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "Playback", NULL, "SRC SEL" }, +}; + +static const struct snd_kcontrol_new axg_spdifout_controls[] = { + SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1), + SOC_SINGLE("Playback Gain Enable Switch", + SPDIFOUT_CTRL1, 26, 1, 0), + SOC_SINGLE("Playback Channels Mix Switch", + SPDIFOUT_CTRL0, 23, 1, 0), +}; + +static int axg_spdifout_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_spdifout *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(priv->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(priv->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_spdifout_component_drv = { + .controls = axg_spdifout_controls, + .num_controls = ARRAY_SIZE(axg_spdifout_controls), + .dapm_widgets = axg_spdifout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_spdifout_dapm_widgets), + .dapm_routes = axg_spdifout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes), + .set_bias_level = axg_spdifout_set_bias_level, +}; + +static const struct regmap_config axg_spdifout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = SPDIFOUT_MUTE_VAL, +}; + +static const struct of_device_id axg_spdifout_of_match[] = { + { .compatible = "amlogic,axg-spdifout", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_spdifout_of_match); + +static int axg_spdifout_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_spdifout *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv, + axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv)); +} + +static struct platform_driver axg_spdifout_pdrv = { + .probe = axg_spdifout_probe, + .driver = { + .name = "axg-spdifout", + .of_match_table = axg_spdifout_of_match, + }, +}; +module_platform_driver(axg_spdifout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c new file mode 100644 index 000000000000..43e390f9358a --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "axg-tdm-formatter.h" + +struct axg_tdm_formatter { + struct list_head list; + struct axg_tdm_stream *stream; + const struct axg_tdm_formatter_driver *drv; + struct clk *pclk; + struct clk *sclk; + struct clk *lrclk; + struct clk *sclk_sel; + struct clk *lrclk_sel; + bool enabled; + struct regmap *map; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset) +{ + unsigned int val, ch = ts->channels; + unsigned long mask; + int i, j; + + /* + * Distribute the channels of the stream over the available slots + * of each TDM lane + */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + val = 0; + mask = ts->mask[i]; + + for (j = find_first_bit(&mask, 32); + (j < 32) && ch; + j = find_next_bit(&mask, 32, j + 1)) { + val |= 1 << j; + ch -= 1; + } + + regmap_write(map, offset, val); + offset += regmap_get_reg_stride(map); + } + + /* + * If we still have channel left at the end of the process, it means + * the stream has more channels than we can accommodate and we should + * have caught this earlier. + */ + if (WARN_ON(ch != 0)) { + pr_err("channel mask error\n"); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); + +static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + bool invert = formatter->drv->invert_sclk; + int ret; + + /* Do nothing if the formatter is already enabled */ + if (formatter->enabled) + return 0; + + /* + * If sclk is inverted, invert it back and provide the inversion + * required by the formatter + */ + invert ^= axg_tdm_sclk_invert(ts->iface->fmt); + ret = clk_set_phase(formatter->sclk, invert ? 180 : 0); + if (ret) + return ret; + + /* Setup the stream parameter in the formatter */ + ret = formatter->drv->ops->prepare(formatter->map, formatter->stream); + if (ret) + return ret; + + /* Enable the signal clocks feeding the formatter */ + ret = clk_prepare_enable(formatter->sclk); + if (ret) + return ret; + + ret = clk_prepare_enable(formatter->lrclk); + if (ret) { + clk_disable_unprepare(formatter->sclk); + return ret; + } + + /* Finally, actually enable the formatter */ + formatter->drv->ops->enable(formatter->map); + formatter->enabled = true; + + return 0; +} + +static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter) +{ + /* Do nothing if the formatter is already disabled */ + if (!formatter->enabled) + return; + + formatter->drv->ops->disable(formatter->map); + clk_disable_unprepare(formatter->lrclk); + clk_disable_unprepare(formatter->sclk); + formatter->enabled = false; +} + +static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + int ret = 0; + + mutex_lock(&ts->lock); + + /* Catch up if the stream is already running when we attach */ + if (ts->ready) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to enable formatter\n"); + goto out; + } + } + + list_add_tail(&formatter->list, &ts->formatter_list); +out: + mutex_unlock(&ts->lock); + return ret; +} + +static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + + mutex_lock(&ts->lock); + list_del(&formatter->list); + mutex_unlock(&ts->lock); + + axg_tdm_formatter_disable(formatter); +} + +static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter, + struct snd_soc_dapm_widget *w) +{ + struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w); + int ret; + + /* + * If we don't get a stream at this stage, it would mean that the + * widget is powering up but is not attached to any backend DAI. + * It should not happen, ever ! + */ + if (WARN_ON(!ts)) + return -ENODEV; + + /* Clock our device */ + ret = clk_prepare_enable(formatter->pclk); + if (ret) + return ret; + + /* Reparent the bit clock to the TDM interface */ + ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk); + if (ret) + goto disable_pclk; + + /* Reparent the sample clock to the TDM interface */ + ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk); + if (ret) + goto disable_pclk; + + formatter->stream = ts; + ret = axg_tdm_formatter_attach(formatter); + if (ret) + goto disable_pclk; + + return 0; + +disable_pclk: + clk_disable_unprepare(formatter->pclk); + return ret; +} + +static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter) +{ + axg_tdm_formatter_dettach(formatter); + clk_disable_unprepare(formatter->pclk); + formatter->stream = NULL; +} + +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = axg_tdm_formatter_power_up(formatter, w); + break; + + case SND_SOC_DAPM_PRE_PMD: + axg_tdm_formatter_power_down(formatter); + break; + + default: + dev_err(c->dev, "Unexpected event %d\n", event); + return -EINVAL; + } + + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_event); + +int axg_tdm_formatter_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_tdm_formatter_driver *drv; + struct axg_tdm_formatter *formatter; + struct resource *res; + void __iomem *regs; + int ret; + + drv = of_device_get_match_data(dev); + if (!drv) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL); + if (!formatter) + return -ENOMEM; + platform_set_drvdata(pdev, formatter); + formatter->drv = drv; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg); + if (IS_ERR(formatter->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(formatter->map)); + return PTR_ERR(formatter->map); + } + + /* Peripharal clock */ + formatter->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(formatter->pclk)) { + ret = PTR_ERR(formatter->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock */ + formatter->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(formatter->sclk)) { + ret = PTR_ERR(formatter->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Formatter sample clock */ + formatter->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(formatter->lrclk)) { + ret = PTR_ERR(formatter->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock input multiplexer */ + formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); + if (IS_ERR(formatter->sclk_sel)) { + ret = PTR_ERR(formatter->sclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk_sel: %d\n", ret); + return ret; + } + + /* Formatter sample clock input multiplexer */ + formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); + if (IS_ERR(formatter->lrclk_sel)) { + ret = PTR_ERR(formatter->lrclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, drv->component_drv, + NULL, 0); +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe); + +int axg_tdm_stream_start(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + int ret = 0; + + mutex_lock(&ts->lock); + ts->ready = true; + + /* Start all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to start tdm stream\n"); + goto out; + } + } + +out: + mutex_unlock(&ts->lock); + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_start); + +void axg_tdm_stream_stop(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + + mutex_lock(&ts->lock); + ts->ready = false; + + /* Stop all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + axg_tdm_formatter_disable(formatter); + } + + mutex_unlock(&ts->lock); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_stop); + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface) +{ + struct axg_tdm_stream *ts; + + ts = kzalloc(sizeof(*ts), GFP_KERNEL); + if (ts) { + INIT_LIST_HEAD(&ts->formatter_list); + mutex_init(&ts->lock); + ts->iface = iface; + } + + return ts; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc); + +void axg_tdm_stream_free(struct axg_tdm_stream *ts) +{ + /* + * If the list is not empty, it would mean that one of the formatter + * widget is still powered and attached to the interface while we + * we are removing the TDM DAI. It should not be possible + */ + WARN_ON(!list_empty(&ts->formatter_list)); + mutex_destroy(&ts->lock); + kfree(ts); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_free); + +MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h new file mode 100644 index 000000000000..cf947caf3cb1 --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -0,0 +1,39 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_TDM_FORMATTER_H +#define _MESON_AXG_TDM_FORMATTER_H + +#include "axg-tdm.h" + +struct platform_device; +struct regmap; +struct snd_soc_dapm_widget; +struct snd_kcontrol; + +struct axg_tdm_formatter_ops { + struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w); + void (*enable)(struct regmap *map); + void (*disable)(struct regmap *map); + int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts); +}; + +struct axg_tdm_formatter_driver { + const struct snd_soc_component_driver *component_drv; + const struct regmap_config *regmap_cfg; + const struct axg_tdm_formatter_ops *ops; + bool invert_sclk; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset); +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event); +int axg_tdm_formatter_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_TDM_FORMATTER_H */ diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c new file mode 100644 index 000000000000..7b8baf46d968 --- /dev/null +++ b/sound/soc/meson/axg-tdm-interface.c @@ -0,0 +1,542 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm.h" + +enum { + TDM_IFACE_PAD, + TDM_IFACE_LOOPBACK, +}; + +static unsigned int axg_tdm_slots_total(u32 *mask) +{ + unsigned int slots = 0; + int i; + + if (!mask) + return 0; + + /* Count the total number of slots provided by all 4 lanes */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) + slots += hweight32(mask[i]); + + return slots; +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *tx = (struct axg_tdm_stream *) + dai->playback_dma_data; + struct axg_tdm_stream *rx = (struct axg_tdm_stream *) + dai->capture_dma_data; + unsigned int tx_slots, rx_slots; + + tx_slots = axg_tdm_slots_total(tx_mask); + rx_slots = axg_tdm_slots_total(rx_mask); + + /* We should at least have a slot for a valid interface */ + if (!tx_slots && !rx_slots) { + dev_err(dai->dev, "interface has no slot\n"); + return -EINVAL; + } + + /* + * Amend the dai driver channel number and let dpcm channel merge do + * its job + */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + } + + iface->slots = slots; + + switch (slot_width) { + case 0: + /* defaults width to 32 if not provided */ + iface->slot_width = 32; + break; + case 8: + case 16: + case 24: + case 32: + iface->slot_width = slot_width; + break; + default: + dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); + +static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret = -ENOTSUPP; + + if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) { + if (!iface->mclk) { + dev_warn(dai->dev, "master clock not provided\n"); + } else { + ret = clk_set_rate(iface->mclk, freq); + if (!ret) + iface->mclk_rate = freq; + } + } + + return ret; +} + +static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + /* These modes are not supported */ + if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) { + dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + return -EINVAL; + } + + /* If the TDM interface is the clock master, it requires mclk */ + if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(dai->dev, "cpu clock master: mclk missing\n"); + return -ENODEV; + } + + iface->fmt = fmt; + return 0; +} + +static int axg_tdm_iface_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = + snd_soc_dai_get_dma_data(dai, substream); + int ret; + + if (!axg_tdm_slots_total(ts->mask)) { + dev_err(dai->dev, "interface has not slots\n"); + return -EINVAL; + } + + /* Apply component wide rate symmetry */ + if (dai->component->active) { + ret = snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + iface->rate); + if (ret < 0) { + dev_err(dai->dev, + "can't set iface rate constraint\n"); + return ret; + } + } + + return 0; +} + +static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + unsigned int channels = params_channels(params); + unsigned int width = params_width(params); + + /* Save rate and sample_bits for component symmetry */ + iface->rate = params_rate(params); + + /* Make sure this interface can cope with the stream */ + if (axg_tdm_slots_total(ts->mask) < channels) { + dev_err(dai->dev, "not enough slots for channels\n"); + return -EINVAL; + } + + if (iface->slot_width < width) { + dev_err(dai->dev, "incompatible slots width for stream\n"); + return -EINVAL; + } + + /* Save the parameter for tdmout/tdmin widgets */ + ts->physical_width = params_physical_width(params); + ts->width = params_width(params); + ts->channels = params_channels(params); + + return 0; +} + +static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned int ratio_num; + int ret; + + ret = clk_set_rate(iface->lrclk, params_rate(params)); + if (ret) { + dev_err(dai->dev, "setting sample clock failed: %d\n", ret); + return ret; + } + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + /* 50% duty cycle ratio */ + ratio_num = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* + * A zero duty cycle ratio will result in setting the mininum + * ratio possible which, for this clock, is 1 cycle of the + * parent bclk clock high and the rest low, This is exactly + * what we want here. + */ + ratio_num = 0; + break; + + default: + return -EINVAL; + } + + ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); + if (ret) { + dev_err(dai->dev, + "setting sample clock duty cycle failed: %d\n", ret); + return ret; + } + + /* Set sample clock inversion */ + ret = clk_set_phase(iface->lrclk, + axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); + if (ret) { + dev_err(dai->dev, + "setting sample clock phase failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned long srate; + int ret; + + srate = iface->slots * iface->slot_width * params_rate(params); + + if (!iface->mclk_rate) { + /* If no specific mclk is requested, default to bit clock * 4 */ + clk_set_rate(iface->mclk, 4 * srate); + } else { + /* Check if we can actually get the bit clock from mclk */ + if (iface->mclk_rate % srate) { + dev_err(dai->dev, + "can't derive sclk %lu from mclk %lu\n", + srate, iface->mclk_rate); + return -EINVAL; + } + } + + ret = clk_set_rate(iface->sclk, srate); + if (ret) { + dev_err(dai->dev, "setting bit clock failed: %d\n", ret); + return ret; + } + + /* Set the bit clock inversion */ + ret = clk_set_phase(iface->sclk, + axg_tdm_sclk_invert(iface->fmt) ? 0 : 180); + if (ret) { + dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + if (iface->slots > 2) { + dev_err(dai->dev, "bad slot number for format: %d\n", + iface->slots); + return -EINVAL; + } + break; + + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + break; + + default: + dev_err(dai->dev, "unsupported dai format\n"); + return -EINVAL; + } + + ret = axg_tdm_iface_set_stream(substream, params, dai); + if (ret) + return ret; + + if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) { + ret = axg_tdm_iface_set_sclk(dai, params); + if (ret) + return ret; + + ret = axg_tdm_iface_set_lrclk(dai, params); + if (ret) + return ret; + } + + return 0; +} + +static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Stop all attached formatters */ + axg_tdm_stream_stop(ts); + + return 0; +} + +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); +} + +static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) +{ + if (dai->capture_dma_data) + axg_tdm_stream_free(dai->capture_dma_data); + + if (dai->playback_dma_data) + axg_tdm_stream_free(dai->playback_dma_data); + + return 0; +} + +static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_widget) { + dai->capture_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->capture_dma_data) + return -ENOMEM; + } + + if (dai->playback_widget) { + dai->playback_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->playback_dma_data) { + axg_tdm_iface_remove_dai(dai); + return -ENOMEM; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops axg_tdm_iface_ops = { + .set_sysclk = axg_tdm_iface_set_sysclk, + .set_fmt = axg_tdm_iface_set_fmt, + .startup = axg_tdm_iface_startup, + .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, + .hw_free = axg_tdm_iface_hw_free, +}; + +/* TDM Backend DAIs */ +static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { + [TDM_IFACE_PAD] = { + .name = "TDM Pad", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_PAD, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, + [TDM_IFACE_LOOPBACK] = { + .name = "TDM Loopback", + .capture = { + .stream_name = "Loopback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_LOOPBACK, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, +}; + +static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(iface->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(iface->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { + .set_bias_level = axg_tdm_iface_set_bias_level, +}; + +static const struct of_device_id axg_tdm_iface_of_match[] = { + { .compatible = "amlogic,axg-tdm-iface", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match); + +static int axg_tdm_iface_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_soc_dai_driver *dai_drv; + struct axg_tdm_iface *iface; + int ret, i; + + iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL); + if (!iface) + return -ENOMEM; + platform_set_drvdata(pdev, iface); + + /* + * Duplicate dai driver: depending on the slot masks configuration + * We'll change the number of channel provided by DAI stream, so dpcm + * channel merge can be done properly + */ + dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv), + sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++) + memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i], + sizeof(*dai_drv)); + + /* Bit clock provided on the pad */ + iface->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(iface->sclk)) { + ret = PTR_ERR(iface->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Sample clock provided on the pad */ + iface->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(iface->lrclk)) { + ret = PTR_ERR(iface->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* + * mclk maybe be missing when the cpu dai is in slave mode and + * the codec does not require it to provide a master clock. + * At this point, ignore the error if mclk is missing. We'll + * throw an error if the cpu dai is master and mclk is missing + */ + iface->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(iface->mclk)) { + ret = PTR_ERR(iface->mclk); + if (ret == -ENOENT) { + iface->mclk = NULL; + } else { + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + } + + return devm_snd_soc_register_component(dev, + &axg_tdm_iface_component_drv, dai_drv, + ARRAY_SIZE(axg_tdm_iface_dai_drv)); +} + +static struct platform_driver axg_tdm_iface_pdrv = { + .probe = axg_tdm_iface_probe, + .driver = { + .name = "axg-tdm-iface", + .of_match_table = axg_tdm_iface_of_match, + }, +}; +module_platform_driver(axg_tdm_iface_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM interface driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h new file mode 100644 index 000000000000..e578b6f40a07 --- /dev/null +++ b/sound/soc/meson/axg-tdm.h @@ -0,0 +1,78 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_TDM_H +#define _MESON_AXG_TDM_H + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#define AXG_TDM_NUM_LANES 4 +#define AXG_TDM_CHANNEL_MAX 128 +#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct axg_tdm_iface { + struct clk *sclk; + struct clk *lrclk; + struct clk *mclk; + unsigned long mclk_rate; + + /* format is common to all the DAIs of the iface */ + unsigned int fmt; + unsigned int slots; + unsigned int slot_width; + + /* For component wide symmetry */ + int rate; +}; + +static inline bool axg_tdm_lrclk_invert(unsigned int fmt) +{ + return (fmt & SND_SOC_DAIFMT_I2S) ^ + !!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF)); +} + +static inline bool axg_tdm_sclk_invert(unsigned int fmt) +{ + return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF); +} + +struct axg_tdm_stream { + struct axg_tdm_iface *iface; + struct list_head formatter_list; + struct mutex lock; + unsigned int channels; + unsigned int width; + unsigned int physical_width; + u32 *mask; + bool ready; +}; + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface); +void axg_tdm_stream_free(struct axg_tdm_stream *ts); +int axg_tdm_stream_start(struct axg_tdm_stream *ts); +void axg_tdm_stream_stop(struct axg_tdm_stream *ts); + +static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) +{ + axg_tdm_stream_stop(ts); + return axg_tdm_stream_start(ts); +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width); + +#endif /* _MESON_AXG_TDM_H */ diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c new file mode 100644 index 000000000000..bbac44c81688 --- /dev/null +++ b/sound/soc/meson/axg-tdmin.c @@ -0,0 +1,229 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm-formatter.h" + +#define TDMIN_CTRL 0x00 +#define TDMIN_CTRL_ENABLE BIT(31) +#define TDMIN_CTRL_I2S_MODE BIT(30) +#define TDMIN_CTRL_RST_OUT BIT(29) +#define TDMIN_CTRL_RST_IN BIT(28) +#define TDMIN_CTRL_WS_INV BIT(25) +#define TDMIN_CTRL_SEL_SHIFT 20 +#define TDMIN_CTRL_IN_BIT_SKEW_MASK GENMASK(18, 16) +#define TDMIN_CTRL_IN_BIT_SKEW(x) ((x) << 16) +#define TDMIN_CTRL_LSB_FIRST BIT(5) +#define TDMIN_CTRL_BITNUM_MASK GENMASK(4, 0) +#define TDMIN_CTRL_BITNUM(x) ((x) << 0) +#define TDMIN_SWAP 0x04 +#define TDMIN_MASK0 0x08 +#define TDMIN_MASK1 0x0c +#define TDMIN_MASK2 0x10 +#define TDMIN_MASK3 0x14 +#define TDMIN_STAT 0x18 +#define TDMIN_MUTE_VAL 0x1c +#define TDMIN_MUTE0 0x20 +#define TDMIN_MUTE1 0x24 +#define TDMIN_MUTE2 0x28 +#define TDMIN_MUTE3 0x2c + +static const struct regmap_config axg_tdmin_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMIN_MUTE3, +}; + +static const char * const axg_tdmin_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5", +}; + +/* Change to special mux control to reset dapm */ +static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL, + TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts); + +static const struct snd_kcontrol_new axg_tdmin_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum); + +static struct snd_soc_dai * +axg_tdmin_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + if (p->source->id == snd_soc_dapm_dai_out) + return (struct snd_soc_dai *)p->source->priv; + + be = axg_tdmin_get_be(p->source); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmin_get_be(w); + + if (!be) + return NULL; + + return be->capture_dma_data; +} + +static void axg_tdmin_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT); + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_IN, TDMIN_CTRL_RST_IN); + + /* Actually enable tdmin */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE); +} + +static void axg_tdmin_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0); +} + +static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMIN_CTRL_IN_BIT_SKEW(3); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val = TDMIN_CTRL_IN_BIT_SKEW(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set stream format mode */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + val |= TDMIN_CTRL_I2S_MODE; + break; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMIN_CTRL_WS_INV; + + /* Set the slot width */ + val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1); + + /* + * The following also reset LSB_FIRST which result in the formatter + * placing the first bit received at bit 31 + */ + regmap_update_bits(map, TDMIN_CTRL, + (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV | + TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST | + TDMIN_CTRL_BITNUM_MASK), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMIN_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux), + SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 5", "IN 5" }, + { "DEC", NULL, "SRC SEL" }, + { "OUT", NULL, "DEC" }, +}; + +static const struct snd_soc_component_driver axg_tdmin_component_drv = { + .dapm_widgets = axg_tdmin_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmin_dapm_widgets), + .dapm_routes = axg_tdmin_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmin_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmin_ops = { + .get_stream = axg_tdmin_get_tdm_stream, + .prepare = axg_tdmin_prepare, + .enable = axg_tdmin_enable, + .disable = axg_tdmin_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmin_drv = { + .component_drv = &axg_tdmin_component_drv, + .regmap_cfg = &axg_tdmin_regmap_cfg, + .ops = &axg_tdmin_ops, + .invert_sclk = false, +}; + +static const struct of_device_id axg_tdmin_of_match[] = { + { + .compatible = "amlogic,axg-tdmin", + .data = &axg_tdmin_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); + +static struct platform_driver axg_tdmin_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmin", + .of_match_table = axg_tdmin_of_match, + }, +}; +module_platform_driver(axg_tdmin_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c new file mode 100644 index 000000000000..f73368ee1088 --- /dev/null +++ b/sound/soc/meson/axg-tdmout.c @@ -0,0 +1,259 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm-formatter.h" + +#define TDMOUT_CTRL0 0x00 +#define TDMOUT_CTRL0_BITNUM_MASK GENMASK(4, 0) +#define TDMOUT_CTRL0_BITNUM(x) ((x) << 0) +#define TDMOUT_CTRL0_SLOTNUM_MASK GENMASK(9, 5) +#define TDMOUT_CTRL0_SLOTNUM(x) ((x) << 5) +#define TDMOUT_CTRL0_INIT_BITNUM_MASK GENMASK(19, 15) +#define TDMOUT_CTRL0_INIT_BITNUM(x) ((x) << 15) +#define TDMOUT_CTRL0_ENABLE BIT(31) +#define TDMOUT_CTRL0_RST_OUT BIT(29) +#define TDMOUT_CTRL0_RST_IN BIT(28) +#define TDMOUT_CTRL1 0x04 +#define TDMOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define TDMOUT_CTRL1_TYPE(x) ((x) << 4) +#define TDMOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define TDMOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define TDMOUT_CTRL1_SEL_SHIFT 24 +#define TDMOUT_CTRL1_GAIN_EN 26 +#define TDMOUT_CTRL1_WS_INV BIT(28) +#define TDMOUT_SWAP 0x08 +#define TDMOUT_MASK0 0x0c +#define TDMOUT_MASK1 0x10 +#define TDMOUT_MASK2 0x14 +#define TDMOUT_MASK3 0x18 +#define TDMOUT_STAT 0x1c +#define TDMOUT_GAIN0 0x20 +#define TDMOUT_GAIN1 0x24 +#define TDMOUT_MUTE_VAL 0x28 +#define TDMOUT_MUTE0 0x2c +#define TDMOUT_MUTE1 0x30 +#define TDMOUT_MUTE2 0x34 +#define TDMOUT_MUTE3 0x38 +#define TDMOUT_MASK_VAL 0x3c + +static const struct regmap_config axg_tdmout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMOUT_MASK_VAL, +}; + +static const struct snd_kcontrol_new axg_tdmout_controls[] = { + SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0), + SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1, 0, 8, 255, 0), + SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0), + SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1, + TDMOUT_CTRL1_GAIN_EN, 1, 0), +}; + +static const char * const tdmout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1, + TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts); + +static const struct snd_kcontrol_new axg_tdmout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum); + +static struct snd_soc_dai * +axg_tdmout_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (!p->connect) + continue; + + if (p->sink->id == snd_soc_dapm_dai_in) + return (struct snd_soc_dai *)p->sink->priv; + + be = axg_tdmout_get_be(p->sink); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmout_get_be(w); + + if (!be) + return NULL; + + return be->playback_dma_data; +} + +static void axg_tdmout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT); + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_IN, TDMOUT_CTRL0_RST_IN); + + /* Actually enable tdmout */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE); +} + +static void axg_tdmout_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0); +} + +static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set the stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMOUT_CTRL0_INIT_BITNUM(1); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val |= TDMOUT_CTRL0_INIT_BITNUM(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set the slot width */ + val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1); + + /* Set the slot number */ + val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1); + + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_INIT_BITNUM_MASK | + TDMOUT_CTRL0_BITNUM_MASK | + TDMOUT_CTRL0_SLOTNUM_MASK, val); + + /* Set the sample width */ + val = TDMOUT_CTRL1_MSB_POS(ts->width - 1); + + /* FIFO data are arranged in chunks of 64bits */ + switch (ts->physical_width) { + case 8: + /* 8 samples of 8 bits */ + val |= TDMOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(4); + break; + default: + pr_err("Unsupported physical width: %u\n", + ts->physical_width); + return -EINVAL; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMOUT_CTRL1_WS_INV; + + regmap_update_bits(map, TDMOUT_CTRL1, + (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK | + TDMOUT_CTRL1_WS_INV), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMOUT_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux), + SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "ENC", NULL, "SRC SEL" }, + { "OUT", NULL, "ENC" }, +}; + +static const struct snd_soc_component_driver axg_tdmout_component_drv = { + .controls = axg_tdmout_controls, + .num_controls = ARRAY_SIZE(axg_tdmout_controls), + .dapm_widgets = axg_tdmout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmout_dapm_widgets), + .dapm_routes = axg_tdmout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmout_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmout_ops = { + .get_stream = axg_tdmout_get_tdm_stream, + .prepare = axg_tdmout_prepare, + .enable = axg_tdmout_enable, + .disable = axg_tdmout_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmout_drv = { + .component_drv = &axg_tdmout_component_drv, + .regmap_cfg = &axg_tdmout_regmap_cfg, + .ops = &axg_tdmout_ops, + .invert_sclk = true, +}; + +static const struct of_device_id axg_tdmout_of_match[] = { + { + .compatible = "amlogic,axg-tdmout", + .data = &axg_tdmout_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmout_of_match); + +static struct platform_driver axg_tdmout_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmout", + .of_match_table = axg_tdmout_of_match, + }, +}; +module_platform_driver(axg_tdmout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c new file mode 100644 index 000000000000..c2c9bb312586 --- /dev/null +++ b/sound/soc/meson/axg-toddr.c @@ -0,0 +1,199 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +/* This driver implements the frontend capture DAI of AXG based SoCs */ + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +#define CTRL0_TODDR_SEL_RESAMPLE BIT(30) +#define CTRL0_TODDR_EXT_SIGNED BIT(29) +#define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) +#define CTRL0_TODDR_TYPE(x) ((x) << 13) +#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) +#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) +#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) +#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) + +static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE); +} + +static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int type, width, msb = 31; + + /* + * NOTE: + * Almost all backend will place the MSB at bit 31, except SPDIF Input + * which will put it at index 28. When adding support for the SPDIF + * Input, we'll need to find which type of backend we are connected to. + */ + + switch (params_physical_width(params)) { + case 8: + type = 0; /* 8 samples of 8 bits */ + break; + case 16: + type = 2; /* 4 samples of 16 bits - right justified */ + break; + case 32: + type = 4; /* 2 samples of 32 bits - right justified */ + break; + default: + return -EINVAL; + } + + width = params_width(params); + + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_TODDR_TYPE_MASK | + CTRL0_TODDR_MSB_POS_MASK | + CTRL0_TODDR_LSB_POS_MASK, + CTRL0_TODDR_TYPE(type) | + CTRL0_TODDR_MSB_POS(msb) | + CTRL0_TODDR_LSB_POS(msb - (width - 1))); + + return 0; +} + +static int axg_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Select orginal data - resampling not supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0); + + /* Only signed format are supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED, + CTRL0_TODDR_EXT_SIGNED); + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0); + + /* TODDR does not have a configurable fifo depth */ + fifo_threshold = AXG_FIFO_MIN_CNT - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK, + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static const struct snd_soc_dai_ops axg_toddr_ops = { + .hw_params = axg_toddr_dai_hw_params, + .startup = axg_toddr_dai_startup, + .shutdown = axg_toddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_toddr_dai_drv = { + .name = "TODDR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_toddr_ops, + .pcm_new = axg_toddr_pcm_new, +}; + +static const char * const axg_toddr_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6" +}; + +static const unsigned int axg_toddr_sel_values[] = { + 0, 1, 2, 3, 4, 6 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0, + CTRL0_SEL_SHIFT, CTRL0_SEL_MASK, + axg_toddr_sel_texts, axg_toddr_sel_values); + +static const struct snd_kcontrol_new axg_toddr_in_mux = + SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux), + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = { + { "Capture", NULL, "SRC SEL" }, + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 6", "IN 6" }, +}; + +static const struct snd_soc_component_driver axg_toddr_component_drv = { + .dapm_widgets = axg_toddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets), + .dapm_routes = axg_toddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_toddr_match_data = { + .component_drv = &axg_toddr_component_drv, + .dai_drv = &axg_toddr_dai_drv +}; + +static const struct of_device_id axg_toddr_of_match[] = { + { + .compatible = "amlogic,axg-toddr", + .data = &axg_toddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_toddr_of_match); + +static struct platform_driver axg_toddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-toddr", + .of_match_table = axg_toddr_of_match, + }, +}; +module_platform_driver(axg_toddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG capture fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 77a30f0f0c96..4dce494dfbd3 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -22,7 +22,7 @@ * */ -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/spinlock.h> #include <linux/tty.h> #include <linux/module.h> @@ -32,7 +32,6 @@ #include <asm/mach-types.h> -#include <mach/board-ams-delta.h> #include <linux/platform_data/asoc-ti-mcbsp.h> #include "omap-mcbsp.h" @@ -213,7 +212,6 @@ static const struct snd_kcontrol_new ams_delta_audio_controls[] = { static struct snd_soc_jack ams_delta_hook_switch; static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = { { - .gpio = 4, .name = "hook_switch", .report = SND_JACK_HEADSET, .invert = 1, @@ -259,6 +257,7 @@ static struct timer_list cx81801_timer; static bool cx81801_cmd_pending; static bool ams_delta_muted; static DEFINE_SPINLOCK(ams_delta_lock); +static struct gpio_desc *gpiod_modem_codec; static void cx81801_timeout(struct timer_list *unused) { @@ -272,7 +271,7 @@ static void cx81801_timeout(struct timer_list *unused) /* Reconnect the codec DAI back from the modem to the CPU DAI * only if digital mute still off */ if (!muted) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); + gpiod_set_value(gpiod_modem_codec, 0); } /* Line discipline .open() */ @@ -381,8 +380,7 @@ static void cx81801_receive(struct tty_struct *tty, /* Apply config pulse by connecting the codec to the modem * if not already done */ if (apply) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, - AMS_DELTA_LATCH2_MODEM_CODEC); + gpiod_set_value(gpiod_modem_codec, 1); break; } } @@ -432,8 +430,7 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) spin_unlock_bh(&ams_delta_lock); if (apply) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, - mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0); + gpiod_set_value(gpiod_modem_codec, !!mute); return 0; } @@ -469,14 +466,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Store a pointer to the codec structure for tty ldisc use */ cx20442_codec = rtd->codec_dai->component; - /* Set up digital mute if not provided by the codec */ - if (!codec_dai->driver->ops) { - codec_dai->driver->ops = &ams_delta_dai_ops; - } else { - ams_delta_ops.startup = ams_delta_startup; - ams_delta_ops.shutdown = ams_delta_shutdown; - } - /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, @@ -486,7 +475,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) "Failed to allocate resources for hook switch, " "will continue without one.\n"); else { - ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ret = snd_soc_jack_add_gpiods(card->dev, &ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); if (ret) @@ -495,6 +484,21 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) "will continue with hook switch inactive.\n"); } + gpiod_modem_codec = devm_gpiod_get(card->dev, "modem_codec", + GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_modem_codec)) { + dev_warn(card->dev, "Failed to obtain modem_codec GPIO\n"); + return 0; + } + + /* Set up digital mute if not provided by the codec */ + if (!codec_dai->driver->ops) { + codec_dai->driver->ops = &ams_delta_dai_ops; + } else { + ams_delta_ops.startup = ams_delta_startup; + ams_delta_ops.shutdown = ams_delta_shutdown; + } + /* Register optional line discipline for over the modem control */ ret = tty_register_ldisc(N_V253, &cx81801_ops); if (ret) { diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 15ccbf479c96..d5ae9eb8c756 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -40,7 +40,7 @@ struct abe_twl6040 { int mclk_freq; /* MCLK frequency speed for twl6040 */ }; -struct platform_device *dmic_codec_dev; +static struct platform_device *dmic_codec_dev; static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 51dd7c65096b..fe966272bd0c 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, switch (channels) { case 6: dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; + /* fall through */ case 4: dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; + /* fall through */ case 2: dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; break; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e97360f9890..4c1be36c2207 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; + /* fall through */ case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; + /* fall through */ case 3: link_mask |= 1 << 2; + /* fall through */ case 2: link_mask |= 1 << 1; + /* fall through */ case 1: link_mask |= 1 << 0; break; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..776e148b0aa2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,19 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP - tristate + tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 7c998ea4ebee..12d4513ebe8a 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -425,8 +425,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) if (priv->sspa == NULL) return -ENOMEM; - priv->dma_params = devm_kzalloc(&pdev->dev, - 2 * sizeof(struct snd_dmaengine_dai_dma_data), + priv->dma_params = devm_kcalloc(&pdev->dev, + 2, sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (priv->dma_params == NULL) return -ENOMEM; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc986080130..69033e1a84e6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,7 +34,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" /* @@ -42,6 +41,8 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; + unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; unsigned int configured_dai_fmt; @@ -105,9 +106,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - - dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - &ssp->drcmr_tx : &ssp->drcmr_rx; + dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "tx" : "rx"; snd_soc_dai_set_dma_data(cpu_dai, substream, dma); @@ -194,21 +194,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) pxa_ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -221,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -265,66 +265,17 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; - if (ssp->type == PXA3xx_SSP) - val &= ~SSACD_SCDX8; - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; - case PXA_SSP_CLK_SCDB_8: - if (ssp->type == PXA3xx_SSP) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; - default: - return -EINVAL; - } - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); - break; - default: - return -ENODEV; - } - - return 0; -} - -/* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); - switch (freq_out) { + switch (freq) { case 5622000: break; case 11345000: @@ -355,7 +306,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, u64 tmp = 19968; tmp *= 1000000; - do_div(tmp, freq_out); + do_div(tmp, freq); val = tmp; val = (val << 16) | 64; @@ -365,7 +316,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, dev_dbg(&ssp->pdev->dev, "Using SSACDD %x to supply %uHz\n", - val, freq_out); + val, freq); break; } @@ -535,6 +486,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; @@ -570,6 +522,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) return 0; } +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + /* * Set the SSP audio DMA parameters and sample size. * Can be called multiple times by oss emulation. @@ -581,11 +551,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -625,11 +596,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, } pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd, acds; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + acds = m->acds; + + /* The values in the table are for 16 bits */ + if (width == 32) + acds--; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { + if (((priv->sysclk / bclk) == 64) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -773,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { @@ -814,8 +840,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, @@ -843,6 +867,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..9f779657bc86 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, @@ -236,7 +214,21 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pxa2xx_ac97_dt_ids[] = { + { .compatible = "marvell,pxa250-ac97", }, + { .compatible = "marvell,pxa270-ac97", }, + { .compatible = "marvell,pxa300-ac97", }, + { } }; +MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); + +#endif static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { @@ -296,6 +288,7 @@ static struct platform_driver pxa2xx_ac97_driver = { #ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif + .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..42820121e5b9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, @@ -366,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 8b6a70e94c01..72eaaef1b426 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,70 +20,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "../../arm/pxa2xx-pcm.h" - -static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma; - - dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - return __pxa2xx_pcm_hw_params(substream, params); -} - -static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - __pxa2xx_pcm_hw_free(substream); - - return 0; -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = __pxa2xx_pcm_open, - .close = __pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = pxa2xx_pcm_hw_params, - .hw_free = pxa2xx_pcm_hw_free, - .prepare = __pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - static const struct snd_soc_component_driver pxa2xx_soc_platform = { .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, @@ -96,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev) NULL, 0); } -#ifdef CONFIG_OF -static const struct of_device_id snd_soc_pxa_audio_match[] = { - { .compatible = "mrvl,pxa-pcm-audio" }, - { } -}; -MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); -#endif - static struct platform_driver pxa_pcm_driver = { .driver = { .name = "pxa-pcm-audio", - .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 87838fa27997..2a4c912d1e48 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. +config SND_SOC_QCOM_COMMON + tristate + config SND_SOC_QDSP6_COMMON tristate @@ -86,7 +89,18 @@ config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. Say Y if you want to use audio device on this SoCs + +config SND_SOC_SDM845 + tristate "SoC Machine driver for SDM845 boards" + depends on QCOM_APR + select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON + help + To add support for audio on Qualcomm Technologies Inc. + SDM845 SoC-based systems. + Say Y if you want to use audio device on this SoCs. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 206945bb9ba1..41b2c7a23a4d 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -14,10 +14,14 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o snd-soc-apq8096-objs := apq8096.o +snd-soc-sdm845-objs := sdm845.o +snd-soc-qcom-common-objs := common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o +obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o +obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 704428735e3c..1dd23bba1bed 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -147,7 +147,8 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) num_links = of_get_child_count(node); /* Allocate the private data and the DAI link array */ - data = devm_kzalloc(dev, sizeof(*data) + sizeof(*link) * num_links, + data = devm_kzalloc(dev, + struct_size(data, dai_link, num_links), GFP_KERNEL); if (!data) return ERR_PTR(-ENOMEM); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 561cd429e6f2..1543e85629f8 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -1,14 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2018, Linaro Limited -#include <linux/soc/qcom/apr.h> #include <linux/module.h> -#include <linux/component.h> #include <linux/platform_device.h> #include <linux/of_device.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> +#include "common.h" static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -24,211 +23,57 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int apq8096_sbc_parse_of(struct snd_soc_card *card) +static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct device_node *np; - struct device_node *codec = NULL; - struct device_node *platform = NULL; - struct device_node *cpu = NULL; - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - int ret, num_links; - - ret = snd_soc_of_parse_card_name(card, "qcom,model"); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - - /* DAPM routes */ - if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, - "qcom,audio-routing"); - if (ret) - return ret; - } - - /* Populate links */ - num_links = of_get_child_count(dev->of_node); + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; - /* Allocate the DAI link array */ - card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; - - card->num_links = num_links; - link = card->dai_link; - - for_each_child_of_node(dev->of_node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); - ret = -EINVAL; - goto err; - } - - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); - if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); - goto err; - } - - platform = of_get_child_by_name(np, "platform"); - codec = of_get_child_by_name(np, "codec"); - if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, - "sound-dai", - 0); - if (!link->platform_of_node) { - dev_err(card->dev, "platform dai not found\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); - goto err; - } - link->no_pcm = 1; - link->ignore_pmdown_time = 1; + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; - link->dynamic = 1; - } - - link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - - link->dpcm_playback = 1; - link->dpcm_capture = 1; - link->stream_name = link->name; link++; } - - return 0; -err: - of_node_put(cpu); - of_node_put(codec); - of_node_put(platform); - kfree(card->dai_link); - return ret; } -static int apq8096_bind(struct device *dev) +static int apq8096_platform_probe(struct platform_device *pdev) { struct snd_soc_card *card; + struct device *dev = &pdev->dev; int ret; card = kzalloc(sizeof(*card), GFP_KERNEL); if (!card) return -ENOMEM; - component_bind_all(dev, card); card->dev = dev; - ret = apq8096_sbc_parse_of(card); + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); goto err; } + apq8096_add_be_ops(card); ret = snd_soc_register_card(card); if (ret) - goto err; + goto err_card_register; return 0; +err_card_register: + kfree(card->dai_link); err: - component_unbind_all(dev, card); kfree(card); return ret; } -static void apq8096_unbind(struct device *dev) +static int apq8096_platform_remove(struct platform_device *pdev) { - struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); snd_soc_unregister_card(card); - component_unbind_all(dev, card); kfree(card->dai_link); kfree(card); -} - -static const struct component_master_ops apq8096_ops = { - .bind = apq8096_bind, - .unbind = apq8096_unbind, -}; - -static int apq8016_compare_of(struct device *dev, void *data) -{ - return dev->of_node == data; -} - -static void apq8016_release_of(struct device *dev, void *data) -{ - of_node_put(data); -} - -static int add_audio_components(struct device *dev, - struct component_match **matchptr) -{ - struct device_node *np, *platform, *cpu, *node, *dai_node; - - node = dev->of_node; - - for_each_child_of_node(node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (cpu) { - dai_node = of_parse_phandle(cpu, "sound-dai", 0); - of_node_get(dai_node); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - - platform = of_get_child_by_name(np, "platform"); - if (platform) { - dai_node = of_parse_phandle(platform, "sound-dai", 0); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - } - - return 0; -} - -static int apq8096_platform_probe(struct platform_device *pdev) -{ - struct component_match *match = NULL; - int ret; - - ret = add_audio_components(&pdev->dev, &match); - if (ret) - return ret; - - return component_master_add_with_match(&pdev->dev, &apq8096_ops, match); -} - -static int apq8096_platform_remove(struct platform_device *pdev) -{ - component_master_del(&pdev->dev, &apq8096_ops); return 0; } @@ -245,7 +90,6 @@ static struct platform_driver msm_snd_apq8096_driver = { .remove = apq8096_platform_remove, .driver = { .name = "msm-snd-apq8096", - .owner = THIS_MODULE, .of_match_table = msm_snd_apq8096_dt_match, }, }; diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c new file mode 100644 index 000000000000..eb1b9da05dd4 --- /dev/null +++ b/sound/soc/qcom/common.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include <linux/module.h> +#include "common.h" + +int qcom_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "Can't find cpu DT node\n"); + ret = -EINVAL; + goto err; + } + + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + if (!link->cpu_of_node) { + dev_err(card->dev, "error getting cpu phandle\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + if (ret) { + dev_err(card->dev, "error getting cpu dai name\n"); + goto err; + } + + platform = of_get_child_by_name(np, "platform"); + codec = of_get_child_by_name(np, "codec"); + if (codec && platform) { + link->platform_of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platform_of_node) { + dev_err(card->dev, "platform dai not found\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "codec dai not found\n"); + goto err; + } + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + } else { + link->platform_of_node = link->cpu_of_node; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + link->dynamic = 1; + } + + link->ignore_suspend = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + link->dpcm_playback = 1; + link->dpcm_capture = 1; + link->stream_name = link->name; + link++; + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} +EXPORT_SYMBOL(qcom_snd_parse_of); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h new file mode 100644 index 000000000000..f05c05b12bd7 --- /dev/null +++ b/sound/soc/qcom/common.h @@ -0,0 +1,11 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_COMMON_H__ +#define __QCOM_SND_COMMON_H__ + +#include <sound/soc.h> + +int qcom_snd_parse_of(struct snd_soc_card *card); + +#endif diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 31fe78aa207f..d07271ea4c45 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return IRQ_NONE; } dev_warn(soc_runtime->dev, "xrun warning\n"); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 9983c665a941..932c3ebfd252 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -64,7 +64,6 @@ struct q6adm { struct aprv2_ibasic_rsp_result_t result; struct mutex lock; wait_queue_head_t matrix_map_wait; - struct platform_device *pdev_routing; }; struct q6adm_cmd_device_open_v5 { @@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close); static int q6adm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6adm *adm; adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); @@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - dais_np = of_get_child_by_name(dev->of_node, "routing"); - if (dais_np) { - adm->pdev_routing = of_platform_device_create(dais_np, - "q6routing", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6adm_remove(struct apr_device *adev) { - struct q6adm *adm = dev_get_drvdata(&adev->dev); - - if (adm->pdev_routing) - of_platform_device_destroy(&adm->pdev_routing->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 9ba95956ada8..60ff4a2d3577 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -4,7 +4,6 @@ #include <linux/err.h> #include <linux/init.h> -#include <linux/component.h> #include <linux/module.h> #include <linux/device.h> #include <linux/platform_device.h> @@ -81,7 +80,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; - slim->num_channels = params_channels(params); slim->sample_rate = params_rate(params); switch (params_format(params)) { @@ -385,23 +383,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai, struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id]; int i; - if (!rx_slot) { - pr_err("%s: rx slot not found\n", __func__); - return -EINVAL; - } + if (dai->id & 0x1) { + /* TX */ + if (!tx_slot) { + pr_err("%s: tx slot not found\n", __func__); + return -EINVAL; + } - for (i = 0; i < rx_num; i++) { - pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: find number of channels[%d] ch[%d]\n", - __func__, i, rx_slot[i]); - } + for (i = 0; i < tx_num; i++) + pcfg->slim.ch_mapping[i] = tx_slot[i]; - pcfg->slim.num_channels = rx_num; + pcfg->slim.num_channels = tx_num; - pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__, - (dai->id - SLIMBUS_0_RX) / 2, rx_num, - pcfg->slim.ch_mapping[0], - pcfg->slim.ch_mapping[1]); + + } else { + if (!rx_slot) { + pr_err("%s: rx slot not found\n", __func__); + return -EINVAL; + } + + for (i = 0; i < rx_num; i++) + pcfg->slim.ch_mapping[i] = rx_slot[i]; + + pcfg->slim.num_channels = rx_num; + + } return 0; } @@ -446,6 +452,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"}, {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"}, + {"SLIMBUS_0_TX", NULL, "Slimbus Capture"}, + {"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"}, + {"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"}, + {"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"}, + {"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"}, + {"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"}, + {"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"}, + {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"}, {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"}, {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, @@ -640,6 +654,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rate_max = 192000, }, }, { + .name = "SLIMBUS_0_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_0_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + }, { .playback = { .stream_name = "Slimbus1 Playback", .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | @@ -658,6 +690,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, }, { + .name = "SLIMBUS_1_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_1_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus1 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + }, { .playback = { .stream_name = "Slimbus2 Playback", .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | @@ -675,6 +725,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_2_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_2_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_2_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus2 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus3 Playback", @@ -693,6 +762,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_3_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_3_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_3_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus3 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus4 Playback", @@ -711,6 +799,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_4_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_4_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_4_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus4 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus5 Playback", @@ -729,6 +836,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_5_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_5_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_5_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus5 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus6 Playback", @@ -747,6 +873,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_6_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_6_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_6_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus6 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Primary MI2S Playback", @@ -975,6 +1120,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", @@ -1252,11 +1404,12 @@ static void of_q6afe_parse_dai_data(struct device *dev, } } -static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) +static int q6afe_dai_dev_probe(struct platform_device *pdev) { struct q6afe_dai_data *dai_data; + struct device *dev = &pdev->dev; - dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL); + dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL); if (!dai_data) return -ENOMEM; @@ -1264,41 +1417,22 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) of_q6afe_parse_dai_data(dev, dai_data); - return snd_soc_register_component(dev, &q6afe_dai_component, + return devm_snd_soc_register_component(dev, &q6afe_dai_component, q6afe_dais, ARRAY_SIZE(q6afe_dais)); } -static void q6afe_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6afe_dai_data *dai_data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - kfree(dai_data); -} - -static const struct component_ops q6afe_dai_comp_ops = { - .bind = q6afe_dai_bind, - .unbind = q6afe_dai_unbind, +static const struct of_device_id q6afe_dai_device_id[] = { + { .compatible = "qcom,q6afe-dais" }, + {}, }; - -static int q6afe_dai_dev_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6afe_dai_comp_ops); -} - -static int q6afe_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6afe_dai_comp_ops); - return 0; -} +MODULE_DEVICE_TABLE(of, q6afe_dai_device_id); static struct platform_driver q6afe_dai_platform_driver = { .driver = { .name = "q6afe-dai", + .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, - .remove = q6afe_dai_dev_remove, }; module_platform_driver(q6afe_dai_platform_driver); diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..000775b4bba8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -316,7 +316,6 @@ struct q6afe { struct mutex lock; struct list_head port_list; spinlock_t port_list_lock; - struct platform_device *pdev_dais; }; struct afe_port_cmd_device_start { @@ -515,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { SLIMBUS_5_RX, 1, 1}, [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, SLIMBUS_6_RX, 1, 1}, + [SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX, + SLIMBUS_0_TX, 0, 1}, + [SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX, + SLIMBUS_1_TX, 0, 1}, + [SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX, + SLIMBUS_2_TX, 0, 1}, + [SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX, + SLIMBUS_3_TX, 0, 1}, + [SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX, + SLIMBUS_4_TX, 0, 1}, + [SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX, + SLIMBUS_5_TX, 0, 1}, + [SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX, + SLIMBUS_6_TX, 0, 1}, [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX, PRIMARY_MI2S_RX, 1, 1}, [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX, @@ -777,7 +790,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1014,7 +1027,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1355,7 +1368,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); } @@ -1373,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_MULTICHAN_HDMI_RX: cfg_type = AFE_PARAM_ID_HDMI_CONFIG; break; + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX: @@ -1438,7 +1458,6 @@ static int q6afe_probe(struct apr_device *adev) { struct q6afe *afe; struct device *dev = &adev->dev; - struct device_node *dais_np; afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); if (!afe) @@ -1453,22 +1472,12 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - afe->pdev_dais = of_platform_device_create(dais_np, - "q6afe-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6afe_remove(struct apr_device *adev) { - struct q6afe *afe = dev_get_drvdata(&adev->dev); - - if (afe->pdev_dais) - of_platform_device_destroy(&afe->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 360936703b3d..9db9a2944ef2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -7,7 +7,6 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/component.h> #include <sound/soc.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -563,14 +562,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; -static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) +static int q6asm_dai_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct device_node *node = dev->of_node; struct of_phandle_args args; struct q6asm_dai_data *pdata; int rc; - pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL); + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) return -ENOMEM; @@ -582,43 +582,23 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(dev, pdata); - return snd_soc_register_component(dev, &q6asm_fe_dai_component, + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, q6asm_fe_dais, ARRAY_SIZE(q6asm_fe_dais)); } -static void q6asm_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6asm_dai_data *pdata = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(pdata); -} - -static const struct component_ops q6asm_dai_comp_ops = { - .bind = q6asm_dai_bind, - .unbind = q6asm_dai_unbind, +static const struct of_device_id q6asm_dai_device_id[] = { + { .compatible = "qcom,q6asm-dais" }, + {}, }; - -static int q6asm_dai_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6asm_dai_comp_ops); -} - -static int q6asm_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6asm_dai_comp_ops); - return 0; -} +MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); static struct platform_driver q6asm_dai_platform_driver = { .driver = { .name = "q6asm-dai", + .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, - .remove = q6asm_dai_dev_remove, }; module_platform_driver(q6asm_dai_platform_driver); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 530852385cad..2b2c7233bb5f 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -174,10 +174,8 @@ struct q6asm { struct device *dev; struct q6core_svc_api_info ainfo; wait_queue_head_t mem_wait; - struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; - struct platform_device *pdev_dais; }; struct audio_client { @@ -1344,7 +1342,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); static int q6asm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6asm *q6asm; q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL); @@ -1359,22 +1356,12 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - q6asm->pdev_dais = of_platform_device_create(dais_np, - "q6asm-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6asm_remove(struct apr_device *adev) { - struct q6asm *q6asm = dev_get_drvdata(&adev->dev); - - if (q6asm->pdev_dais) - of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 7a19d6278406..dc94c5c53788 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -8,7 +8,6 @@ #include <linux/platform_device.h> #include <linux/of_platform.h> #include <linux/bitops.h> -#include <linux/component.h> #include <linux/mutex.h> #include <linux/of_device.h> #include <linux/slab.h> @@ -68,6 +67,13 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ + { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ + { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ + { mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" }, \ + { mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" }, \ + { mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" }, \ + { mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" }, \ { mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"}, \ { mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"}, \ { mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"}, \ @@ -122,6 +128,27 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ @@ -310,7 +337,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode, session->channels, topology, perf_mode, session->bits_per_sample, 0, 0); - if (!copp) { + if (IS_ERR_OR_NULL(copp)) { mutex_unlock(&routing_data->lock); return -EINVAL; } @@ -949,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = { .num_dapm_routes = ARRAY_SIZE(intercon), }; -static int q6routing_dai_bind(struct device *dev, struct device *master, - void *data) +static int q6pcm_routing_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL); if (!routing_data) return -ENOMEM; @@ -961,41 +989,28 @@ static int q6routing_dai_bind(struct device *dev, struct device *master, mutex_init(&routing_data->lock); dev_set_drvdata(dev, routing_data); - return snd_soc_register_component(dev, &msm_soc_routing_component, + return devm_snd_soc_register_component(dev, &msm_soc_routing_component, NULL, 0); } -static void q6routing_dai_unbind(struct device *dev, struct device *master, - void *d) +static int q6pcm_routing_remove(struct platform_device *pdev) { - struct msm_routing_data *data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(data); - + kfree(routing_data); routing_data = NULL; -} - -static const struct component_ops q6routing_dai_comp_ops = { - .bind = q6routing_dai_bind, - .unbind = q6routing_dai_unbind, -}; -static int q6pcm_routing_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6routing_dai_comp_ops); -} - -static int q6pcm_routing_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6routing_dai_comp_ops); return 0; } +static const struct of_device_id q6pcm_routing_device_id[] = { + { .compatible = "qcom,q6adm-routing" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id); + static struct platform_driver q6pcm_routing_platform_driver = { .driver = { .name = "q6routing", + .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, .remove = q6pcm_routing_remove, diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c new file mode 100644 index 000000000000..2a781d87ee65 --- /dev/null +++ b/sound/soc/qcom/sdm845.c @@ -0,0 +1,285 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2018, The Linux Foundation. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "common.h" +#include "qdsp6/q6afe.h" + +#define DEFAULT_SAMPLE_RATE_48K 48000 +#define DEFAULT_MCLK_RATE 24576000 +#define DEFAULT_BCLK_RATE 12288000 + +struct sdm845_snd_data { + struct snd_soc_card *card; + uint32_t pri_mi2s_clk_count; + uint32_t quat_tdm_clk_count; +}; + +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; + +static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + int channels, slot_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 32; + break; + default: + dev_err(rtd->dev, "%s: invalid param format 0x%x\n", + __func__, params_format(params)); + return -EINVAL; + } + + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, + channels, tdm_slot_offset); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } else { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, channels, + tdm_slot_offset, 0, NULL); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } +end: + return ret; +} + +static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + switch (cpu_dai->id) { + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + ret = sdm845_tdm_snd_hw_params(substream, params); + break; + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return ret; +} + +static int sdm845_snd_startup(struct snd_pcm_substream *substream) +{ + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (++(data->pri_mi2s_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(cpu_dai, fmt); + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (++(data->quat_tdm_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return 0; +} + +static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (--(data->pri_mi2s_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + 0, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + }; + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (--(data->quat_tdm_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } +} + +static struct snd_soc_ops sdm845_be_ops = { + .hw_params = sdm845_snd_hw_params, + .startup = sdm845_snd_startup, + .shutdown = sdm845_snd_shutdown, +}; + +static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K; + channels->min = channels->max = 2; + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static void sdm845_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; + + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) { + link->ops = &sdm845_be_ops; + link->be_hw_params_fixup = sdm845_be_hw_params_fixup; + } + link++; + } +} + +static int sdm845_snd_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sdm845_snd_data *data; + struct device *dev = &pdev->dev; + int ret; + + card = kzalloc(sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + /* Allocate the private data */ + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } + + card->dev = dev; + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); + if (ret) { + dev_err(dev, "Error parsing OF data\n"); + goto parse_dt_fail; + } + + data->card = card; + snd_soc_card_set_drvdata(card, data); + + sdm845_add_be_ops(card); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(dev, "Sound card registration failed\n"); + goto register_card_fail; + } + return ret; + +register_card_fail: + kfree(card->dai_link); +parse_dt_fail: + kfree(data); +data_alloc_fail: + kfree(card); + return ret; +} + +static int sdm845_snd_platform_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(data); + kfree(card); + return 0; +} + +static const struct of_device_id sdm845_snd_device_id[] = { + { .compatible = "qcom,sdm845-sndcard" }, + {}, +}; +MODULE_DEVICE_TABLE(of, sdm845_snd_device_id); + +static struct platform_driver sdm845_snd_driver = { + .probe = sdm845_snd_platform_probe, + .remove = sdm845_snd_platform_remove, + .driver = { + .name = "msm-snd-sdm845", + .of_match_table = sdm845_snd_device_id, + }, +}; +module_platform_driver(sdm845_snd_driver); + +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 05b078e7b87f..65e814d46006 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,10 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f184168f9a41..f2a51ae2b674 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -462,7 +462,7 @@ static int rockchip_sound_of_parse_dais(struct device *dev, num_routes = 0; for (i = 0; i < ARRAY_SIZE(rockchip_routes); i++) num_routes += rockchip_routes[i].num_routes; - routes = devm_kzalloc(dev, num_routes * sizeof(*routes), + routes = devm_kcalloc(dev, num_routes, sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 950823d69e9c..60d43d53a8f5 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -22,6 +22,7 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" +#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); return ret; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c new file mode 100644 index 000000000000..f77538319221 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -0,0 +1,45 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "rockchip_pcm.h" + +static const struct snd_pcm_hardware snd_rockchip_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, +}; + +static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { + .pcm_hardware = &snd_rockchip_hardware, + .prealloc_buffer_size = 32 * 1024, +}; + +int rockchip_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h new file mode 100644 index 000000000000..d6c36115c60a --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.h @@ -0,0 +1,14 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_PCM_H +#define _ROCKCHIP_PCM_H + +int rockchip_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 4db4fd56db35..881c32498808 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (!rk_dailink.cpu_of_node) { dev_err(&pdev->dev, "Property 'rockchip,i2s-controller' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_codec_of_node; } rk_dailink.platform_of_node = rk_dailink.cpu_of_node; @@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Soc parse card name failed %d\n", ret); - return ret; + goto put_cpu_of_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "Soc register card failed %d\n", ret); - return ret; + goto put_cpu_of_node; } return ret; + +put_cpu_of_node: + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; +put_codec_of_node: + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return ret; +} + +static int snd_rk_mc_remove(struct platform_device *pdev) +{ + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return 0; } static const struct of_device_id rockchip_rt5645_of_match[] = { @@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f914ed45db7d..d6c62aa13041 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; + /* fall through */ case 4: val |= MOD_DC1_EN; break; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800bf3a8..dc20f0f7080a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 2dc3b762fdd9..922fb6aa3ed1 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -1,16 +1,14 @@ -/* - * SH7760 ("camelot") DMABRG audio DMA unit support - * - * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which - * trigger an interrupt when one half of the programmed transfer size - * has been xmitted. - * - * FIXME: little-endian only for now - */ +// SPDX-License-Identifier: GPL-2.0 +// +// SH7760 ("camelot") DMABRG audio DMA unit support +// +// Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which +// trigger an interrupt when one half of the programmed transfer size +// has been xmitted. +// +// FIXME: little-endian only for now #include <linux/module.h> #include <linux/gfp.h> @@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = { module_platform_driver(sh7760_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3bae06dd121f..aa7e902f0c02 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1,16 +1,12 @@ -/* - * Fifo-attached Serial Interface (FSI) support for SH7724 - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Based on ssi.c - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Fifo-attached Serial Interface (FSI) support for SH7724 +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Based on ssi.c +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> #include <linux/delay.h> #include <linux/dma-mapping.h> diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 624aaf569fef..c2b496398e6b 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -1,13 +1,11 @@ -/* - * Hitachi Audio Controller (AC97) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable HAC output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Hitachi Audio Controller (AC97) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable HAC output pins! /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only * the FIRST can be used since ASoC does not pass any information to the @@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = { module_platform_driver(hac_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index ecb057ff9fbb..8739c9f60672 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -1,12 +1,8 @@ -/* - * ALSA SoC driver for Migo-R - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC driver for Migo-R +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> #include <linux/clkdev.h> #include <linux/device.h> diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 9c3d5aed99d1..5d1ff8ef26f9 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,3 @@ +# SPDX-License-Identifier: GPL-2.0 snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 4672688cac32..3a3064dda57f 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -1,12 +1,9 @@ -/* - * Helper routines for R-Car sound ADG. - * - * Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Helper routines for R-Car sound ADG. +// +// Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk-provider.h> #include "rsnd.h" diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 4221937ae79b..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car CMD support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car CMD support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" struct rsnd_cmd { @@ -89,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); @@ -155,7 +152,7 @@ int rsnd_cmd_probe(struct rsnd_priv *priv) if (!nr) return 0; - cmd = devm_kzalloc(dev, sizeof(*cmd) * nr, GFP_KERNEL); + cmd = devm_kcalloc(dev, nr, sizeof(*cmd), GFP_KERNEL); if (!cmd) return -ENOMEM; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6bbdddef426e..f8425d8b44d2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on fsi.c - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRU/SCU/SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on fsi.c +// Kuninori Morimoto <morimoto.kuninori@renesas.com> /* * Renesas R-Car sound device structure @@ -552,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +static struct snd_soc_dai_driver +*rsnd_daidrv_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= rsnd_rdai_nr(priv))) + return NULL; + + return priv->daidrv + id; +} + #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai) static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) { @@ -1037,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, int io_i; rdai = rsnd_rdai_get(priv, dai_i); - drv = priv->daidrv + dai_i; + drv = rsnd_daidrv_get(priv, dai_i); io_playback = &rdai->playback; io_capture = &rdai->capture; @@ -1116,8 +1121,8 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) if (!nr) return -EINVAL; - rdrv = devm_kzalloc(dev, sizeof(*rdrv) * nr, GFP_KERNEL); - rdai = devm_kzalloc(dev, sizeof(*rdai) * nr, GFP_KERNEL); + rdrv = devm_kcalloc(dev, nr, sizeof(*rdrv), GFP_KERNEL); + rdai = devm_kcalloc(dev, nr, sizeof(*rdai), GFP_KERNEL); if (!rdrv || !rdai) return -ENOMEM; @@ -1612,7 +1617,7 @@ static struct platform_driver rsnd_driver = { }; module_platform_driver(rsnd_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Renesas R-Car audio driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index d201d551866d..6a55aa753003 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -1,12 +1,9 @@ -/* - * ctu.c - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ctu.c +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" #define CTU_NAME_SIZE 16 @@ -378,7 +375,7 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) goto rsnd_ctu_probe_done; } - ctu = devm_kzalloc(dev, sizeof(*ctu) * nr, GFP_KERNEL); + ctu = devm_kcalloc(dev, nr, sizeof(*ctu), GFP_KERNEL); if (!ctu) { ret = -ENOMEM; goto rsnd_ctu_probe_done; diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ef82b94d038b..fe63ef8600d0 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car Audio DMAC support - * - * Copyright (C) 2015 Renesas Electronics Corp. - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Audio DMAC support +// +// Copyright (C) 2015 Renesas Electronics Corp. +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/delay.h> #include <linux/of_dma.h> #include "rsnd.h" diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index dbe54f024d68..2b16e0ce6bc5 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car DVC support - * - * Copyright (C) 2014 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car DVC support +// +// Copyright (C) 2014 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * Playback Volume @@ -344,7 +340,7 @@ int rsnd_dvc_probe(struct rsnd_priv *priv) goto rsnd_dvc_probe_done; } - dvc = devm_kzalloc(dev, sizeof(*dvc) * nr, GFP_KERNEL); + dvc = devm_kcalloc(dev, nr, sizeof(*dvc), GFP_KERNEL); if (!dvc) { ret = -ENOMEM; goto rsnd_dvc_probe_done; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 25642e92dae0..0230301fe078 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car Gen1 SRU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Gen1 SRU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * #define DEBUG diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 7998380766f6..8e3b57eaa708 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -1,12 +1,8 @@ -/* - * mix.c - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// mix.c +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * CTUn MIXn @@ -294,7 +290,7 @@ int rsnd_mix_probe(struct rsnd_priv *priv) goto rsnd_mix_probe_done; } - mix = devm_kzalloc(dev, sizeof(*mix) * nr, GFP_KERNEL); + mix = devm_kcalloc(dev, nr, sizeof(*mix), GFP_KERNEL); if (!mix) { ret = -ENOMEM; goto rsnd_mix_probe_done; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6d7280d2d9be..96d93330b1e1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -1,13 +1,10 @@ -/* - * Renesas R-Car - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #ifndef RSND_H #define RSND_H diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a727e71587b6..beccfbac7581 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car SRC support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRC support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * you can enable below define if you don't need @@ -575,7 +571,7 @@ int rsnd_src_probe(struct rsnd_priv *priv) goto rsnd_src_probe_done; } - src = devm_kzalloc(dev, sizeof(*src) * nr, GFP_KERNEL); + src = devm_kcalloc(dev, nr, sizeof(*src), GFP_KERNEL); if (!src) { ret = -ENOMEM; goto rsnd_src_probe_done; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 98dd120d830a..8304e4ec9242 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on fsi.c - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on fsi.c +// Kuninori Morimoto <morimoto.kuninori@renesas.com> /* * you can enable below define if you don't need @@ -1127,7 +1123,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) goto rsnd_ssi_probe_done; } - ssi = devm_kzalloc(dev, sizeof(*ssi) * nr, GFP_KERNEL); + ssi = devm_kcalloc(dev, nr, sizeof(*ssi), GFP_KERNEL); if (!ssi) { ret = -ENOMEM; goto rsnd_ssi_probe_done; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 6ff8a36c2c82..016fbf5ac242 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -1,12 +1,9 @@ -/* - * Renesas R-Car SSIU support - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU support +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" #define SSIU_NAME "ssiu" @@ -258,7 +255,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) /* same number to SSI */ nr = priv->ssi_nr; - ssiu = devm_kzalloc(dev, sizeof(*ssiu) * nr, GFP_KERNEL); + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 4a3568a9bf59..4bb4c13cf860 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -1,10 +1,8 @@ -/* - * Generic AC97 sound support for SH7760 - * - * (c) 2007 Manuel Lauss - * - * Licensed under the GPLv2. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Generic AC97 sound support for SH7760 +// +// (c) 2007 Manuel Lauss #include <linux/module.h> #include <linux/moduleparam.h> @@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void) module_init(sh7760_ac97_init); module_exit(sh7760_ac97_exit); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 6088d627c0e4..63a508fdfe78 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -1,23 +1,9 @@ -/* - * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> #ifndef SIU_H #define SIU_H diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index ee2211635e92..f2a386fcd92e 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -1,23 +1,9 @@ -/* - * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> #include <linux/delay.h> #include <linux/firmware.h> diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 172909570ed5..e263757e4a69 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -1,23 +1,10 @@ -/* - * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> + #include <linux/delay.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 89ed1b107ac5..8125fa3840b6 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -1,14 +1,11 @@ -/* - * Serial Sound Interface (I2S) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable SSI output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Serial Sound Interface (I2S) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable SSI output pins! /* * LIMITATIONS: @@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = { module_platform_driver(sh4_ssi_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf969d0..d70fcd4a1adf 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f214bca..c086786e4471 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> #include <linux/ctype.h> #include <linux/delay.h> diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 3d7e1ff79139..b8e72b52db30 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,18 +1,8 @@ -/* - * soc-apci.c - support for ACPI enumeration. - * - * Copyright (c) 2013-15, Intel Corporation. - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// soc-apci.c - support for ACPI enumeration. +// +// Copyright (c) 2013-15, Intel Corporation. #include <sound/soc-acpi.h> diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index e095115fa9f9..409d082e80d1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -1,18 +1,12 @@ -/* - * soc-compress.c -- ALSA SoC Compress - * - * Copyright (C) 2012 Intel Corp. - * - * Authors: Namarta Kohli <namartax.kohli@intel.com> - * Ramesh Babu K V <ramesh.babu@linux.intel.com> - * Vinod Koul <vinod.koul@linux.intel.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-compress.c -- ALSA SoC Compress +// +// Copyright (C) 2012 Intel Corp. +// +// Authors: Namarta Kohli <namartax.kohli@intel.com> +// Ramesh Babu K V <ramesh.babu@linux.intel.com> +// Vinod Koul <vinod.koul@linux.intel.com> #include <linux/kernel.h> #include <linux/init.h> @@ -146,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) stream = SNDRV_PCM_STREAM_CAPTURE; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto be_err; + else if (ret == 0) + dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + fe->dpcm[stream].runtime = fe_substream->runtime; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto out; + } if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -159,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = soc_compr_components_open(cstream, &component); if (ret < 0) - goto machine_err; + goto open_err; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { ret = fe->dai_link->compr_ops->startup(cstream); @@ -170,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) } } - fe->dpcm[stream].runtime = fe_substream->runtime; - - ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) - goto fe_err; - else if (ret == 0) - dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", - fe->dai_link->name, stream ? "capture" : "playback"); - - /* calculate valid and active FE <-> BE dpcms */ - dpcm_process_paths(fe, stream, &list, 1); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { - /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - goto path_err; - } - dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); @@ -207,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; -path_err: - dpcm_path_put(&list); -fe_err: - if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) - fe->dai_link->compr_ops->shutdown(cstream); machine_err: soc_compr_components_free(cstream, component); - +open_err: if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: + dpcm_path_put(&list); +be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); return ret; @@ -557,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) { ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai); if (ret < 0) @@ -583,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, goto out; } - /* - * Create an empty hw_params for the BE as the machine driver must - * fix this up to match DSP decoder and ASRC configuration. - * I.e. machine driver fixup for compressed BE is mandatory. - */ - memset(&fe->dpcm[fe_substream->stream].hw_params, 0, - sizeof(struct snd_pcm_hw_params)); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) - goto out; - - ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) - goto out; - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3d56f1fe5914..9cfe10d8040c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1,26 +1,21 @@ -/* - * soc-core.c -- ALSA SoC Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * o Add hw rules to enforce rates, etc. - * o More testing with other codecs/machines. - * o Add more codecs and platforms to ensure good API coverage. - * o Support TDM on PCM and I2S - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-core.c -- ALSA SoC Audio Layer +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> +// +// TODO: +// o Add hw rules to enforce rates, etc. +// o More testing with other codecs/machines. +// o Add more codecs and platforms to ensure good API coverage. +// o Support TDM on PCM and I2S #include <linux/module.h> #include <linux/moduleparam.h> @@ -373,8 +368,8 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( if (!rtd->dai_link->ops) rtd->dai_link->ops = &null_snd_soc_ops; - rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) * - dai_link->num_codecs, + rtd->codec_dais = kcalloc(dai_link->num_codecs, + sizeof(struct snd_soc_dai *), GFP_KERNEL); if (!rtd->codec_dais) { kfree(rtd); @@ -533,6 +528,7 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + /* fall through */ case SND_SOC_BIAS_OFF: if (component->driver->suspend) @@ -852,6 +848,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1195,15 +1194,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); +static void soc_set_of_name_prefix(struct snd_soc_component *component) +{ + struct device_node *component_of_node = component->dev->of_node; + const char *str; + int ret; + + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + ret = of_property_read_string(component_of_node, "sound-name-prefix", + &str); + if (!ret) + component->name_prefix = str; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { int i; - if (card->codec_conf == NULL) - return; - - for (i = 0; i < card->num_configs; i++) { + for (i = 0; i < card->num_configs && card->codec_conf; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; struct device_node *component_of_node = component->dev->of_node; @@ -1215,8 +1226,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card, if (map->dev_name && strcmp(component->name, map->dev_name)) continue; component->name_prefix = map->name_prefix; - break; + return; } + + /* + * If there is no configuration table or no match in the table, + * check if a prefix is provided in the node + */ + soc_set_of_name_prefix(component); } static int soc_probe_component(struct snd_soc_card *card, @@ -1461,7 +1478,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1526,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1557,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1884,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1961,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); @@ -2523,6 +2632,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); /** + * snd_soc_dai_get_channel_map - Get DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + */ +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->get_channel_map) + return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable @@ -3258,9 +3389,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); -static int snd_soc_of_get_slot_mask(struct device_node *np, - const char *prop_name, - unsigned int *mask) +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) { u32 val; const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); @@ -3275,6 +3406,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return val; } +EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, @@ -3354,7 +3486,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + routes = devm_kcalloc(card->dev, num_routes, sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -3678,8 +3810,8 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, dev_err(dev, "Bad phandle in 'sound-dai'\n"); return num_codecs; } - component = devm_kzalloc(dev, - sizeof *component * num_codecs, + component = devm_kcalloc(dev, + num_codecs, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8ede773b1db8..7e96793050c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1,27 +1,21 @@ -/* - * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Features: - * o Changes power status of internal codec blocks depending on the - * dynamic configuration of codec internal audio paths and active - * DACs/ADCs. - * o Platform power domain - can support external components i.e. amps and - * mic/headphone insertion events. - * o Automatic Mic Bias support - * o Jack insertion power event initiation - e.g. hp insertion will enable - * sinks, dacs, etc - * o Delayed power down of audio subsystem to reduce pops between a quick - * device reopen. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-dapm.c -- ALSA SoC Dynamic Audio Power Management +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// +// Features: +// o Changes power status of internal codec blocks depending on the +// dynamic configuration of codec internal audio paths and active +// DACs/ADCs. +// o Platform power domain - can support external components i.e. amps and +// mic/headphone insertion events. +// o Automatic Mic Bias support +// o Jack insertion power event initiation - e.g. hp insertion will enable +// sinks, dacs, etc +// o Delayed power down of audio subsystem to reduce pops between a quick +// device reopen. #include <linux/module.h> #include <linux/moduleparam.h> @@ -1086,7 +1080,7 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, list_for_each(it, widgets) size++; - *list = kzalloc(sizeof(**list) + size * sizeof(*w), GFP_KERNEL); + *list = kzalloc(struct_size(*list, widgets, size), GFP_KERNEL); if (*list == NULL) return -ENOMEM; @@ -3055,7 +3049,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) continue; if (w->num_kcontrols) { - w->kcontrols = kzalloc(w->num_kcontrols * + w->kcontrols = kcalloc(w->num_kcontrols, sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { @@ -3662,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745df1412..a9ea172a66a7 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -1,13 +1,8 @@ -/* - * soc-devres.c -- ALSA SoC Audio Layer devres functions - * - * Copyright (C) 2013 Linaro Ltd - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-devres.c -- ALSA SoC Audio Layer devres functions +// +// Copyright (C) 2013 Linaro Ltd #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..52fd7af952a5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -1,17 +1,8 @@ -/* - * Copyright (C) 2013, Analog Devices Inc. - * Author: Lars-Peter Clausen <lars@metafoo.de> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2013, Analog Devices Inc. +// Author: Lars-Peter Clausen <lars@metafoo.de> + #include <linux/module.h> #include <linux/init.h> #include <linux/dmaengine.h> @@ -197,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea case 32: case 64: if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); + hw.formats |= pcm_format_to_bits(i); break; default: /* Unsupported types */ @@ -343,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -359,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 026cd5347e53..1ff9175e9d5e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -1,15 +1,10 @@ -/* - * soc-io.c -- ASoC register I/O helpers - * - * Copyright 2009-2011 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-io.c -- ASoC register I/O helpers +// +// Copyright 2009-2011 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> #include <linux/i2c.h> #include <linux/spi/spi.h> diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b2b16044ae80..c7b990abdbaa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -1,15 +1,10 @@ -/* - * soc-jack.c -- ALSA SoC jack handling - * - * Copyright 2008 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-jack.c -- ALSA SoC jack handling +// +// Copyright 2008 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> #include <sound/jack.h> #include <sound/soc.h> diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7144a51ddfa9..592efb370c44 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -1,20 +1,15 @@ -/* - * soc-ops.c -- Generic ASoC operations - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ops.c -- Generic ASoC operations +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5feae9666822..e8b98bfd4cf1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1,20 +1,14 @@ -/* - * soc-pcm.c -- ALSA SoC PCM - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Authors: Liam Girdwood <lrg@ti.com> - * Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-pcm.c -- ALSA SoC PCM +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Authors: Liam Girdwood <lrg@ti.com> +// Mark Brown <broonie@opensource.wolfsonmicro.com> #include <linux/kernel.h> #include <linux/init.h> @@ -448,6 +442,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->close) + continue; + + component->driver->ops->close(substream); + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -462,7 +479,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; const char *codec_dai_name = "multicodec"; - int i, ret = 0, __ret; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); for (i = 0; i < rtd->num_codecs; i++) @@ -486,7 +503,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -494,16 +510,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) !component->driver->ops->open) continue; - __ret = component->driver->ops->open(substream); - if (__ret < 0) { + ret = component->driver->ops->open(substream); + if (ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; @@ -612,15 +627,7 @@ codec_dai_err: } component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, component); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -714,15 +721,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, NULL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { @@ -860,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { @@ -874,6 +885,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->hw_free) + continue; + + component->driver->ops->hw_free(substream); + } + + return 0; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -886,7 +920,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret = 0, __ret; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops->hw_params) { @@ -944,7 +978,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto interface_err; - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -952,16 +985,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, !component->driver->ops->hw_params) continue; - __ret = component->driver->ops->hw_params(substream, params); - if (__ret < 0) { + ret = component->driver->ops->hw_params(substream, params); + if (ret < 0) { dev_err(component->dev, "ASoC: %s hw params failed: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); @@ -977,15 +1009,7 @@ out: return ret; component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, component); if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -1014,8 +1038,6 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; @@ -1052,15 +1074,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) rtd->dai_link->ops->hw_free(substream); /* free any component resources */ - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { @@ -1165,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous total delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1176,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) offset = component->driver->ops->pointer(substream); break; } + /* base delay if assigned in pointer callback */ + delay = runtime->delay; if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); @@ -1658,29 +1677,28 @@ unwind: } static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *stream, - u64 formats) + struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) - runtime->hw.formats &= formats & stream->formats; + runtime->hw.formats &= stream->formats; else - runtime->hw.formats = formats & stream->formats; + runtime->hw.formats = stream->formats; runtime->hw.rates = stream->rates; } -static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) +static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, + u64 *formats) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; - u64 formats = ULLONG_MAX; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) - return formats; + return; /* * It returns merged BE codec format @@ -1708,11 +1726,118 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) else codec_stream = &codec_dai_drv->capture; - formats &= codec_stream->formats; + *formats &= codec_stream->formats; } } +} + +static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; - return formats; + if (!fe->dai_link->dpcm_merged_chan) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *channels_min = max(*channels_min, + codec_stream->channels_min); + *channels_max = min(*channels_max, + codec_stream->channels_max); + } + } +} + +static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, + unsigned int *rates, + unsigned int *rate_min, + unsigned int *rate_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_rate) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *rate_min = max(*rate_min, cpu_stream->rate_min); + *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); + + for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *rate_min = max(*rate_min, codec_stream->rate_min); + *rate_max = min_not_zero(*rate_max, + codec_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, + codec_stream->rates); + } + } } static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) @@ -1721,12 +1846,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - u64 format = dpcm_runtime_base_format(substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + + dpcm_runtime_merge_format(substream, &runtime->hw.formats); + dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, + &runtime->hw.channels_max); + dpcm_runtime_merge_rate(substream, &runtime->hw.rates, + &runtime->hw.rate_min, &runtime->hw.rate_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); @@ -2551,106 +2681,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3fd5d9c867b9..66e77e020745 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1,29 +1,24 @@ -/* - * soc-topology.c -- ALSA SoC Topology - * - * Copyright (C) 2012 Texas Instruments Inc. - * Copyright (C) 2015 Intel Corporation. - * - * Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com> - * K, Mythri P <mythri.p.k@intel.com> - * Prusty, Subhransu S <subhransu.s.prusty@intel.com> - * B, Jayachandran <jayachandran.b@intel.com> - * Abdullah, Omair M <omair.m.abdullah@intel.com> - * Jin, Yao <yao.jin@intel.com> - * Lin, Mengdong <mengdong.lin@intel.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Add support to read audio firmware topology alongside firmware text. The - * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, - * equalizers, firmware, coefficients etc. - * - * This file only manages the core ALSA and ASoC components, all other bespoke - * firmware topology data is passed to component drivers for bespoke handling. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-topology.c -- ALSA SoC Topology +// +// Copyright (C) 2012 Texas Instruments Inc. +// Copyright (C) 2015 Intel Corporation. +// +// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com> +// K, Mythri P <mythri.p.k@intel.com> +// Prusty, Subhransu S <subhransu.s.prusty@intel.com> +// B, Jayachandran <jayachandran.b@intel.com> +// Abdullah, Omair M <omair.m.abdullah@intel.com> +// Jin, Yao <yao.jin@intel.com> +// Lin, Mengdong <mengdong.lin@intel.com> +// +// Add support to read audio firmware topology alongside firmware text. The +// topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, +// equalizers, firmware, coefficients etc. +// +// This file only manages the core ALSA and ASoC components, all other bespoke +// firmware topology data is passed to component drivers for bespoke handling. #include <linux/kernel.h> #include <linux/export.h> @@ -259,7 +254,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -291,7 +286,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -302,27 +298,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -643,7 +642,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -885,7 +885,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, int i, ret; se->dobj.control.dtexts = - kzalloc(sizeof(char *) * ec->items, GFP_KERNEL); + kcalloc(ec->items, sizeof(char *), GFP_KERNEL); if (se->dobj.control.dtexts == NULL) return -ENOMEM; @@ -1100,6 +1100,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic kcontrol to component driver. */ +static int soc_tplg_add_route(struct soc_tplg *tplg, + struct snd_soc_dapm_route *route) +{ + if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + return tplg->ops->dapm_route_load(tplg->comp, tplg->index, + route); + + return 0; +} + static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1148,6 +1159,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, else route.control = elem->control; + soc_tplg_add_route(tplg, &route); + /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, &route, 1); } @@ -1702,7 +1715,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1772,7 +1785,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2080,7 +2093,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2202,7 +2215,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2311,7 +2324,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index a863bb3f66c2..e0c93496c0cd 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -1,17 +1,11 @@ -/* - * soc-util.c -- ALSA SoC Audio Layer utility functions - * - * Copyright 2009 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * Liam Girdwood <lrg@slimlogic.co.uk> - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-util.c -- ALSA SoC Audio Layer utility functions +// +// Copyright 2009 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> +// Liam Girdwood <lrg@slimlogic.co.uk> #include <linux/platform_device.h> #include <linux/export.h> diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e..313dab2857ef 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index ee0055e60852..7b63d35ef428 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(reader->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 48f9ddd94016..9b2681397dba 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO + select SND_PCM_IEC958 help Say Y if you want to enable SAI for STM32 diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..706ff005234f 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { @@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd) static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; - struct snd_soc_pcm_runtime *rtd; - struct stm32_adfsdm_priv *priv; substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - if (substream) { - rtd = substream->private_data; - priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - + if (substream) snd_pcm_lib_preallocate_free_for_all(pcm); - } } static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index cfeb219e1d78..06fba9650ac4 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -96,7 +96,8 @@ * @slot_mask: rx or tx active slots mask. set at init or at runtime * @data_size: PCM data width. corresponds to PCM substream width. * @spdif_frm_cnt: S/PDIF playback frame counter - * @spdif_status_bits: S/PDIF status bits + * @snd_aes_iec958: iec958 data + * @ctrl_lock: control lock */ struct stm32_sai_sub_data { struct platform_device *pdev; @@ -125,7 +126,8 @@ struct stm32_sai_sub_data { int slot_mask; int data_size; unsigned int spdif_frm_cnt; - unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES]; + struct snd_aes_iec958 iec958; + struct mutex ctrl_lock; /* protect resources accessed by controls */ }; enum stm32_sai_fifo_th { @@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) } } -static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = { - 0, 0, 0, IEC958_AES3_CON_FS_48000, -}; - static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, @@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .fast_io = true, }; +static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(uctl->value.iec958.status, sai->iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(sai->iec958.status, uctl->value.iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static const struct snd_kcontrol_new iec958_ctls = { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_pcm_iec958_info, + .get = snd_pcm_iec958_get, + .put = snd_pcm_iec958_put, +}; + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -259,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) status = SNDRV_PCM_STATE_XRUN; } - if (status != SNDRV_PCM_STATE_RUNNING) { - snd_pcm_stream_lock(sai->substream); - snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(sai->substream); - } + if (status != SNDRV_PCM_STATE_RUNNING) + snd_pcm_stop_xrun(sai->substream); return IRQ_HANDLED; } @@ -619,6 +657,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) } } +static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai) +{ + unsigned char *cs = sai->iec958.status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_GENERAL; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID; +} + +static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai, + struct snd_pcm_runtime *runtime) +{ + if (!runtime) + return; + + /* Force the sample rate according to runtime rate */ + mutex_lock(&sai->ctrl_lock); + switch (runtime->rate) { + case 22050: + sai->iec958.status[3] = IEC958_AES3_CON_FS_22050; + break; + case 44100: + sai->iec958.status[3] = IEC958_AES3_CON_FS_44100; + break; + case 88200: + sai->iec958.status[3] = IEC958_AES3_CON_FS_88200; + break; + case 176400: + sai->iec958.status[3] = IEC958_AES3_CON_FS_176400; + break; + case 24000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_24000; + break; + case 48000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_48000; + break; + case 96000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_192000; + break; + case 32000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_32000; + break; + default: + sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID; + break; + } + mutex_unlock(&sai->ctrl_lock); +} + static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { @@ -709,7 +800,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream, sai->data_size = params_width(params); - if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + /* Rate not already set in runtime structure */ + substream->runtime->rate = params_rate(params); + stm32_sai_set_iec958_status(sai, substream->runtime); + } else { ret = stm32_sai_set_slots(cpu_dai); if (ret < 0) return ret; @@ -789,6 +884,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, sai->substream = NULL; } +static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__); + return snd_ctl_add(rtd->pcm->card, + snd_ctl_new1(&iec958_ctls, sai)); + } + + return 0; +} + static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); @@ -809,6 +918,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); + /* Next settings are not relevant for spdif mode */ + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + return 0; + cr1_mask = SAI_XCR1_RX_TX; if (STM_SAI_IS_CAPTURE(sai)) cr1 |= SAI_XCR1_RX_TX; @@ -820,10 +933,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) sai->synco, sai->synci); } - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - memcpy(sai->spdif_status_bits, default_status_bits, - sizeof(default_status_bits)); - cr1_mask |= SAI_XCR1_SYNCEN_MASK; cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync); @@ -861,7 +970,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, /* Set channel status bit */ byte = frm_cnt >> 3; mask = 1 << (frm_cnt - (byte << 3)); - if (sai->spdif_status_bits[byte] & mask) + if (sai->iec958.status[byte] & mask) *ptr |= 0x04000000; ptr++; @@ -888,6 +997,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { { .probe = stm32_sai_dai_probe, + .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ .playback = { .channels_min = 1, @@ -998,6 +1108,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n"); return -EINVAL; } + stm32_sai_init_iec958_status(sai); sai->spdif = true; sai->master = true; } @@ -1114,6 +1225,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; + mutex_init(&sai->ctrl_lock); platform_set_drvdata(pdev, sai); sai->pdata = dev_get_drvdata(pdev->dev.parent); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 0e4805c7b4ca..7081f15302cc 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + tegra_rt5677_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; err: return ret; } @@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + tegra_rt5677_dai.platform_of_node = NULL; + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + return 0; } diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -265,6 +265,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } /** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + +/** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only * @rate: Sampling rate in Hz. @@ -276,7 +327,7 @@ void aio_port_reset(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -395,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -460,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 80daec17be25..ee90e6c3937c 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_out) { struct uniphier_aio *aio = uniphier_priv(dai); - struct device *dev = &aio->chip->pdev->dev; int ret; if (!is_valid_pll(aio->chip, pll_id)) return -EINVAL; - if (!aio->chip->plls[pll_id].enable) { - dev_err(dev, "PLL(%d) is not implemented\n", pll_id); - return -ENOTSUPP; - } ret = aio_chip_set_pll(aio->chip, pll_id, freq_out); if (ret < 0) @@ -624,15 +619,17 @@ int uniphier_aio_probe(struct platform_device *pdev) return PTR_ERR(chip->rst); chip->num_aios = chip->chip_spec->num_dais; - chip->aios = devm_kzalloc(dev, - sizeof(struct uniphier_aio) * chip->num_aios, + chip->aios = devm_kcalloc(dev, + chip->num_aios, sizeof(struct uniphier_aio), GFP_KERNEL); if (!chip->aios) return -ENOMEM; chip->num_plls = chip->chip_spec->num_plls; - chip->plls = devm_kzalloc(dev, sizeof(struct uniphier_aio_pll) * - chip->num_plls, GFP_KERNEL); + chip->plls = devm_kcalloc(dev, + chip->num_plls, + sizeof(struct uniphier_aio_pll), + GFP_KERNEL); if (!chip->plls) return -ENOMEM; memcpy(chip->plls, chip->chip_spec->plls, diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * @@ -322,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { diff --git a/sound/sound_core.c b/sound/sound_core.c index b4efb22db561..40ad000c2e3c 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -413,7 +413,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit, break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, - name, S_IRUSR | S_IWUSR, dev); + name, 0600, dev); } EXPORT_SYMBOL(register_sound_special_device); @@ -440,7 +440,7 @@ EXPORT_SYMBOL(register_sound_special); int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, - "mixer", S_IRUSR | S_IWUSR, NULL); + "mixer", 0600, NULL); } EXPORT_SYMBOL(register_sound_mixer); @@ -468,7 +468,7 @@ EXPORT_SYMBOL(register_sound_mixer); int register_sound_dsp(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[3], fops, dev, 3, 131, - "dsp", S_IWUSR | S_IRUSR, NULL); + "dsp", 0600, NULL); } EXPORT_SYMBOL(register_sound_dsp); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index abc7bd5055eb..7609eceba1a2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2518,7 +2518,7 @@ static void snd_dbri_proc(struct snd_card *card) #ifdef DBRI_DEBUG if (!snd_card_proc_new(card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); - entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ + entry->mode = S_IFREG | 0444; /* Readable only. */ } #endif } @@ -2542,7 +2542,7 @@ static int snd_dbri_create(struct snd_card *card, dbri->irq = irq; dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_ATOMIC); + &dbri->dma_dvma, GFP_KERNEL); if (!dbri->dma) return -ENOMEM; diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 224a6a5d1c0e..2dd2518a71d3 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -591,12 +591,14 @@ static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt) int i; for (i = 0; i < PCM_N_URBS; i++) { - rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB - * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + rt->out_urbs[i].buffer = kcalloc(PCM_MAX_PACKET_SIZE, + PCM_N_PACKETS_PER_URB, + GFP_KERNEL); if (!rt->out_urbs[i].buffer) return -ENOMEM; - rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB - * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + rt->in_urbs[i].buffer = kcalloc(PCM_MAX_PACKET_SIZE, + PCM_N_PACKETS_PER_URB, + GFP_KERNEL); if (!rt->in_urbs[i].buffer) return -ENOMEM; } diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index fb1c1eac0b5e..f35d29f49ffe 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -728,7 +728,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); - urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); + urbs = kmalloc_array(N_URBS, sizeof(*urbs), GFP_KERNEL); if (!urbs) { *ret = -ENOMEM; return NULL; @@ -742,7 +742,8 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) } urbs[i]->transfer_buffer = - kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); + kmalloc_array(BYTES_PER_FRAME, FRAMES_PER_URB, + GFP_KERNEL); if (!urbs[i]->transfer_buffer) { *ret = -ENOMEM; return urbs; @@ -857,7 +858,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) &snd_usb_caiaq_ops); cdev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc_array(N_URBS, sizeof(struct snd_usb_caiaq_cb_info), GFP_KERNEL); if (!cdev->data_cb_info) diff --git a/sound/usb/card.c b/sound/usb/card.c index 4a1c6bb3dfa0..a1ed798a1c6b 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -86,6 +86,8 @@ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +bool snd_usb_use_vmalloc = true; + module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); module_param_array(id, charp, NULL, 0444); @@ -105,6 +107,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); +MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging @@ -221,32 +225,13 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; struct usb_interface_descriptor *altsd; - void *control_header; int i, protocol; - int rest_bytes; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - control_header = snd_usb_find_csint_desc(host_iface->extra, - host_iface->extralen, - NULL, UAC_HEADER); altsd = get_iface_desc(host_iface); protocol = altsd->bInterfaceProtocol; - if (!control_header) { - dev_err(&dev->dev, "cannot find UAC_HEADER\n"); - return -EINVAL; - } - - rest_bytes = (void *)(host_iface->extra + host_iface->extralen) - - control_header; - - /* just to be sure -- this shouldn't hit at all */ - if (rest_bytes <= 0) { - dev_err(&dev->dev, "invalid control header\n"); - return -EINVAL; - } - switch (protocol) { default: dev_warn(&dev->dev, @@ -255,7 +240,25 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) /* fall through */ case UAC_VERSION_1: { - struct uac1_ac_header_descriptor *h1 = control_header; + struct uac1_ac_header_descriptor *h1; + int rest_bytes; + + h1 = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, UAC_HEADER); + if (!h1) { + dev_err(&dev->dev, "cannot find UAC_HEADER\n"); + return -EINVAL; + } + + rest_bytes = (void *)(host_iface->extra + + host_iface->extralen) - (void *)h1; + + /* just to be sure -- this shouldn't hit at all */ + if (rest_bytes <= 0) { + dev_err(&dev->dev, "invalid control header\n"); + return -EINVAL; + } if (rest_bytes < sizeof(*h1)) { dev_err(&dev->dev, "too short v1 buffer descriptor\n"); @@ -308,6 +311,20 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) return -EINVAL; } + if (protocol == UAC_VERSION_3) { + int badd = assoc->bFunctionSubClass; + + if (badd != UAC3_FUNCTION_SUBCLASS_FULL_ADC_3_0 && + (badd < UAC3_FUNCTION_SUBCLASS_GENERIC_IO || + badd > UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE)) { + dev_err(&dev->dev, + "Unsupported UAC3 BADD profile\n"); + return -EINVAL; + } + + chip->badd_profile = badd; + } + for (i = 0; i < assoc->bInterfaceCount; i++) { int intf = assoc->bFirstInterface + i; @@ -329,8 +346,9 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * */ -static int snd_usb_audio_free(struct snd_usb_audio *chip) +static void snd_usb_audio_free(struct snd_card *card) { + struct snd_usb_audio *chip = card->private_data; struct snd_usb_endpoint *ep, *n; list_for_each_entry_safe(ep, n, &chip->ep_list, list) @@ -339,14 +357,90 @@ static int snd_usb_audio_free(struct snd_usb_audio *chip) mutex_destroy(&chip->mutex); if (!atomic_read(&chip->shutdown)) dev_set_drvdata(&chip->dev->dev, NULL); - kfree(chip); - return 0; } -static int snd_usb_audio_dev_free(struct snd_device *device) +static void usb_audio_make_shortname(struct usb_device *dev, + struct snd_usb_audio *chip, + const struct snd_usb_audio_quirk *quirk) { - struct snd_usb_audio *chip = device->device_data; - return snd_usb_audio_free(chip); + struct snd_card *card = chip->card; + + if (quirk && quirk->product_name && *quirk->product_name) { + strlcpy(card->shortname, quirk->product_name, + sizeof(card->shortname)); + return; + } + + /* retrieve the device string as shortname */ + if (!dev->descriptor.iProduct || + usb_string(dev, dev->descriptor.iProduct, + card->shortname, sizeof(card->shortname)) <= 0) { + /* no name available from anywhere, so use ID */ + sprintf(card->shortname, "USB Device %#04x:%#04x", + USB_ID_VENDOR(chip->usb_id), + USB_ID_PRODUCT(chip->usb_id)); + } + + strim(card->shortname); +} + +static void usb_audio_make_longname(struct usb_device *dev, + struct snd_usb_audio *chip, + const struct snd_usb_audio_quirk *quirk) +{ + struct snd_card *card = chip->card; + int len; + + /* shortcut - if any pre-defined string is given, use it */ + if (quirk && quirk->profile_name && *quirk->profile_name) { + strlcpy(card->longname, quirk->profile_name, + sizeof(card->longname)); + return; + } + + if (quirk && quirk->vendor_name && *quirk->vendor_name) { + len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname)); + } else { + /* retrieve the vendor and device strings as longname */ + if (dev->descriptor.iManufacturer) + len = usb_string(dev, dev->descriptor.iManufacturer, + card->longname, sizeof(card->longname)); + else + len = 0; + /* we don't really care if there isn't any vendor string */ + } + if (len > 0) { + strim(card->longname); + if (*card->longname) + strlcat(card->longname, " ", sizeof(card->longname)); + } + + strlcat(card->longname, card->shortname, sizeof(card->longname)); + + len = strlcat(card->longname, " at ", sizeof(card->longname)); + + if (len < sizeof(card->longname)) + usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); + + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + strlcat(card->longname, ", low speed", sizeof(card->longname)); + break; + case USB_SPEED_FULL: + strlcat(card->longname, ", full speed", sizeof(card->longname)); + break; + case USB_SPEED_HIGH: + strlcat(card->longname, ", high speed", sizeof(card->longname)); + break; + case USB_SPEED_SUPER: + strlcat(card->longname, ", super speed", sizeof(card->longname)); + break; + case USB_SPEED_SUPER_PLUS: + strlcat(card->longname, ", super speed plus", sizeof(card->longname)); + break; + default: + break; + } } /* @@ -360,11 +454,8 @@ static int snd_usb_audio_create(struct usb_interface *intf, { struct snd_card *card; struct snd_usb_audio *chip; - int err, len; + int err; char component[14]; - static struct snd_device_ops ops = { - .dev_free = snd_usb_audio_dev_free, - }; *rchip = NULL; @@ -382,18 +473,13 @@ static int snd_usb_audio_create(struct usb_interface *intf, } err = snd_card_new(&intf->dev, index[idx], id[idx], THIS_MODULE, - 0, &card); + sizeof(*chip), &card); if (err < 0) { dev_err(&dev->dev, "cannot create card instance %d\n", idx); return err; } - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { - snd_card_free(card); - return -ENOMEM; - } - + chip = card->private_data; mutex_init(&chip->mutex); init_waitqueue_head(&chip->shutdown_wait); chip->index = idx; @@ -411,75 +497,15 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_usb_audio_free(chip); - snd_card_free(card); - return err; - } + card->private_free = snd_usb_audio_free; strcpy(card->driver, "USB-Audio"); sprintf(component, "USB%04x:%04x", USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); snd_component_add(card, component); - /* retrieve the device string as shortname */ - if (quirk && quirk->product_name && *quirk->product_name) { - strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname)); - } else { - if (!dev->descriptor.iProduct || - usb_string(dev, dev->descriptor.iProduct, - card->shortname, sizeof(card->shortname)) <= 0) { - /* no name available from anywhere, so use ID */ - sprintf(card->shortname, "USB Device %#04x:%#04x", - USB_ID_VENDOR(chip->usb_id), - USB_ID_PRODUCT(chip->usb_id)); - } - } - strim(card->shortname); - - /* retrieve the vendor and device strings as longname */ - if (quirk && quirk->vendor_name && *quirk->vendor_name) { - len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname)); - } else { - if (dev->descriptor.iManufacturer) - len = usb_string(dev, dev->descriptor.iManufacturer, - card->longname, sizeof(card->longname)); - else - len = 0; - /* we don't really care if there isn't any vendor string */ - } - if (len > 0) { - strim(card->longname); - if (*card->longname) - strlcat(card->longname, " ", sizeof(card->longname)); - } - - strlcat(card->longname, card->shortname, sizeof(card->longname)); - - len = strlcat(card->longname, " at ", sizeof(card->longname)); - - if (len < sizeof(card->longname)) - usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); - - switch (snd_usb_get_speed(dev)) { - case USB_SPEED_LOW: - strlcat(card->longname, ", low speed", sizeof(card->longname)); - break; - case USB_SPEED_FULL: - strlcat(card->longname, ", full speed", sizeof(card->longname)); - break; - case USB_SPEED_HIGH: - strlcat(card->longname, ", high speed", sizeof(card->longname)); - break; - case USB_SPEED_SUPER: - strlcat(card->longname, ", super speed", sizeof(card->longname)); - break; - case USB_SPEED_SUPER_PLUS: - strlcat(card->longname, ", super speed plus", sizeof(card->longname)); - break; - default: - break; - } + usb_audio_make_shortname(dev, chip, quirk); + usb_audio_make_longname(dev, chip, quirk); snd_usb_audio_create_proc(chip); diff --git a/sound/usb/card.h b/sound/usb/card.h index 1406292d50ec..9b41b7dda84f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -32,6 +32,7 @@ struct audioformat { struct snd_pcm_chmap_elem *chmap; /* (optional) channel map */ bool dsd_dop; /* add DOP headers in case of DSD samples */ bool dsd_bitrev; /* reverse the bits of each DSD sample */ + bool dsd_raw; /* altsetting is raw DSD */ }; struct snd_usb_substream; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 0b030d8fe3fa..c79749613fa6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -443,10 +443,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data)); + if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n", iface, fmt->altsetting, rate, ep); return err; @@ -460,10 +461,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if (chip->sample_rate_read_error > 2) return 0; - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data)); + if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); chip->sample_rate_read_error++; @@ -587,8 +589,15 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); - case UAC_VERSION_2: case UAC_VERSION_3: + if (chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { + if (rate != UAC3_BADD_SAMPLING_RATE) + return -ENXIO; + else + return 0; + } + /* fall through */ + case UAC_VERSION_2: return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } } diff --git a/sound/usb/format.c b/sound/usb/format.c index 49e7ec6d2399..fd13ac11b136 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -64,8 +64,11 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubslotSize; - if (format & UAC2_FORMAT_TYPE_I_RAW_DATA) + if (format & UAC2_FORMAT_TYPE_I_RAW_DATA) { pcm_formats |= SNDRV_PCM_FMTBIT_SPECIAL; + /* flag potentially raw DSD capable altsettings */ + fp->dsd_raw = true; + } format <<= 1; break; @@ -188,7 +191,8 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof */ int r, idx; - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + fp->rate_table = kmalloc_array(nr_rates, sizeof(int), + GFP_KERNEL); if (fp->rate_table == NULL) return -ENOMEM; @@ -362,7 +366,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, goto err_free; } - fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); + fp->rate_table = kmalloc_array(fp->nr_rates, sizeof(int), GFP_KERNEL); if (!fp->rate_table) { ret = -ENOMEM; goto err_free; diff --git a/sound/usb/helper.h b/sound/usb/helper.h index 4463e6d6dcb3..d338bd0e0ca6 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -18,16 +18,12 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) */ -#ifndef get_iface_desc #define get_iface_desc(iface) (&(iface)->desc) #define get_endpoint(alt,ep) (&(alt)->endpoint[ep].desc) #define get_ep_desc(ep) (&(ep)->desc) #define get_cfg_desc(cfg) (&(cfg)->desc) -#endif -#ifndef snd_usb_get_speed #define snd_usb_get_speed(dev) ((dev)->speed) -#endif static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip) { diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 947d6168f24a..d8a14d769f48 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -264,8 +264,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm) struct usb_line6 *line6 = line6pcm->line6; int i; - line6pcm->in.urbs = kzalloc( - sizeof(struct urb *) * line6->iso_buffers, GFP_KERNEL); + line6pcm->in.urbs = kcalloc(line6->iso_buffers, sizeof(struct urb *), + GFP_KERNEL); if (line6pcm->in.urbs == NULL) return -ENOMEM; diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index b3854f8c0c67..72c6f8e82a7e 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -158,8 +158,10 @@ static int line6_buffer_acquire(struct snd_line6_pcm *line6pcm, /* Invoked multiple times in a row so allocate once only */ if (!test_and_set_bit(type, &pstr->opened) && !pstr->buffer) { - pstr->buffer = kmalloc(line6pcm->line6->iso_buffers * - LINE6_ISO_PACKETS * pkt_size, GFP_KERNEL); + pstr->buffer = + kmalloc(array3_size(line6pcm->line6->iso_buffers, + LINE6_ISO_PACKETS, pkt_size), + GFP_KERNEL); if (!pstr->buffer) return -ENOMEM; } diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 819e9b2d1d6e..dec89d2beb57 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -409,8 +409,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm) struct usb_line6 *line6 = line6pcm->line6; int i; - line6pcm->out.urbs = kzalloc( - sizeof(struct urb *) * line6->iso_buffers, GFP_KERNEL); + line6pcm->out.urbs = kcalloc(line6->iso_buffers, sizeof(struct urb *), + GFP_KERNEL); if (line6pcm->out.urbs == NULL) return -ENOMEM; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index bb5ab7a7dfa5..ca963e94ec03 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -112,14 +112,12 @@ enum { #include "mixer_maps.c" static const struct usbmix_name_map * -find_map(struct mixer_build *state, int unitid, int control) +find_map(const struct usbmix_name_map *p, int unitid, int control) { - const struct usbmix_name_map *p = state->map; - if (!p) return NULL; - for (p = state->map; p->id; p++) { + for (; p->id; p++) { if (p->id == unitid && (!control || !p->control || control == p->control)) return p; @@ -201,10 +199,10 @@ static void *find_audio_control_unit(struct mixer_build *state, /* * copy a string with the given id */ -static int snd_usb_copy_string_desc(struct mixer_build *state, +static int snd_usb_copy_string_desc(struct snd_usb_audio *chip, int index, char *buf, int maxlen) { - int len = usb_string(state->chip->dev, index, buf, maxlen - 1); + int len = usb_string(chip->dev, index, buf, maxlen - 1); if (len < 0) return 0; @@ -600,7 +598,8 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, while (snd_ctl_find_id(mixer->chip->card, &kctl->id)) kctl->id.index++; - if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) { + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) { usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", err); return err; @@ -658,14 +657,14 @@ static struct iterm_name_combo { { 0 }, }; -static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm, +static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iterm, unsigned char *name, int maxlen, int term_only) { struct iterm_name_combo *names; int len; if (iterm->name) { - len = snd_usb_copy_string_desc(state, iterm->name, + len = snd_usb_copy_string_desc(chip, iterm->name, name, maxlen); if (len) return len; @@ -719,6 +718,66 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm } /* + * Get logical cluster information for UAC3 devices. + */ +static int get_cluster_channels_v3(struct mixer_build *state, unsigned int cluster_id) +{ + struct uac3_cluster_header_descriptor c_header; + int err; + + err = snd_usb_ctl_msg(state->chip->dev, + usb_rcvctrlpipe(state->chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(state->chip), + &c_header, sizeof(c_header)); + if (err < 0) + goto error; + if (err != sizeof(c_header)) { + err = -EIO; + goto error; + } + + return c_header.bNrChannels; + +error: + usb_audio_err(state->chip, "cannot request logical cluster ID: %d (err: %d)\n", cluster_id, err); + return err; +} + +/* + * Get number of channels for a Mixer Unit. + */ +static int uac_mixer_unit_get_channels(struct mixer_build *state, + struct uac_mixer_unit_descriptor *desc) +{ + int mu_channels; + + if (desc->bLength < 11) + return -EINVAL; + if (!desc->bNrInPins) + return -EINVAL; + + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_3: + mu_channels = get_cluster_channels_v3(state, + uac3_mixer_unit_wClusterDescrID(desc)); + break; + } + + if (!mu_channels) + return -EINVAL; + + return mu_channels; +} + +/* * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ @@ -844,8 +903,12 @@ static int check_input_term(struct mixer_build *state, int id, term->id = id; term->type = le16_to_cpu(d->wTerminalType); - /* REVISIT: UAC3 IT doesn't have channels/cfg */ - term->channels = 0; + err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); + if (err < 0) + return err; + term->channels = err; + + /* REVISIT: UAC3 IT doesn't have channels cfg */ term->chconfig = 0; term->name = le16_to_cpu(d->wTerminalDescrStr); @@ -865,6 +928,18 @@ static int check_input_term(struct mixer_build *state, int id, term->name = le16_to_cpu(d->wClockSourceStr); return 0; } + case UAC3_MIXER_UNIT: { + struct uac_mixer_unit_descriptor *d = p1; + + err = uac_mixer_unit_get_channels(state, d); + if (err < 0) + return err; + + term->channels = err; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + + return 0; + } default: return -ENODEV; } @@ -1258,6 +1333,51 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + struct snd_usb_audio *chip = cval->head.mixer->chip; + int idx = 0, validx, ret, val; + + validx = cval->control << 8 | 0; + + ret = snd_usb_lock_shutdown(chip) ? -EIO : 0; + if (ret) + goto error; + + idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + if (cval->head.mixer->protocol == UAC_VERSION_2) { + struct uac2_connectors_ctl_blk uac2_conn; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, idx, &uac2_conn, sizeof(uac2_conn)); + val = !!uac2_conn.bNrChannels; + } else { /* UAC_VERSION_3 */ + struct uac3_insertion_ctl_blk uac3_conn; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, idx, &uac3_conn, sizeof(uac3_conn)); + val = !!uac3_conn.bmConInserted; + } + + snd_usb_unlock_shutdown(chip); + + if (ret < 0) { +error: + usb_audio_err(chip, + "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", + UAC_GET_CUR, validx, idx, cval->val_type); + return ret; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + static struct snd_kcontrol_new usb_feature_unit_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later manually */ @@ -1288,6 +1408,15 @@ static struct snd_kcontrol_new usb_bool_master_control_ctl_ro = { .put = NULL, }; +static const struct snd_kcontrol_new usb_connector_ctl_ro = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", /* will be filled later manually */ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ctl_boolean_mono_info, + .get = mixer_ctl_connector_get, + .put = NULL, +}; + /* * This symbol is exported in order to allow the mixer quirks to * hook up to the standard feature unit control mechanism @@ -1341,16 +1470,16 @@ static struct usb_feature_control_info *get_feature_control_info(int control) return NULL; } -static void build_feature_ctl(struct mixer_build *state, void *raw_desc, - unsigned int ctl_mask, int control, - struct usb_audio_term *iterm, int unitid, - int readonly_mask) +static void __build_feature_ctl(struct usb_mixer_interface *mixer, + const struct usbmix_name_map *imap, + unsigned int ctl_mask, int control, + struct usb_audio_term *iterm, + struct usb_audio_term *oterm, + int unitid, int nameid, int readonly_mask) { - struct uac_feature_unit_descriptor *desc = raw_desc; struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; - int nameid = uac_feature_unit_iFeature(desc); struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; const struct usbmix_name_map *map; @@ -1361,14 +1490,14 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, return; } - map = find_map(state, unitid, control); + map = find_map(imap, unitid, control); if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return; - snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); + snd_usb_mixer_elem_init_std(&cval->head, mixer, unitid); cval->control = control; cval->cmask = ctl_mask; @@ -1377,7 +1506,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kfree(cval); return; } - if (state->mixer->protocol == UAC_VERSION_1) + if (mixer->protocol == UAC_VERSION_1) cval->val_type = ctl_info->type; else /* UAC_VERSION_2 */ cval->val_type = ctl_info->type_uac2 >= 0 ? @@ -1406,7 +1535,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); if (!kctl) { - usb_audio_err(state->chip, "cannot malloc kcontrol\n"); + usb_audio_err(mixer->chip, "cannot malloc kcontrol\n"); kfree(cval); return; } @@ -1415,7 +1544,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; if (!len && nameid) - len = snd_usb_copy_string_desc(state, nameid, + len = snd_usb_copy_string_desc(mixer->chip, nameid, kctl->id.name, sizeof(kctl->id.name)); switch (control) { @@ -1430,10 +1559,12 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, * - otherwise, anonymous name. */ if (!len) { - len = get_term_name(state, iterm, kctl->id.name, - sizeof(kctl->id.name), 1); - if (!len) - len = get_term_name(state, &state->oterm, + if (iterm) + len = get_term_name(mixer->chip, iterm, + kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len && oterm) + len = get_term_name(mixer->chip, oterm, kctl->id.name, sizeof(kctl->id.name), 1); if (!len) @@ -1442,15 +1573,15 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } if (!mapped_name) - check_no_speaker_on_headset(kctl, state->mixer->chip->card); + check_no_speaker_on_headset(kctl, mixer->chip->card); /* * determine the stream direction: * if the connected output is USB stream, then it's likely a * capture stream. otherwise it should be playback (hopefully :) */ - if (!mapped_name && !(state->oterm.type >> 16)) { - if ((state->oterm.type & 0xff00) == 0x0100) + if (!mapped_name && oterm && !(oterm->type >> 16)) { + if ((oterm->type & 0xff00) == 0x0100) append_ctl_name(kctl, " Capture"); else append_ctl_name(kctl, " Playback"); @@ -1478,7 +1609,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } } - snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl); + snd_usb_mixer_fu_apply_quirk(mixer, cval, unitid, kctl); range = (cval->max - cval->min) / cval->res; /* @@ -1487,26 +1618,46 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, * devices. It will definitively catch all buggy Logitech devices. */ if (range > 384) { - usb_audio_warn(state->chip, + usb_audio_warn(mixer->chip, "Warning! Unlikely big volume range (=%u), cval->res is probably wrong.", range); - usb_audio_warn(state->chip, + usb_audio_warn(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d", cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); } - usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", + usb_audio_dbg(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); snd_usb_mixer_add_control(&cval->head, kctl); } -static void get_connector_control_name(struct mixer_build *state, +static void build_feature_ctl(struct mixer_build *state, void *raw_desc, + unsigned int ctl_mask, int control, + struct usb_audio_term *iterm, int unitid, + int readonly_mask) +{ + struct uac_feature_unit_descriptor *desc = raw_desc; + int nameid = uac_feature_unit_iFeature(desc); + + __build_feature_ctl(state->mixer, state->map, ctl_mask, control, + iterm, &state->oterm, unitid, nameid, readonly_mask); +} + +static void build_feature_ctl_badd(struct usb_mixer_interface *mixer, + unsigned int ctl_mask, int control, int unitid, + const struct usbmix_name_map *badd_map) +{ + __build_feature_ctl(mixer, badd_map, ctl_mask, control, + NULL, NULL, unitid, 0, 0); +} + +static void get_connector_control_name(struct usb_mixer_interface *mixer, struct usb_audio_term *term, bool is_input, char *name, int name_size) { - int name_len = get_term_name(state, term, name, name_size, 0); + int name_len = get_term_name(mixer->chip, term, name, name_size, 0); if (name_len == 0) strlcpy(name, "Unknown", name_size); @@ -1523,7 +1674,7 @@ static void get_connector_control_name(struct mixer_build *state, } /* Build a mixer control for a UAC connector control (jack-detect) */ -static void build_connector_control(struct mixer_build *state, +static void build_connector_control(struct usb_mixer_interface *mixer, struct usb_audio_term *term, bool is_input) { struct snd_kcontrol *kctl; @@ -1532,25 +1683,33 @@ static void build_connector_control(struct mixer_build *state, cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return; - snd_usb_mixer_elem_init_std(&cval->head, state->mixer, term->id); + snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); /* - * The first byte from reading the UAC2_TE_CONNECTOR control returns the - * number of channels connected. This boolean ctl will simply report - * if any channels are connected or not. - * (Audio20_final.pdf Table 5-10: Connector Control CUR Parameter Block) + * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the + * number of channels connected. + * + * UAC3: The first byte specifies size of bitmap for the inserted controls. The + * following byte(s) specifies which connectors are inserted. + * + * This boolean ctl will simply report if any channels are connected + * or not. */ - cval->control = UAC2_TE_CONNECTOR; + if (mixer->protocol == UAC_VERSION_2) + cval->control = UAC2_TE_CONNECTOR; + else /* UAC_VERSION_3 */ + cval->control = UAC3_TE_INSERTION; + cval->val_type = USB_MIXER_BOOLEAN; cval->channels = 1; /* report true if any channel is connected */ cval->min = 0; cval->max = 1; - kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval); + kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval); if (!kctl) { - usb_audio_err(state->chip, "cannot malloc kcontrol\n"); + usb_audio_err(mixer->chip, "cannot malloc kcontrol\n"); kfree(cval); return; } - get_connector_control_name(state, term, is_input, kctl->id.name, + get_connector_control_name(mixer, term, is_input, kctl->id.name, sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); @@ -1605,7 +1764,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, } kctl->private_free = snd_usb_mixer_elem_free; - ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + ret = snd_usb_copy_string_desc(state->chip, hdr->iClockSource, name, sizeof(name)); if (ret > 0) snprintf(kctl->id.name, sizeof(kctl->id.name), @@ -1692,7 +1851,8 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, } /* parse the source unit */ - if ((err = parse_audio_unit(state, hdr->bSourceID)) < 0) + err = parse_audio_unit(state, hdr->bSourceID); + if (err < 0) return err; /* determine the input source type and name */ @@ -1806,16 +1966,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, */ static void build_mixer_unit_ctl(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc, - int in_pin, int in_ch, int unitid, - struct usb_audio_term *iterm) + int in_pin, int in_ch, int num_outs, + int unitid, struct usb_audio_term *iterm) { struct usb_mixer_elem_info *cval; - unsigned int num_outs = uac_mixer_unit_bNrChannels(desc); unsigned int i, len; struct snd_kcontrol *kctl; const struct usbmix_name_map *map; - map = find_map(state, unitid, 0); + map = find_map(state->map, unitid, 0); if (check_ignored_ctl(map)) return; @@ -1848,7 +2007,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (!len) - len = get_term_name(state, iterm, kctl->id.name, + len = get_term_name(state->chip, iterm, kctl->id.name, sizeof(kctl->id.name), 0); if (!len) len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1); @@ -1863,16 +2022,28 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, void *raw_desc) { struct usb_audio_term iterm; - struct uac2_input_terminal_descriptor *d = raw_desc; + unsigned int control, bmctls, term_id; - check_input_term(state, d->bTerminalID, &iterm); if (state->mixer->protocol == UAC_VERSION_2) { - /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(le16_to_cpu(d->bmControls), - UAC2_TE_CONNECTOR)) { - build_connector_control(state, &iterm, true); - } + struct uac2_input_terminal_descriptor *d_v2 = raw_desc; + control = UAC2_TE_CONNECTOR; + term_id = d_v2->bTerminalID; + bmctls = le16_to_cpu(d_v2->bmControls); + } else if (state->mixer->protocol == UAC_VERSION_3) { + struct uac3_input_terminal_descriptor *d_v3 = raw_desc; + control = UAC3_TE_INSERTION; + term_id = d_v3->bTerminalID; + bmctls = le32_to_cpu(d_v3->bmControls); + } else { + return 0; /* UAC1. No Insertion control */ } + + check_input_term(state, term_id, &iterm); + + /* Check for jack detection. */ + if (uac_v2v3_control_is_readable(bmctls, control)) + build_connector_control(state->mixer, &iterm, true); + return 0; } @@ -1887,14 +2058,17 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, int input_pins, num_ins, num_outs; int pin, ich, err; - if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || - !(num_outs = uac_mixer_unit_bNrChannels(desc))) { + err = uac_mixer_unit_get_channels(state, desc); + if (err < 0) { usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", unitid); - return -EINVAL; + return err; } + num_outs = err; + input_pins = desc->bNrInPins; + num_ins = 0; ich = 0; for (pin = 0; pin < input_pins; pin++) { @@ -1921,7 +2095,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, } } if (ich_has_controls) - build_mixer_unit_ctl(state, desc, pin, ich, + build_mixer_unit_ctl(state, desc, pin, ich, num_outs, unitid, &iterm); } } @@ -2098,7 +2272,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, } for (i = 0; i < num_ins; i++) { - if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) + err = parse_audio_unit(state, desc->baSourceID[i]); + if (err < 0) return err; } @@ -2114,7 +2289,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) continue; - map = find_map(state, unitid, valinfo->control); + map = find_map(state->map, unitid, valinfo->control); if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -2162,7 +2337,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); len = 0; if (nameid) - len = snd_usb_copy_string_desc(state, nameid, + len = snd_usb_copy_string_desc(state->chip, + nameid, kctl->id.name, sizeof(kctl->id.name)); if (!len) @@ -2310,14 +2486,15 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, } for (i = 0; i < desc->bNrInPins; i++) { - if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) + err = parse_audio_unit(state, desc->baSourceID[i]); + if (err < 0) return err; } if (desc->bNrInPins == 1) /* only one ? nonsense! */ return 0; - map = find_map(state, unitid, 0); + map = find_map(state->map, unitid, 0); if (check_ignored_ctl(map)) return 0; @@ -2338,7 +2515,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; - namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); + namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL); if (!namelist) { kfree(cval); return -ENOMEM; @@ -2358,7 +2535,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, len = check_mapped_selector_name(state, unitid, i, namelist[i], MAX_ITEM_NAME_LEN); if (! len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0) - len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0); + len = get_term_name(state->chip, &iterm, namelist[i], + MAX_ITEM_NAME_LEN, 0); if (! len) sprintf(namelist[i], "Input %u", i); } @@ -2380,12 +2558,12 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, /* if iSelector is given, use it */ nameid = uac_selector_unit_iSelector(desc); if (nameid) - len = snd_usb_copy_string_desc(state, nameid, + len = snd_usb_copy_string_desc(state->chip, nameid, kctl->id.name, sizeof(kctl->id.name)); /* ... or pick up the terminal name at next */ if (!len) - len = get_term_name(state, &state->oterm, + len = get_term_name(state->chip, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 0); /* ... or use the fixed string "USB" as the last resort */ if (!len) @@ -2458,7 +2636,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } else { /* UAC_VERSION_3 */ switch (p1[2]) { case UAC_INPUT_TERMINAL: - return 0; /* NOP */ + return parse_audio_input_terminal(state, unitid, p1); case UAC3_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); case UAC3_CLOCK_SOURCE: @@ -2503,6 +2681,263 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) return 0; } +/* UAC3 predefined channels configuration */ +struct uac3_badd_profile { + int subclass; + const char *name; + int c_chmask; /* capture channels mask */ + int p_chmask; /* playback channels mask */ + int st_chmask; /* side tone mixing channel mask */ +}; + +static struct uac3_badd_profile uac3_badd_profiles[] = { + { + /* + * BAIF, BAOF or combination of both + * IN: Mono or Stereo cfg, Mono alt possible + * OUT: Mono or Stereo cfg, Mono alt possible + */ + .subclass = UAC3_FUNCTION_SUBCLASS_GENERIC_IO, + .name = "GENERIC IO", + .c_chmask = -1, /* dynamic channels */ + .p_chmask = -1, /* dynamic channels */ + }, + { + /* BAOF; Stereo only cfg, Mono alt possible */ + .subclass = UAC3_FUNCTION_SUBCLASS_HEADPHONE, + .name = "HEADPHONE", + .p_chmask = 3, + }, + { + /* BAOF; Mono or Stereo cfg, Mono alt possible */ + .subclass = UAC3_FUNCTION_SUBCLASS_SPEAKER, + .name = "SPEAKER", + .p_chmask = -1, /* dynamic channels */ + }, + { + /* BAIF; Mono or Stereo cfg, Mono alt possible */ + .subclass = UAC3_FUNCTION_SUBCLASS_MICROPHONE, + .name = "MICROPHONE", + .c_chmask = -1, /* dynamic channels */ + }, + { + /* + * BAIOF topology + * IN: Mono only + * OUT: Mono or Stereo cfg, Mono alt possible + */ + .subclass = UAC3_FUNCTION_SUBCLASS_HEADSET, + .name = "HEADSET", + .c_chmask = 1, + .p_chmask = -1, /* dynamic channels */ + .st_chmask = 1, + }, + { + /* BAIOF; IN: Mono only; OUT: Stereo only, Mono alt possible */ + .subclass = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER, + .name = "HEADSET ADAPTER", + .c_chmask = 1, + .p_chmask = 3, + .st_chmask = 1, + }, + { + /* BAIF + BAOF; IN: Mono only; OUT: Mono only */ + .subclass = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE, + .name = "SPEAKERPHONE", + .c_chmask = 1, + .p_chmask = 1, + }, + { 0 } /* terminator */ +}; + +static bool uac3_badd_func_has_valid_channels(struct usb_mixer_interface *mixer, + struct uac3_badd_profile *f, + int c_chmask, int p_chmask) +{ + /* + * If both playback/capture channels are dynamic, make sure + * at least one channel is present + */ + if (f->c_chmask < 0 && f->p_chmask < 0) { + if (!c_chmask && !p_chmask) { + usb_audio_warn(mixer->chip, "BAAD %s: no channels?", + f->name); + return false; + } + return true; + } + + if ((f->c_chmask < 0 && !c_chmask) || + (f->c_chmask >= 0 && f->c_chmask != c_chmask)) { + usb_audio_warn(mixer->chip, "BAAD %s c_chmask mismatch", + f->name); + return false; + } + if ((f->p_chmask < 0 && !p_chmask) || + (f->p_chmask >= 0 && f->p_chmask != p_chmask)) { + usb_audio_warn(mixer->chip, "BAAD %s p_chmask mismatch", + f->name); + return false; + } + return true; +} + +/* + * create mixer controls for UAC3 BADD profiles + * + * UAC3 BADD device doesn't contain CS descriptors thus we will guess everything + * + * BADD device may contain Mixer Unit, which doesn't have any controls, skip it + */ +static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer, + int ctrlif) +{ + struct usb_device *dev = mixer->chip->dev; + struct usb_interface_assoc_descriptor *assoc; + int badd_profile = mixer->chip->badd_profile; + struct uac3_badd_profile *f; + const struct usbmix_ctl_map *map; + int p_chmask = 0, c_chmask = 0, st_chmask = 0; + int i; + + assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + + /* Detect BADD capture/playback channels from AS EP descriptors */ + for (i = 0; i < assoc->bInterfaceCount; i++) { + int intf = assoc->bFirstInterface + i; + + struct usb_interface *iface; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + unsigned int maxpacksize; + char dir_in; + int chmask, num; + + if (intf == ctrlif) + continue; + + iface = usb_ifnum_to_if(dev, intf); + num = iface->num_altsetting; + + if (num < 2) + return -EINVAL; + + /* + * The number of Channels in an AudioStreaming interface + * and the audio sample bit resolution (16 bits or 24 + * bits) can be derived from the wMaxPacketSize field in + * the Standard AS Audio Data Endpoint descriptor in + * Alternate Setting 1 + */ + alts = &iface->altsetting[1]; + altsd = get_iface_desc(alts); + + if (altsd->bNumEndpoints < 1) + return -EINVAL; + + /* check direction */ + dir_in = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN); + maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + + switch (maxpacksize) { + default: + usb_audio_err(mixer->chip, + "incorrect wMaxPacketSize 0x%x for BADD profile\n", + maxpacksize); + return -EINVAL; + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16: + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24: + chmask = 1; + break; + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16: + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24: + chmask = 3; + break; + } + + if (dir_in) + c_chmask = chmask; + else + p_chmask = chmask; + } + + usb_audio_dbg(mixer->chip, + "UAC3 BADD profile 0x%x: detected c_chmask=%d p_chmask=%d\n", + badd_profile, c_chmask, p_chmask); + + /* check the mapping table */ + for (map = uac3_badd_usbmix_ctl_maps; map->id; map++) { + if (map->id == badd_profile) + break; + } + + if (!map->id) + return -EINVAL; + + for (f = uac3_badd_profiles; f->name; f++) { + if (badd_profile == f->subclass) + break; + } + if (!f->name) + return -EINVAL; + if (!uac3_badd_func_has_valid_channels(mixer, f, c_chmask, p_chmask)) + return -EINVAL; + st_chmask = f->st_chmask; + + /* Playback */ + if (p_chmask) { + /* Master channel, always writable */ + build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE, + UAC3_BADD_FU_ID2, map->map); + /* Mono/Stereo volume channels, always writable */ + build_feature_ctl_badd(mixer, p_chmask, UAC_FU_VOLUME, + UAC3_BADD_FU_ID2, map->map); + } + + /* Capture */ + if (c_chmask) { + /* Master channel, always writable */ + build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE, + UAC3_BADD_FU_ID5, map->map); + /* Mono/Stereo volume channels, always writable */ + build_feature_ctl_badd(mixer, c_chmask, UAC_FU_VOLUME, + UAC3_BADD_FU_ID5, map->map); + } + + /* Side tone-mixing */ + if (st_chmask) { + /* Master channel, always writable */ + build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE, + UAC3_BADD_FU_ID7, map->map); + /* Mono volume channel, always writable */ + build_feature_ctl_badd(mixer, 1, UAC_FU_VOLUME, + UAC3_BADD_FU_ID7, map->map); + } + + /* Insertion Control */ + if (f->subclass == UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER) { + struct usb_audio_term iterm, oterm; + + /* Input Term - Insertion control */ + memset(&iterm, 0, sizeof(iterm)); + iterm.id = UAC3_BADD_IT_ID4; + iterm.type = UAC_BIDIR_TERMINAL_HEADSET; + build_connector_control(mixer, &iterm, true); + + /* Output Term - Insertion control */ + memset(&oterm, 0, sizeof(oterm)); + oterm.id = UAC3_BADD_OT_ID3; + oterm.type = UAC_BIDIR_TERMINAL_HEADSET; + build_connector_control(mixer, &oterm, false); + } + + return 0; +} + /* * create mixer controls * @@ -2572,7 +3007,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { - build_connector_control(&state, &state.oterm, + build_connector_control(state.mixer, &state.oterm, false); } } else { /* UAC_VERSION_3 */ @@ -2596,6 +3031,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; + + if (uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), + UAC3_TE_INSERTION)) { + build_connector_control(state.mixer, &state.oterm, + false); + } } } @@ -2606,9 +3047,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = - (struct usb_mixer_elem_info *)list; + mixer_elem_list_to_info(list); /* invalidate cache, so the value is read from the device */ info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, @@ -2619,7 +3060,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " @@ -2645,8 +3086,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (list = mixer->id_elems[unitid]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { snd_iprintf(buffer, " Unit: %i\n", list->id); if (list->kctl) snd_iprintf(buffer, @@ -2676,19 +3116,19 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) count++; if (count == 0) return; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info; if (!list->kctl) continue; - info = (struct usb_mixer_elem_info *)list; + info = mixer_elem_list_to_info(list); if (count > 1 && info->control != control) continue; @@ -2809,6 +3249,48 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) return 0; } +static int keep_iface_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = mixer->chip->keep_iface; + return 0; +} + +static int keep_iface_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + bool keep_iface = !!ucontrol->value.integer.value[0]; + + if (mixer->chip->keep_iface == keep_iface) + return 0; + mixer->chip->keep_iface = keep_iface; + return 1; +} + +static const struct snd_kcontrol_new keep_iface_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "Keep Interface", + .info = snd_ctl_boolean_mono_info, + .get = keep_iface_ctl_get, + .put = keep_iface_ctl_put, +}; + +static int create_keep_iface_ctl(struct usb_mixer_interface *mixer) +{ + struct snd_kcontrol *kctl = snd_ctl_new1(&keep_iface_ctl, mixer); + + /* need only one control per card */ + if (snd_ctl_find_id(mixer->chip->card, &kctl->id)) { + snd_ctl_free_one(kctl); + return 0; + } + + return snd_ctl_add(mixer->chip->card, kctl); +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { @@ -2847,8 +3329,23 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, break; } - if ((err = snd_usb_mixer_controls(mixer)) < 0 || - (err = snd_usb_mixer_status_create(mixer)) < 0) + if (mixer->protocol == UAC_VERSION_3 && + chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { + err = snd_usb_mixer_controls_badd(mixer, ctrlif); + if (err < 0) + goto _error; + } else { + err = snd_usb_mixer_controls(mixer); + if (err < 0) + goto _error; + } + + err = snd_usb_mixer_status_create(mixer); + if (err < 0) + goto _error; + + err = create_keep_iface_ctl(mixer); + if (err < 0) goto _error; snd_usb_mixer_apply_create_quirk(mixer); @@ -2909,7 +3406,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) static int restore_mixer_value(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; if (cval->cmask) { @@ -2945,8 +3442,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (list = mixer->id_elems[id]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, id) { if (list->resume) { err = list->resume(list); if (err < 0) @@ -2956,6 +3452,8 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) } } + snd_usb_mixer_resume_quirk(mixer); + return snd_usb_mixer_activate(mixer); } #endif diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ba27f7ade670..e02653465e29 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -53,6 +53,12 @@ struct usb_mixer_elem_list { usb_mixer_elem_resume_func_t resume; }; +/* iterate over mixer element list of the given unit id */ +#define for_each_mixer_elem(list, mixer, id) \ + for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem) +#define mixer_elem_list_to_info(list) \ + container_of(list, struct usb_mixer_elem_info, head) + struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index eaa03acd4686..71069e110897 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -485,3 +485,68 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { { 0 } /* terminator */ }; +/* + * Control map entries for UAC3 BADD profiles + */ + +static struct usbmix_name_map uac3_badd_generic_io_map[] = { + { UAC3_BADD_FU_ID2, "Generic Out Playback" }, + { UAC3_BADD_FU_ID5, "Generic In Capture" }, + { 0 } /* terminator */ +}; +static struct usbmix_name_map uac3_badd_headphone_map[] = { + { UAC3_BADD_FU_ID2, "Headphone Playback" }, + { 0 } /* terminator */ +}; +static struct usbmix_name_map uac3_badd_speaker_map[] = { + { UAC3_BADD_FU_ID2, "Speaker Playback" }, + { 0 } /* terminator */ +}; +static struct usbmix_name_map uac3_badd_microphone_map[] = { + { UAC3_BADD_FU_ID5, "Mic Capture" }, + { 0 } /* terminator */ +}; +/* Covers also 'headset adapter' profile */ +static struct usbmix_name_map uac3_badd_headset_map[] = { + { UAC3_BADD_FU_ID2, "Headset Playback" }, + { UAC3_BADD_FU_ID5, "Headset Capture" }, + { UAC3_BADD_FU_ID7, "Sidetone Mixing" }, + { 0 } /* terminator */ +}; +static struct usbmix_name_map uac3_badd_speakerphone_map[] = { + { UAC3_BADD_FU_ID2, "Speaker Playback" }, + { UAC3_BADD_FU_ID5, "Mic Capture" }, + { 0 } /* terminator */ +}; + +static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = { + { + .id = UAC3_FUNCTION_SUBCLASS_GENERIC_IO, + .map = uac3_badd_generic_io_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_HEADPHONE, + .map = uac3_badd_headphone_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_SPEAKER, + .map = uac3_badd_speaker_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_MICROPHONE, + .map = uac3_badd_microphone_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_HEADSET, + .map = uac3_badd_headset_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER, + .map = uac3_badd_headset_map, + }, + { + .id = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE, + .map = uac3_badd_speakerphone_map, + }, + { 0 } /* terminator */ +}; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 56537a156580..e82a72fea9a1 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1172,7 +1172,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, @@ -1799,12 +1799,33 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) NULL); } +static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id) +{ + u16 buf = 0; + + snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + ch, snd_usb_ctrl_intf(chip) | (id << 8), + &buf, 2); +} + +static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) +{ + /* fix to 0dB playback volumes */ + dell_dock_init_vol(mixer->chip, 1, 16); + dell_dock_init_vol(mixer->chip, 2, 16); + dell_dock_init_vol(mixer->chip, 1, 19); + dell_dock_init_vol(mixer->chip, 2, 19); + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; struct snd_info_entry *entry; - if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0) + err = snd_usb_soundblaster_remote_init(mixer); + if (err < 0) return err; switch (mixer->chip->usb_id) { @@ -1828,8 +1849,6 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) /* EMU0204 */ case USB_ID(0x041e, 0x3f19): err = snd_emu0204_controls_create(mixer); - if (err < 0) - break; break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ @@ -1884,11 +1903,25 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; + case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ + err = dell_dock_mixer_init(mixer); + break; } return err; } +#ifdef CONFIG_PM +void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) +{ + switch (mixer->chip->usb_id) { + case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ + dell_dock_mixer_init(mixer); + break; + } +} +#endif + void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid) { diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index b5abd328a361..52be26db558f 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,5 +14,9 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); +#ifdef CONFIG_PM +void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); +#endif + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index c33e2378089d..4aeb9488a0c9 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); int i; for (i = 0; i < elem->channels; i++) @@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); if (elem->cached) snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3cbfae6604f9..160f52c4871b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -76,10 +76,9 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_usb_substream *subs; + struct snd_usb_substream *subs = substream->runtime->private_data; unsigned int hwptr_done; - subs = (struct snd_usb_substream *)substream->runtime->private_data; if (atomic_read(&subs->stream->chip->shutdown)) return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); @@ -164,10 +163,11 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, + data, sizeof(data)); + if (err < 0) { usb_audio_err(chip, "%d:%d: cannot set enable PITCH\n", iface, ep); return err; @@ -185,10 +185,11 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, int err; data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC2_EP_CS_PITCH << 8, 0, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC2_EP_CS_PITCH << 8, 0, + data, sizeof(data)); + if (err < 0) { usb_audio_err(chip, "%d:%d: cannot set enable PITCH (v2)\n", iface, fmt->altsetting); return err; @@ -321,6 +322,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_host_interface *alts; struct usb_interface *iface; unsigned int ep; + unsigned int ifnum; /* Implicit feedback sync EPs consumers are always playback EPs */ if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) @@ -330,44 +332,27 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - break; + ifnum = 3; + goto add_sync_ep_from_ifnum; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - case USB_ID(0x2466, 0x8003): + ifnum = 2; + goto add_sync_ep_from_ifnum; + case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ ep = 0x86; - iface = usb_ifnum_to_if(dev, 2); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - case USB_ID(0x1397, 0x0002): + ifnum = 2; + goto add_sync_ep_from_ifnum; + case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx III */ ep = 0x81; - iface = usb_ifnum_to_if(dev, 1); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - + ifnum = 2; + goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ + ep = 0x81; + ifnum = 1; + goto add_sync_ep_from_ifnum; } + if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && altsd->bInterfaceProtocol == 2 && @@ -382,6 +367,14 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, /* No quirk */ return 0; +add_sync_ep_from_ifnum: + iface = usb_ifnum_to_if(dev, ifnum); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + add_sync_ep: subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, ep, !subs->direction, @@ -507,7 +500,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) return -EINVAL; - alts = &iface->altsetting[fmt->altset_idx]; + alts = usb_altnum_to_altsetting(iface, fmt->altsetting); altsd = get_iface_desc(alts); if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) return -EINVAL; @@ -517,21 +510,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* close the old interface */ if (subs->interface >= 0 && subs->interface != fmt->iface) { - err = usb_set_interface(subs->dev, subs->interface, 0); - if (err < 0) { - dev_err(&dev->dev, - "%d:%d: return to setting 0 failed (%d)\n", - fmt->iface, fmt->altsetting, err); - return -EIO; + if (!subs->stream->chip->keep_iface) { + err = usb_set_interface(subs->dev, subs->interface, 0); + if (err < 0) { + dev_err(&dev->dev, + "%d:%d: return to setting 0 failed (%d)\n", + fmt->iface, fmt->altsetting, err); + return -EIO; + } } subs->interface = -1; subs->altset_idx = 0; } /* set interface */ - if (subs->interface != fmt->iface || - subs->altset_idx != fmt->altset_idx) { - + if (iface->cur_altsetting != alts) { err = snd_usb_select_mode_quirk(subs, fmt); if (err < 0) return -EIO; @@ -545,12 +538,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } dev_dbg(&dev->dev, "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); - subs->interface = fmt->iface; - subs->altset_idx = fmt->altset_idx; - snd_usb_set_interface_quirk(dev); } + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, fmt->endpoint, subs->direction, SND_USB_ENDPOINT_TYPE_DATA); @@ -736,7 +728,11 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + if (snd_usb_use_vmalloc) + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + else + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -789,7 +785,11 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) snd_usb_endpoint_deactivate(subs->data_endpoint); snd_usb_unlock_shutdown(subs->stream->chip); } - return snd_pcm_lib_free_vmalloc_buffer(substream); + + if (snd_usb_use_vmalloc) + return snd_pcm_lib_free_vmalloc_buffer(substream); + else + return snd_pcm_lib_free_pages(substream); } /* @@ -1123,7 +1123,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, return 0; subs->rate_list.list = rate_list = - kmalloc(sizeof(int) * count, GFP_KERNEL); + kmalloc_array(count, sizeof(int), GFP_KERNEL); if (!subs->rate_list.list) return -ENOMEM; subs->rate_list.count = count; @@ -1181,9 +1181,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre pt = 125 * (1 << fp->datainterval); ptmin = min(ptmin, pt); } - err = snd_usb_autoresume(subs->stream->chip); - if (err < 0) - return err; param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; if (subs->speed == USB_SPEED_FULL) @@ -1192,30 +1189,37 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (ptmin == 1000) /* if period time doesn't go below 1 ms, no rules needed */ param_period_time_if_needed = -1; - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - ptmin, UINT_MAX); - - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; + + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + ptmin, UINT_MAX); + if (err < 0) + return err; + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, + -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1); + if (err < 0) + return err; if (param_period_time_if_needed >= 0) { err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, @@ -1225,19 +1229,18 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) - goto rep_err; + return err; } - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) - goto rep_err; - return 0; + err = snd_usb_pcm_check_knot(runtime, subs); + if (err < 0) + return err; -rep_err: - snd_usb_autosuspend(subs->stream->chip); - return err; + return snd_usb_autoresume(subs->stream->chip); } -static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_open(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; @@ -1257,14 +1260,16 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) return setup_hw_info(runtime, subs); } -static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_close(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); - if (subs->interface >= 0 && + if (!as->chip->keep_iface && + subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; @@ -1311,7 +1316,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, if (bytes % (runtime->sample_bits >> 3) != 0) { int oldbytes = bytes; bytes = frames * stride; - dev_warn(&subs->dev->dev, + dev_warn_ratelimited(&subs->dev->dev, "Corrected urb data len. %d->%d\n", oldbytes, bytes); } @@ -1619,26 +1624,6 @@ static void retire_playback_urb(struct snd_usb_substream *subs, spin_unlock_irqrestore(&subs->lock, flags); } -static int snd_usb_playback_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_playback_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_capture_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static int snd_usb_capture_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); -} - static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -1700,8 +1685,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream } static const struct snd_pcm_ops snd_usb_playback_ops = { - .open = snd_usb_playback_open, - .close = snd_usb_playback_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1713,8 +1698,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { }; static const struct snd_pcm_ops snd_usb_capture_ops = { - .open = snd_usb_capture_open, - .close = snd_usb_capture_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1725,9 +1710,50 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .mmap = snd_pcm_lib_mmap_vmalloc, }; +static const struct snd_pcm_ops snd_usb_playback_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_playback_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +static const struct snd_pcm_ops snd_usb_capture_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_capture_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { - snd_pcm_set_ops(pcm, stream, - stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_ops : &snd_usb_capture_ops); + const struct snd_pcm_ops *ops; + + if (snd_usb_use_vmalloc) + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_ops : &snd_usb_capture_ops; + else + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_dev_ops : &snd_usb_capture_dev_ops; + snd_pcm_set_ops(pcm, stream, ops); +} + +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) +{ + struct snd_pcm *pcm = subs->stream->pcm; + struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; + struct device *dev = subs->dev->bus->controller; + + if (!snd_usb_use_vmalloc) + snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, + dev, 64*1024, 512*1024); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 35740d5ef268..f77ec58bf1a1 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -10,6 +10,7 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt); +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); #endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 754e632a27bd..8aac48f9c322 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3277,6 +3277,10 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* disabled due to regression for other devices; + * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 + */ +#if 0 { /* * Nura's first gen headphones use Cambridge Silicon Radio's vendor @@ -3324,6 +3328,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +#endif /* disabled */ { /* @@ -3371,5 +3376,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +/* Dell WD15 Dock */ +{ + USB_DEVICE(0x0bda, 0x4014), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD15 Dock", + .profile_name = "Dell-WD15-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acbeb52f6fd6..02b6cc02767f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -851,6 +851,36 @@ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) return 0; /* Successful boot */ } +static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) +{ + int err; + + dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + + /* If the Axe-Fx III has not fully booted, it will timeout when trying + * to enable the audio streaming interface. A more generous timeout is + * used here to detect when the Axe-Fx III has finished booting as the + * set interface message will be acked once it has + */ + err = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + USB_REQ_SET_INTERFACE, USB_RECIP_INTERFACE, + 1, 1, NULL, 0, 120000); + if (err < 0) { + dev_err(&dev->dev, + "failed waiting for Axe-Fx III to boot: %d\n", err); + return err; + } + + dev_dbg(&dev->dev, "Axe-Fx III is now ready\n"); + + err = usb_set_interface(dev, 1, 0); + if (err < 0) + dev_dbg(&dev->dev, + "error stopping Axe-Fx III interface: %d\n", err); + + return 0; +} + /* * Setup quirks */ @@ -1026,6 +1056,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_fasttrackpro_boot_quirk(dev); case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */ return snd_usb_gamecon780_boot_quirk(dev); + case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ + return snd_usb_axefx3_boot_quirk(dev); } return 0; @@ -1327,20 +1359,42 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ + case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ + case USB_ID(0x20b1, 0x0002): /* Wyred 4 Sound DAC-2 DSD */ + case USB_ID(0x20b1, 0x2004): /* Matrix Audio X-SPDIF 2 */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ + case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */ + case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ + case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */ + case USB_ID(0x20b1, 0x2005): /* Denafrips Ares DAC */ case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ + case USB_ID(0x20b1, 0x3021): /* Eastern El. MiniMax Tube DAC Supreme */ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */ + case USB_ID(0x20b1, 0x302d): /* Unison Research Unico CD Due */ + case USB_ID(0x20b1, 0x307b): /* CH Precision C1 DAC */ + case USB_ID(0x20b1, 0x3086): /* Singxer F-1 converter board */ + case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */ + case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ + case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ + case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ + case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ + case USB_ID(0x6b42, 0x0042): /* MSB Technology */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; @@ -1384,6 +1438,20 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; } + /* Mostly generic method to detect many DSD-capable implementations - + * from XMOS/Thesycon + */ + switch (USB_ID_VENDOR(chip->usb_id)) { + case 0x20b1: /* XMOS based devices */ + case 0x25ce: /* Mytek devices */ + if (fp->dsd_raw) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; + default: + break; + + } + return 0; } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 5ed334575fc7..729afd808cc4 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -106,6 +106,8 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->ep_num = fp->endpoint; if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + + snd_usb_preallocate_buffer(subs); } /* kctl callbacks for usb-audio channel maps */ @@ -633,6 +635,395 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, return NULL; } +static struct audioformat * +audio_format_alloc_init(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int protocol, int iface_no, int altset_idx, + int altno, int num_channels, int clock) +{ + struct audioformat *fp; + + fp = kzalloc(sizeof(*fp), GFP_KERNEL); + if (!fp) + return NULL; + + fp->iface = iface_no; + fp->altsetting = altno; + fp->altset_idx = altset_idx; + fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; + fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->protocol = protocol; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + fp->channels = num_channels; + if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH) + fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) + * (fp->maxpacksize & 0x7ff); + fp->clock = clock; + INIT_LIST_HEAD(&fp->list); + + return fp; +} + +static struct audioformat * +snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int protocol, int iface_no, int altset_idx, + int altno, int stream, int bm_quirk) +{ + struct usb_device *dev = chip->dev; + struct uac_format_type_i_continuous_descriptor *fmt; + unsigned int num_channels = 0, chconfig = 0; + struct audioformat *fp; + int clock = 0; + u64 format; + + /* get audio formats */ + if (protocol == UAC_VERSION_1) { + struct uac1_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_AS_GENERAL); + struct uac_input_terminal_descriptor *iterm; + + if (!as) { + dev_err(&dev->dev, + "%u:%d : UAC_AS_GENERAL descriptor not found\n", + iface_no, altno); + return NULL; + } + + if (as->bLength < sizeof(*as)) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_AS_GENERAL desc\n", + iface_no, altno); + return NULL; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + + iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (iterm) { + num_channels = iterm->bNrChannels; + chconfig = le16_to_cpu(iterm->wChannelConfig); + } + } else { /* UAC_VERSION_2 */ + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; + struct uac2_as_header_descriptor *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_AS_GENERAL); + + if (!as) { + dev_err(&dev->dev, + "%u:%d : UAC_AS_GENERAL descriptor not found\n", + iface_no, altno); + return NULL; + } + + if (as->bLength < sizeof(*as)) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_AS_GENERAL desc\n", + iface_no, altno); + return NULL; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + chconfig = le32_to_cpu(as->bmChannelConfig); + + /* + * lookup the terminal associated to this interface + * to extract the clock + */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + if (!chconfig && (num_channels == input_term->bNrChannels)) + chconfig = le32_to_cpu(input_term->bmChannelConfig); + goto found_clock; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + goto found_clock; + } + + dev_err(&dev->dev, + "%u:%d : bogus bTerminalLink %d\n", + iface_no, altno, as->bTerminalLink); + return NULL; + } + +found_clock: + /* get format type */ + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) { + dev_err(&dev->dev, + "%u:%d : no UAC_FORMAT_TYPE desc\n", + iface_no, altno); + return NULL; + } + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) + || ((protocol == UAC_VERSION_2) && + (fmt->bLength < 6))) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_FORMAT_TYPE desc\n", + iface_no, altno); + return NULL; + } + + /* + * Blue Microphones workaround: The last altsetting is + * identical with the previous one, except for a larger + * packet size, but is actually a mislabeled two-channel + * setting; ignore it. + * + * Part 2: analyze quirk flag and format + */ + if (bm_quirk && fmt->bNrChannels == 1 && fmt->bSubframeSize == 2) + return NULL; + + fp = audio_format_alloc_init(chip, alts, protocol, iface_no, + altset_idx, altno, num_channels, clock); + if (!fp) + return ERR_PTR(-ENOMEM); + + fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, + iface_no); + + /* some quirks for attributes here */ + snd_usb_audioformat_attributes_quirk(chip, fp, stream); + + /* ok, let's parse further... */ + if (snd_usb_parse_audio_format(chip, fp, format, + fmt, stream) < 0) { + kfree(fp->rate_table); + kfree(fp); + return NULL; + } + + /* Create chmap */ + if (fp->channels != num_channels) + chconfig = 0; + + fp->chmap = convert_chmap(fp->channels, chconfig, protocol); + + return fp; +} + +static struct audioformat * +snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int iface_no, int altset_idx, + int altno, int stream) +{ + struct usb_device *dev = chip->dev; + struct uac3_input_terminal_descriptor *input_term; + struct uac3_output_terminal_descriptor *output_term; + struct uac3_cluster_header_descriptor *cluster; + struct uac3_as_header_descriptor *as = NULL; + struct uac3_hc_descriptor_header hc_header; + struct snd_pcm_chmap_elem *chmap; + unsigned char badd_profile; + u64 badd_formats = 0; + unsigned int num_channels; + struct audioformat *fp; + u16 cluster_id, wLength; + int clock = 0; + int err; + + badd_profile = chip->badd_profile; + + if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { + unsigned int maxpacksize = + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + + switch (maxpacksize) { + default: + dev_err(&dev->dev, + "%u:%d : incorrect wMaxPacketSize for BADD profile\n", + iface_no, altno); + return NULL; + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16: + badd_formats = SNDRV_PCM_FMTBIT_S16_LE; + num_channels = 1; + break; + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24: + badd_formats = SNDRV_PCM_FMTBIT_S24_3LE; + num_channels = 1; + break; + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16: + badd_formats = SNDRV_PCM_FMTBIT_S16_LE; + num_channels = 2; + break; + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24: + badd_formats = SNDRV_PCM_FMTBIT_S24_3LE; + num_channels = 2; + break; + } + + chmap = kzalloc(sizeof(*chmap), GFP_KERNEL); + if (!chmap) + return ERR_PTR(-ENOMEM); + + if (num_channels == 1) { + chmap->map[0] = SNDRV_CHMAP_MONO; + } else { + chmap->map[0] = SNDRV_CHMAP_FL; + chmap->map[1] = SNDRV_CHMAP_FR; + } + + chmap->channels = num_channels; + clock = UAC3_BADD_CS_ID9; + goto found_clock; + } + + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_AS_GENERAL); + if (!as) { + dev_err(&dev->dev, + "%u:%d : UAC_AS_GENERAL descriptor not found\n", + iface_no, altno); + return NULL; + } + + if (as->bLength < sizeof(*as)) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_AS_GENERAL desc\n", + iface_no, altno); + return NULL; + } + + cluster_id = le16_to_cpu(as->wClusterDescrID); + if (!cluster_id) { + dev_err(&dev->dev, + "%u:%d : no cluster descriptor\n", + iface_no, altno); + return NULL; + } + + /* + * Get number of channels and channel map through + * High Capability Cluster Descriptor + * + * First step: get High Capability header and + * read size of Cluster Descriptor + */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(chip), + &hc_header, sizeof(hc_header)); + if (err < 0) + return ERR_PTR(err); + else if (err != sizeof(hc_header)) { + dev_err(&dev->dev, + "%u:%d : can't get High Capability descriptor\n", + iface_no, altno); + return ERR_PTR(-EIO); + } + + /* + * Second step: allocate needed amount of memory + * and request Cluster Descriptor + */ + wLength = le16_to_cpu(hc_header.wLength); + cluster = kzalloc(wLength, GFP_KERNEL); + if (!cluster) + return ERR_PTR(-ENOMEM); + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(chip), + cluster, wLength); + if (err < 0) { + kfree(cluster); + return ERR_PTR(err); + } else if (err != wLength) { + dev_err(&dev->dev, + "%u:%d : can't get Cluster Descriptor\n", + iface_no, altno); + kfree(cluster); + return ERR_PTR(-EIO); + } + + num_channels = cluster->bNrChannels; + chmap = convert_chmap_v3(cluster); + kfree(cluster); + + /* + * lookup the terminal associated to this interface + * to extract the clock + */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + goto found_clock; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + goto found_clock; + } + + dev_err(&dev->dev, "%u:%d : bogus bTerminalLink %d\n", + iface_no, altno, as->bTerminalLink); + kfree(chmap); + return NULL; + +found_clock: + fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no, + altset_idx, altno, num_channels, clock); + if (!fp) { + kfree(chmap); + return ERR_PTR(-ENOMEM); + } + + fp->chmap = chmap; + + if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { + fp->attributes = 0; /* No attributes */ + + fp->fmt_type = UAC_FORMAT_TYPE_I; + fp->formats = badd_formats; + + fp->nr_rates = 0; /* SNDRV_PCM_RATE_CONTINUOUS */ + fp->rate_min = UAC3_BADD_SAMPLING_RATE; + fp->rate_max = UAC3_BADD_SAMPLING_RATE; + fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + + } else { + fp->attributes = parse_uac_endpoint_attributes(chip, alts, + UAC_VERSION_3, + iface_no); + /* ok, let's parse further... */ + if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { + kfree(fp->chmap); + kfree(fp->rate_table); + kfree(fp); + return NULL; + } + } + + return fp; +} + int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -640,13 +1031,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - u64 format = 0; - unsigned int num_channels = 0; struct audioformat *fp = NULL; - int num, protocol, clock = 0; - struct uac_format_type_i_continuous_descriptor *fmt = NULL; - struct snd_pcm_chmap_elem *chmap_v3 = NULL; - unsigned int chconfig; + int num, protocol; dev = chip->dev; @@ -695,303 +1081,48 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) protocol <= 2) protocol = UAC_VERSION_1; - chconfig = 0; - /* get audio formats */ switch (protocol) { default: dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n", iface_no, altno, protocol); protocol = UAC_VERSION_1; /* fall through */ - - case UAC_VERSION_1: { - struct uac1_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - struct uac_input_terminal_descriptor *iterm; - - if (!as) { - dev_err(&dev->dev, - "%u:%d : UAC_AS_GENERAL descriptor not found\n", - iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - dev_err(&dev->dev, - "%u:%d : invalid UAC_AS_GENERAL desc\n", - iface_no, altno); - continue; - } - - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ - - iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (iterm) { - num_channels = iterm->bNrChannels; - chconfig = le16_to_cpu(iterm->wChannelConfig); - } - - break; - } - + case UAC_VERSION_1: + /* fall through */ case UAC_VERSION_2: { - struct uac2_input_terminal_descriptor *input_term; - struct uac2_output_terminal_descriptor *output_term; - struct uac2_as_header_descriptor *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - dev_err(&dev->dev, - "%u:%d : UAC_AS_GENERAL descriptor not found\n", - iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - dev_err(&dev->dev, - "%u:%d : invalid UAC_AS_GENERAL desc\n", - iface_no, altno); - continue; - } - - num_channels = as->bNrChannels; - format = le32_to_cpu(as->bmFormats); - chconfig = le32_to_cpu(as->bmChannelConfig); - - /* lookup the terminal associated to this interface - * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (input_term) { - clock = input_term->bCSourceID; - if (!chconfig && (num_channels == input_term->bNrChannels)) - chconfig = le32_to_cpu(input_term->bmChannelConfig); - break; - } - - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (output_term) { - clock = output_term->bCSourceID; - break; - } - - dev_err(&dev->dev, - "%u:%d : bogus bTerminalLink %d\n", - iface_no, altno, as->bTerminalLink); - continue; - } - - case UAC_VERSION_3: { - struct uac3_input_terminal_descriptor *input_term; - struct uac3_output_terminal_descriptor *output_term; - struct uac3_as_header_descriptor *as; - struct uac3_cluster_header_descriptor *cluster; - struct uac3_hc_descriptor_header hc_header; - u16 cluster_id, wLength; - - as = snd_usb_find_csint_desc(alts->extra, - alts->extralen, - NULL, UAC_AS_GENERAL); - - if (!as) { - dev_err(&dev->dev, - "%u:%d : UAC_AS_GENERAL descriptor not found\n", - iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - dev_err(&dev->dev, - "%u:%d : invalid UAC_AS_GENERAL desc\n", - iface_no, altno); - continue; - } - - cluster_id = le16_to_cpu(as->wClusterDescrID); - if (!cluster_id) { - dev_err(&dev->dev, - "%u:%d : no cluster descriptor\n", - iface_no, altno); - continue; - } - - /* - * Get number of channels and channel map through - * High Capability Cluster Descriptor - * - * First step: get High Capability header and - * read size of Cluster Descriptor - */ - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), - UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - cluster_id, - snd_usb_ctrl_intf(chip), - &hc_header, sizeof(hc_header)); - if (err < 0) - return err; - else if (err != sizeof(hc_header)) { - dev_err(&dev->dev, - "%u:%d : can't get High Capability descriptor\n", - iface_no, altno); - return -EIO; - } - - /* - * Second step: allocate needed amount of memory - * and request Cluster Descriptor - */ - wLength = le16_to_cpu(hc_header.wLength); - cluster = kzalloc(wLength, GFP_KERNEL); - if (!cluster) - return -ENOMEM; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), - UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - cluster_id, - snd_usb_ctrl_intf(chip), - cluster, wLength); - if (err < 0) { - kfree(cluster); - return err; - } else if (err != wLength) { - dev_err(&dev->dev, - "%u:%d : can't get Cluster Descriptor\n", - iface_no, altno); - kfree(cluster); - return -EIO; - } - - num_channels = cluster->bNrChannels; - chmap_v3 = convert_chmap_v3(cluster); - - kfree(cluster); - - format = le64_to_cpu(as->bmFormats); - - /* lookup the terminal associated to this interface - * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor( - chip->ctrl_intf, - as->bTerminalLink); - - if (input_term) { - clock = input_term->bCSourceID; - break; - } - - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); - if (output_term) { - clock = output_term->bCSourceID; - break; - } - - dev_err(&dev->dev, - "%u:%d : bogus bTerminalLink %d\n", - iface_no, altno, as->bTerminalLink); - continue; - } - } - - if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { - /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, - alts->extralen, - NULL, UAC_FORMAT_TYPE); - if (!fmt) { - dev_err(&dev->dev, - "%u:%d : no UAC_FORMAT_TYPE desc\n", - iface_no, altno); - continue; - } - if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) - || ((protocol == UAC_VERSION_2) && - (fmt->bLength < 6))) { - dev_err(&dev->dev, - "%u:%d : invalid UAC_FORMAT_TYPE desc\n", - iface_no, altno); - continue; - } + int bm_quirk = 0; /* * Blue Microphones workaround: The last altsetting is * identical with the previous one, except for a larger * packet size, but is actually a mislabeled two-channel * setting; ignore it. + * + * Part 1: prepare quirk flag */ - if (fmt->bNrChannels == 1 && - fmt->bSubframeSize == 2 && - altno == 2 && num == 3 && + if (altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) - continue; - } - - fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; + bm_quirk = 1; - fp->iface = iface_no; - fp->altsetting = altno; - fp->altset_idx = i; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = snd_usb_parse_datainterval(chip, alts); - fp->protocol = protocol; - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->channels = num_channels; - if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) - fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) - * (fp->maxpacksize & 0x7ff); - fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); - fp->clock = clock; - INIT_LIST_HEAD(&fp->list); - - /* some quirks for attributes here */ - snd_usb_audioformat_attributes_quirk(chip, fp, stream); - - /* ok, let's parse further... */ - if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { - if (snd_usb_parse_audio_format(chip, fp, format, - fmt, stream) < 0) { - kfree(fp->rate_table); - kfree(fp); - fp = NULL; - continue; - } - } else { - struct uac3_as_header_descriptor *as; - - as = snd_usb_find_csint_desc(alts->extra, - alts->extralen, - NULL, UAC_AS_GENERAL); - - if (snd_usb_parse_audio_format_v3(chip, fp, as, - stream) < 0) { - kfree(fp->rate_table); - kfree(fp); - fp = NULL; - continue; - } + fp = snd_usb_get_audioformat_uac12(chip, alts, protocol, + iface_no, i, altno, + stream, bm_quirk); + break; + } + case UAC_VERSION_3: + fp = snd_usb_get_audioformat_uac3(chip, alts, + iface_no, i, altno, stream); + break; } - /* Create chmap */ - if (fp->channels != num_channels) - chconfig = 0; - - if (protocol == UAC_VERSION_3) - fp->chmap = chmap_v3; - else - fp->chmap = convert_chmap(fp->channels, chconfig, - protocol); + if (!fp) + continue; + else if (IS_ERR(fp)) + return PTR_ERR(fp); dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); err = snd_usb_add_audio_stream(chip, stream, fp); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4d5c89a7ba2b..b9faeca645fd 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -49,6 +49,8 @@ struct snd_usb_audio { int num_suspended_intf; int sample_rate_read_error; + int badd_profile; /* UAC3 BADD profile */ + struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ int pcm_devs; @@ -59,6 +61,9 @@ struct snd_usb_audio { int setup; /* from the 'device_setup' module param */ bool autoclock; /* from the 'autoclock' module param */ + bool keep_iface; /* keep interface/altset after closing + * or parameter change + */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; @@ -109,6 +114,7 @@ enum quirk_type { struct snd_usb_audio_quirk { const char *vendor_name; const char *product_name; + const char *profile_name; /* override the card->longname */ int16_t ifnum; uint16_t type; const void *data; @@ -121,4 +127,6 @@ struct snd_usb_audio_quirk { int snd_usb_lock_shutdown(struct snd_usb_audio *chip); void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); +extern bool snd_usb_use_vmalloc; + #endif /* __USBAUDIO_H */ diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 0ddf29267d70..da4a5a541512 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -266,7 +266,9 @@ int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y) int err = 0, i; - if (NULL == (usX2Y->AS04.buffer = kmalloc(URB_DataLen_AsyncSeq*URBS_AsyncSeq, GFP_KERNEL))) { + usX2Y->AS04.buffer = kmalloc_array(URBS_AsyncSeq, + URB_DataLen_AsyncSeq, GFP_KERNEL); + if (NULL == usX2Y->AS04.buffer) { err = -ENOMEM; } else for (i = 0; i < URBS_AsyncSeq; ++i) { diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 345e439aa95b..2b833054e3b0 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -436,7 +436,9 @@ static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs) } if (!is_playback && !(*purb)->transfer_buffer) { /* allocate a capture buffer per urb */ - (*purb)->transfer_buffer = kmalloc(subs->maxpacksize * nr_of_packs(), GFP_KERNEL); + (*purb)->transfer_buffer = + kmalloc_array(subs->maxpacksize, + nr_of_packs(), GFP_KERNEL); if (NULL == (*purb)->transfer_buffer) { usX2Y_urbs_release(subs); return -ENOMEM; @@ -662,7 +664,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) err = -ENOMEM; goto cleanup; } - usbdata = kmalloc(sizeof(int) * NOOF_SETRATE_URBS, GFP_KERNEL); + usbdata = kmalloc_array(NOOF_SETRATE_URBS, sizeof(int), + GFP_KERNEL); if (NULL == usbdata) { err = -ENOMEM; goto cleanup; diff --git a/sound/xen/Kconfig b/sound/xen/Kconfig new file mode 100644 index 000000000000..4f1fceea82d2 --- /dev/null +++ b/sound/xen/Kconfig @@ -0,0 +1,10 @@ +# ALSA Xen drivers + +config SND_XEN_FRONTEND + tristate "Xen para-virtualized sound frontend driver" + depends on XEN + select SND_PCM + select XEN_XENBUS_FRONTEND + help + Choose this option if you want to enable a para-virtualized + frontend sound driver for Xen guest OSes. diff --git a/sound/xen/Makefile b/sound/xen/Makefile new file mode 100644 index 000000000000..1e6470ecc2f2 --- /dev/null +++ b/sound/xen/Makefile @@ -0,0 +1,9 @@ +# SPDX-License-Identifier: GPL-2.0 OR MIT + +snd_xen_front-objs := xen_snd_front.o \ + xen_snd_front_cfg.o \ + xen_snd_front_evtchnl.o \ + xen_snd_front_shbuf.o \ + xen_snd_front_alsa.o + +obj-$(CONFIG_SND_XEN_FRONTEND) += snd_xen_front.o diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c new file mode 100644 index 000000000000..b089b13b5160 --- /dev/null +++ b/sound/xen/xen_snd_front.c @@ -0,0 +1,397 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/delay.h> +#include <linux/module.h> + +#include <xen/page.h> +#include <xen/platform_pci.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include <xen/interface/io/sndif.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_evtchnl.h" +#include "xen_snd_front_shbuf.h" + +static struct xensnd_req * +be_stream_prepare_req(struct xen_snd_front_evtchnl *evtchnl, u8 operation) +{ + struct xensnd_req *req; + + req = RING_GET_REQUEST(&evtchnl->u.req.ring, + evtchnl->u.req.ring.req_prod_pvt); + req->operation = operation; + req->id = evtchnl->evt_next_id++; + evtchnl->evt_id = req->id; + return req; +} + +static int be_stream_do_io(struct xen_snd_front_evtchnl *evtchnl) +{ + if (unlikely(evtchnl->state != EVTCHNL_STATE_CONNECTED)) + return -EIO; + + reinit_completion(&evtchnl->u.req.completion); + xen_snd_front_evtchnl_flush(evtchnl); + return 0; +} + +static int be_stream_wait_io(struct xen_snd_front_evtchnl *evtchnl) +{ + if (wait_for_completion_timeout(&evtchnl->u.req.completion, + msecs_to_jiffies(VSND_WAIT_BACK_MS)) <= 0) + return -ETIMEDOUT; + + return evtchnl->u.req.resp_status; +} + +int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, + struct xensnd_query_hw_param *hw_param_req, + struct xensnd_query_hw_param *hw_param_resp) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_HW_PARAM_QUERY); + req->op.hw_param = *hw_param_req; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + if (ret == 0) + *hw_param_resp = evtchnl->u.req.resp.hw_param; + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, + struct xen_snd_front_shbuf *sh_buf, + u8 format, unsigned int channels, + unsigned int rate, u32 buffer_sz, + u32 period_sz) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_OPEN); + req->op.open.pcm_format = format; + req->op.open.pcm_channels = channels; + req->op.open.pcm_rate = rate; + req->op.open.buffer_sz = buffer_sz; + req->op.open.period_sz = period_sz; + req->op.open.gref_directory = xen_snd_front_shbuf_get_dir_start(sh_buf); + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_CLOSE); + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_write(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_WRITE); + req->op.rw.length = count; + req->op.rw.offset = pos; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_read(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_READ); + req->op.rw.length = count; + req->op.rw.offset = pos; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_trigger(struct xen_snd_front_evtchnl *evtchnl, + int type) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_TRIGGER); + req->op.trigger.type = type; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +static void xen_snd_drv_fini(struct xen_snd_front_info *front_info) +{ + xen_snd_front_alsa_fini(front_info); + xen_snd_front_evtchnl_free_all(front_info); +} + +static int sndback_initwait(struct xen_snd_front_info *front_info) +{ + int num_streams; + int ret; + + ret = xen_snd_front_cfg_card(front_info, &num_streams); + if (ret < 0) + return ret; + + /* create event channels for all streams and publish */ + ret = xen_snd_front_evtchnl_create_all(front_info, num_streams); + if (ret < 0) + return ret; + + return xen_snd_front_evtchnl_publish_all(front_info); +} + +static int sndback_connect(struct xen_snd_front_info *front_info) +{ + return xen_snd_front_alsa_init(front_info); +} + +static void sndback_disconnect(struct xen_snd_front_info *front_info) +{ + xen_snd_drv_fini(front_info); + xenbus_switch_state(front_info->xb_dev, XenbusStateInitialising); +} + +static void sndback_changed(struct xenbus_device *xb_dev, + enum xenbus_state backend_state) +{ + struct xen_snd_front_info *front_info = dev_get_drvdata(&xb_dev->dev); + int ret; + + dev_dbg(&xb_dev->dev, "Backend state is %s, front is %s\n", + xenbus_strstate(backend_state), + xenbus_strstate(xb_dev->state)); + + switch (backend_state) { + case XenbusStateReconfiguring: + /* fall through */ + case XenbusStateReconfigured: + /* fall through */ + case XenbusStateInitialised: + /* fall through */ + break; + + case XenbusStateInitialising: + /* Recovering after backend unexpected closure. */ + sndback_disconnect(front_info); + break; + + case XenbusStateInitWait: + /* Recovering after backend unexpected closure. */ + sndback_disconnect(front_info); + + ret = sndback_initwait(front_info); + if (ret < 0) + xenbus_dev_fatal(xb_dev, ret, "initializing frontend"); + else + xenbus_switch_state(xb_dev, XenbusStateInitialised); + break; + + case XenbusStateConnected: + if (xb_dev->state != XenbusStateInitialised) + break; + + ret = sndback_connect(front_info); + if (ret < 0) + xenbus_dev_fatal(xb_dev, ret, "initializing frontend"); + else + xenbus_switch_state(xb_dev, XenbusStateConnected); + break; + + case XenbusStateClosing: + /* + * In this state backend starts freeing resources, + * so let it go into closed state first, so we can also + * remove ours. + */ + break; + + case XenbusStateUnknown: + /* fall through */ + case XenbusStateClosed: + if (xb_dev->state == XenbusStateClosed) + break; + + sndback_disconnect(front_info); + break; + } +} + +static int xen_drv_probe(struct xenbus_device *xb_dev, + const struct xenbus_device_id *id) +{ + struct xen_snd_front_info *front_info; + + front_info = devm_kzalloc(&xb_dev->dev, + sizeof(*front_info), GFP_KERNEL); + if (!front_info) + return -ENOMEM; + + front_info->xb_dev = xb_dev; + dev_set_drvdata(&xb_dev->dev, front_info); + + return xenbus_switch_state(xb_dev, XenbusStateInitialising); +} + +static int xen_drv_remove(struct xenbus_device *dev) +{ + struct xen_snd_front_info *front_info = dev_get_drvdata(&dev->dev); + int to = 100; + + xenbus_switch_state(dev, XenbusStateClosing); + + /* + * On driver removal it is disconnected from XenBus, + * so no backend state change events come via .otherend_changed + * callback. This prevents us from exiting gracefully, e.g. + * signaling the backend to free event channels, waiting for its + * state to change to XenbusStateClosed and cleaning at our end. + * Normally when front driver removed backend will finally go into + * XenbusStateInitWait state. + * + * Workaround: read backend's state manually and wait with time-out. + */ + while ((xenbus_read_unsigned(front_info->xb_dev->otherend, "state", + XenbusStateUnknown) != XenbusStateInitWait) && + --to) + msleep(10); + + if (!to) { + unsigned int state; + + state = xenbus_read_unsigned(front_info->xb_dev->otherend, + "state", XenbusStateUnknown); + pr_err("Backend state is %s while removing driver\n", + xenbus_strstate(state)); + } + + xen_snd_drv_fini(front_info); + xenbus_frontend_closed(dev); + return 0; +} + +static const struct xenbus_device_id xen_drv_ids[] = { + { XENSND_DRIVER_NAME }, + { "" } +}; + +static struct xenbus_driver xen_driver = { + .ids = xen_drv_ids, + .probe = xen_drv_probe, + .remove = xen_drv_remove, + .otherend_changed = sndback_changed, +}; + +static int __init xen_drv_init(void) +{ + if (!xen_domain()) + return -ENODEV; + + if (!xen_has_pv_devices()) + return -ENODEV; + + /* At the moment we only support case with XEN_PAGE_SIZE == PAGE_SIZE */ + if (XEN_PAGE_SIZE != PAGE_SIZE) { + pr_err(XENSND_DRIVER_NAME ": different kernel and Xen page sizes are not supported: XEN_PAGE_SIZE (%lu) != PAGE_SIZE (%lu)\n", + XEN_PAGE_SIZE, PAGE_SIZE); + return -ENODEV; + } + + pr_info("Initialising Xen " XENSND_DRIVER_NAME " frontend driver\n"); + return xenbus_register_frontend(&xen_driver); +} + +static void __exit xen_drv_fini(void) +{ + pr_info("Unregistering Xen " XENSND_DRIVER_NAME " frontend driver\n"); + xenbus_unregister_driver(&xen_driver); +} + +module_init(xen_drv_init); +module_exit(xen_drv_fini); + +MODULE_DESCRIPTION("Xen virtual sound device frontend"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("xen:" XENSND_DRIVER_NAME); +MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual soundcard}}"); diff --git a/sound/xen/xen_snd_front.h b/sound/xen/xen_snd_front.h new file mode 100644 index 000000000000..a2ea2463bcc5 --- /dev/null +++ b/sound/xen/xen_snd_front.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_H +#define __XEN_SND_FRONT_H + +#include "xen_snd_front_cfg.h" + +struct xen_snd_front_card_info; +struct xen_snd_front_evtchnl; +struct xen_snd_front_evtchnl_pair; +struct xen_snd_front_shbuf; +struct xensnd_query_hw_param; + +struct xen_snd_front_info { + struct xenbus_device *xb_dev; + + struct xen_snd_front_card_info *card_info; + + int num_evt_pairs; + struct xen_snd_front_evtchnl_pair *evt_pairs; + + struct xen_front_cfg_card cfg; +}; + +int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, + struct xensnd_query_hw_param *hw_param_req, + struct xensnd_query_hw_param *hw_param_resp); + +int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, + struct xen_snd_front_shbuf *sh_buf, + u8 format, unsigned int channels, + unsigned int rate, u32 buffer_sz, + u32 period_sz); + +int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl); + +int xen_snd_front_stream_write(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count); + +int xen_snd_front_stream_read(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count); + +int xen_snd_front_stream_trigger(struct xen_snd_front_evtchnl *evtchnl, + int type); + +#endif /* __XEN_SND_FRONT_H */ diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c new file mode 100644 index 000000000000..5a2bd70a2fa1 --- /dev/null +++ b/sound/xen/xen_snd_front_alsa.c @@ -0,0 +1,822 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include <xen/xenbus.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_cfg.h" +#include "xen_snd_front_evtchnl.h" +#include "xen_snd_front_shbuf.h" + +struct xen_snd_front_pcm_stream_info { + struct xen_snd_front_info *front_info; + struct xen_snd_front_evtchnl_pair *evt_pair; + struct xen_snd_front_shbuf sh_buf; + int index; + + bool is_open; + struct snd_pcm_hardware pcm_hw; + + /* Number of processed frames as reported by the backend. */ + snd_pcm_uframes_t be_cur_frame; + /* Current HW pointer to be reported via .period callback. */ + atomic_t hw_ptr; + /* Modulo of the number of processed frames - for period detection. */ + u32 out_frames; +}; + +struct xen_snd_front_pcm_instance_info { + struct xen_snd_front_card_info *card_info; + struct snd_pcm *pcm; + struct snd_pcm_hardware pcm_hw; + int num_pcm_streams_pb; + struct xen_snd_front_pcm_stream_info *streams_pb; + int num_pcm_streams_cap; + struct xen_snd_front_pcm_stream_info *streams_cap; +}; + +struct xen_snd_front_card_info { + struct xen_snd_front_info *front_info; + struct snd_card *card; + struct snd_pcm_hardware pcm_hw; + int num_pcm_instances; + struct xen_snd_front_pcm_instance_info *pcm_instances; +}; + +struct alsa_sndif_sample_format { + u8 sndif; + snd_pcm_format_t alsa; +}; + +struct alsa_sndif_hw_param { + u8 sndif; + snd_pcm_hw_param_t alsa; +}; + +static const struct alsa_sndif_sample_format ALSA_SNDIF_FORMATS[] = { + { + .sndif = XENSND_PCM_FORMAT_U8, + .alsa = SNDRV_PCM_FORMAT_U8 + }, + { + .sndif = XENSND_PCM_FORMAT_S8, + .alsa = SNDRV_PCM_FORMAT_S8 + }, + { + .sndif = XENSND_PCM_FORMAT_U16_LE, + .alsa = SNDRV_PCM_FORMAT_U16_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U16_BE, + .alsa = SNDRV_PCM_FORMAT_U16_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S16_LE, + .alsa = SNDRV_PCM_FORMAT_S16_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S16_BE, + .alsa = SNDRV_PCM_FORMAT_S16_BE + }, + { + .sndif = XENSND_PCM_FORMAT_U24_LE, + .alsa = SNDRV_PCM_FORMAT_U24_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U24_BE, + .alsa = SNDRV_PCM_FORMAT_U24_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S24_LE, + .alsa = SNDRV_PCM_FORMAT_S24_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S24_BE, + .alsa = SNDRV_PCM_FORMAT_S24_BE + }, + { + .sndif = XENSND_PCM_FORMAT_U32_LE, + .alsa = SNDRV_PCM_FORMAT_U32_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U32_BE, + .alsa = SNDRV_PCM_FORMAT_U32_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S32_LE, + .alsa = SNDRV_PCM_FORMAT_S32_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S32_BE, + .alsa = SNDRV_PCM_FORMAT_S32_BE + }, + { + .sndif = XENSND_PCM_FORMAT_A_LAW, + .alsa = SNDRV_PCM_FORMAT_A_LAW + }, + { + .sndif = XENSND_PCM_FORMAT_MU_LAW, + .alsa = SNDRV_PCM_FORMAT_MU_LAW + }, + { + .sndif = XENSND_PCM_FORMAT_F32_LE, + .alsa = SNDRV_PCM_FORMAT_FLOAT_LE + }, + { + .sndif = XENSND_PCM_FORMAT_F32_BE, + .alsa = SNDRV_PCM_FORMAT_FLOAT_BE + }, + { + .sndif = XENSND_PCM_FORMAT_F64_LE, + .alsa = SNDRV_PCM_FORMAT_FLOAT64_LE + }, + { + .sndif = XENSND_PCM_FORMAT_F64_BE, + .alsa = SNDRV_PCM_FORMAT_FLOAT64_BE + }, + { + .sndif = XENSND_PCM_FORMAT_IEC958_SUBFRAME_LE, + .alsa = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE + }, + { + .sndif = XENSND_PCM_FORMAT_IEC958_SUBFRAME_BE, + .alsa = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE + }, + { + .sndif = XENSND_PCM_FORMAT_IMA_ADPCM, + .alsa = SNDRV_PCM_FORMAT_IMA_ADPCM + }, + { + .sndif = XENSND_PCM_FORMAT_MPEG, + .alsa = SNDRV_PCM_FORMAT_MPEG + }, + { + .sndif = XENSND_PCM_FORMAT_GSM, + .alsa = SNDRV_PCM_FORMAT_GSM + }, +}; + +static int to_sndif_format(snd_pcm_format_t format) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (ALSA_SNDIF_FORMATS[i].alsa == format) + return ALSA_SNDIF_FORMATS[i].sndif; + + return -EINVAL; +} + +static u64 to_sndif_formats_mask(u64 alsa_formats) +{ + u64 mask; + int i; + + mask = 0; + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) + mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; + + return mask; +} + +static u64 to_alsa_formats_mask(u64 sndif_formats) +{ + u64 mask; + int i; + + mask = 0; + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) + mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; + + return mask; +} + +static void stream_clear(struct xen_snd_front_pcm_stream_info *stream) +{ + stream->is_open = false; + stream->be_cur_frame = 0; + stream->out_frames = 0; + atomic_set(&stream->hw_ptr, 0); + xen_snd_front_evtchnl_pair_clear(stream->evt_pair); + xen_snd_front_shbuf_clear(&stream->sh_buf); +} + +static void stream_free(struct xen_snd_front_pcm_stream_info *stream) +{ + xen_snd_front_shbuf_free(&stream->sh_buf); + stream_clear(stream); +} + +static struct xen_snd_front_pcm_stream_info * +stream_get(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_instance_info *pcm_instance = + snd_pcm_substream_chip(substream); + struct xen_snd_front_pcm_stream_info *stream; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream = &pcm_instance->streams_pb[substream->number]; + else + stream = &pcm_instance->streams_cap[substream->number]; + + return stream; +} + +static int alsa_hw_rule(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct xen_snd_front_pcm_stream_info *stream = rule->private; + struct device *dev = &stream->front_info->xb_dev->dev; + struct snd_mask *formats = + hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *rates = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *period = + hw_param_interval(params, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + struct snd_interval *buffer = + hw_param_interval(params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + struct xensnd_query_hw_param req; + struct xensnd_query_hw_param resp; + struct snd_interval interval; + struct snd_mask mask; + u64 sndif_formats; + int changed, ret; + + /* Collect all the values we need for the query. */ + + req.formats = to_sndif_formats_mask((u64)formats->bits[0] | + (u64)(formats->bits[1]) << 32); + + req.rates.min = rates->min; + req.rates.max = rates->max; + + req.channels.min = channels->min; + req.channels.max = channels->max; + + req.buffer.min = buffer->min; + req.buffer.max = buffer->max; + + req.period.min = period->min; + req.period.max = period->max; + + ret = xen_snd_front_stream_query_hw_param(&stream->evt_pair->req, + &req, &resp); + if (ret < 0) { + /* Check if this is due to backend communication error. */ + if (ret == -EIO || ret == -ETIMEDOUT) + dev_err(dev, "Failed to query ALSA HW parameters\n"); + return ret; + } + + /* Refine HW parameters after the query. */ + changed = 0; + + sndif_formats = to_alsa_formats_mask(resp.formats); + snd_mask_none(&mask); + mask.bits[0] = (u32)sndif_formats; + mask.bits[1] = (u32)(sndif_formats >> 32); + ret = snd_mask_refine(formats, &mask); + if (ret < 0) + return ret; + changed |= ret; + + interval.openmin = 0; + interval.openmax = 0; + interval.integer = 1; + + interval.min = resp.rates.min; + interval.max = resp.rates.max; + ret = snd_interval_refine(rates, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.channels.min; + interval.max = resp.channels.max; + ret = snd_interval_refine(channels, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.buffer.min; + interval.max = resp.buffer.max; + ret = snd_interval_refine(buffer, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.period.min; + interval.max = resp.period.max; + ret = snd_interval_refine(period, &interval); + if (ret < 0) + return ret; + changed |= ret; + + return changed; +} + +static int alsa_open(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_instance_info *pcm_instance = + snd_pcm_substream_chip(substream); + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct xen_snd_front_info *front_info = + pcm_instance->card_info->front_info; + struct device *dev = &front_info->xb_dev->dev; + int ret; + + /* + * Return our HW properties: override defaults with those configured + * via XenStore. + */ + runtime->hw = stream->pcm_hw; + runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_NONINTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE); + runtime->hw.info |= SNDRV_PCM_INFO_INTERLEAVED; + + stream->evt_pair = &front_info->evt_pairs[stream->index]; + + stream->front_info = front_info; + + stream->evt_pair->evt.u.evt.substream = substream; + + stream_clear(stream); + + xen_snd_front_evtchnl_pair_set_connected(stream->evt_pair, true); + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_FORMAT\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_RATE\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_CHANNELS\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_PERIOD_SIZE\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_BUFFER_SIZE\n"); + return ret; + } + + return 0; +} + +static int alsa_close(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + xen_snd_front_evtchnl_pair_set_connected(stream->evt_pair, false); + return 0; +} + +static int alsa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + /* + * This callback may be called multiple times, + * so free the previously allocated shared buffer if any. + */ + stream_free(stream); + + ret = xen_snd_front_shbuf_alloc(stream->front_info->xb_dev, + &stream->sh_buf, + params_buffer_bytes(params)); + if (ret < 0) { + stream_free(stream); + dev_err(&stream->front_info->xb_dev->dev, + "Failed to allocate buffers for stream with index %d\n", + stream->index); + return ret; + } + + return 0; +} + +static int alsa_hw_free(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + ret = xen_snd_front_stream_close(&stream->evt_pair->req); + stream_free(stream); + return ret; +} + +static int alsa_prepare(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (!stream->is_open) { + struct snd_pcm_runtime *runtime = substream->runtime; + u8 sndif_format; + int ret; + + ret = to_sndif_format(runtime->format); + if (ret < 0) { + dev_err(&stream->front_info->xb_dev->dev, + "Unsupported sample format: %d\n", + runtime->format); + return ret; + } + sndif_format = ret; + + ret = xen_snd_front_stream_prepare(&stream->evt_pair->req, + &stream->sh_buf, + sndif_format, + runtime->channels, + runtime->rate, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream)); + if (ret < 0) + return ret; + + stream->is_open = true; + } + + return 0; +} + +static int alsa_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int type; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + type = XENSND_OP_TRIGGER_START; + break; + + case SNDRV_PCM_TRIGGER_RESUME: + type = XENSND_OP_TRIGGER_RESUME; + break; + + case SNDRV_PCM_TRIGGER_STOP: + type = XENSND_OP_TRIGGER_STOP; + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + type = XENSND_OP_TRIGGER_PAUSE; + break; + + default: + return -EINVAL; + } + + return xen_snd_front_stream_trigger(&stream->evt_pair->req, type); +} + +void xen_snd_front_alsa_handle_cur_pos(struct xen_snd_front_evtchnl *evtchnl, + u64 pos_bytes) +{ + struct snd_pcm_substream *substream = evtchnl->u.evt.substream; + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + snd_pcm_uframes_t delta, new_hw_ptr, cur_frame; + + cur_frame = bytes_to_frames(substream->runtime, pos_bytes); + + delta = cur_frame - stream->be_cur_frame; + stream->be_cur_frame = cur_frame; + + new_hw_ptr = (snd_pcm_uframes_t)atomic_read(&stream->hw_ptr); + new_hw_ptr = (new_hw_ptr + delta) % substream->runtime->buffer_size; + atomic_set(&stream->hw_ptr, (int)new_hw_ptr); + + stream->out_frames += delta; + if (stream->out_frames > substream->runtime->period_size) { + stream->out_frames %= substream->runtime->period_size; + snd_pcm_period_elapsed(substream); + } +} + +static snd_pcm_uframes_t alsa_pointer(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + return (snd_pcm_uframes_t)atomic_read(&stream->hw_ptr); +} + +static int alsa_pb_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void __user *src, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + if (copy_from_user(stream->sh_buf.buffer + pos, src, count)) + return -EFAULT; + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +static int alsa_pb_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void *src, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + memcpy(stream->sh_buf.buffer + pos, src, count); + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +static int alsa_cap_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void __user *dst, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); + if (ret < 0) + return ret; + + return copy_to_user(dst, stream->sh_buf.buffer + pos, count) ? + -EFAULT : 0; +} + +static int alsa_cap_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void *dst, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); + if (ret < 0) + return ret; + + memcpy(dst, stream->sh_buf.buffer + pos, count); + + return 0; +} + +static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + memset(stream->sh_buf.buffer + pos, 0, count); + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +/* + * FIXME: The mmaped data transfer is asynchronous and there is no + * ack signal from user-space when it is done. This is the + * reason it is not implemented in the PV driver as we do need + * to know when the buffer can be transferred to the backend. + */ + +static struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, +}; + +static struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, +}; + +static int new_pcm_instance(struct xen_snd_front_card_info *card_info, + struct xen_front_cfg_pcm_instance *instance_cfg, + struct xen_snd_front_pcm_instance_info *pcm_instance_info) +{ + struct snd_pcm *pcm; + int ret, i; + + dev_dbg(&card_info->front_info->xb_dev->dev, + "New PCM device \"%s\" with id %d playback %d capture %d", + instance_cfg->name, + instance_cfg->device_id, + instance_cfg->num_streams_pb, + instance_cfg->num_streams_cap); + + pcm_instance_info->card_info = card_info; + + pcm_instance_info->pcm_hw = instance_cfg->pcm_hw; + + if (instance_cfg->num_streams_pb) { + pcm_instance_info->streams_pb = + devm_kcalloc(&card_info->card->card_dev, + instance_cfg->num_streams_pb, + sizeof(struct xen_snd_front_pcm_stream_info), + GFP_KERNEL); + if (!pcm_instance_info->streams_pb) + return -ENOMEM; + } + + if (instance_cfg->num_streams_cap) { + pcm_instance_info->streams_cap = + devm_kcalloc(&card_info->card->card_dev, + instance_cfg->num_streams_cap, + sizeof(struct xen_snd_front_pcm_stream_info), + GFP_KERNEL); + if (!pcm_instance_info->streams_cap) + return -ENOMEM; + } + + pcm_instance_info->num_pcm_streams_pb = + instance_cfg->num_streams_pb; + pcm_instance_info->num_pcm_streams_cap = + instance_cfg->num_streams_cap; + + for (i = 0; i < pcm_instance_info->num_pcm_streams_pb; i++) { + pcm_instance_info->streams_pb[i].pcm_hw = + instance_cfg->streams_pb[i].pcm_hw; + pcm_instance_info->streams_pb[i].index = + instance_cfg->streams_pb[i].index; + } + + for (i = 0; i < pcm_instance_info->num_pcm_streams_cap; i++) { + pcm_instance_info->streams_cap[i].pcm_hw = + instance_cfg->streams_cap[i].pcm_hw; + pcm_instance_info->streams_cap[i].index = + instance_cfg->streams_cap[i].index; + } + + ret = snd_pcm_new(card_info->card, instance_cfg->name, + instance_cfg->device_id, + instance_cfg->num_streams_pb, + instance_cfg->num_streams_cap, + &pcm); + if (ret < 0) + return ret; + + pcm->private_data = pcm_instance_info; + pcm->info_flags = 0; + /* we want to handle all PCM operations in non-atomic context */ + pcm->nonatomic = true; + strncpy(pcm->name, "Virtual card PCM", sizeof(pcm->name)); + + if (instance_cfg->num_streams_pb) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_drv_alsa_playback_ops); + + if (instance_cfg->num_streams_cap) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_drv_alsa_capture_ops); + + pcm_instance_info->pcm = pcm; + return 0; +} + +int xen_snd_front_alsa_init(struct xen_snd_front_info *front_info) +{ + struct device *dev = &front_info->xb_dev->dev; + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct xen_snd_front_card_info *card_info; + struct snd_card *card; + int ret, i; + + dev_dbg(dev, "Creating virtual sound card\n"); + + ret = snd_card_new(dev, 0, XENSND_DRIVER_NAME, THIS_MODULE, + sizeof(struct xen_snd_front_card_info), &card); + if (ret < 0) + return ret; + + card_info = card->private_data; + card_info->front_info = front_info; + front_info->card_info = card_info; + card_info->card = card; + card_info->pcm_instances = + devm_kcalloc(dev, cfg->num_pcm_instances, + sizeof(struct xen_snd_front_pcm_instance_info), + GFP_KERNEL); + if (!card_info->pcm_instances) { + ret = -ENOMEM; + goto fail; + } + + card_info->num_pcm_instances = cfg->num_pcm_instances; + card_info->pcm_hw = cfg->pcm_hw; + + for (i = 0; i < cfg->num_pcm_instances; i++) { + ret = new_pcm_instance(card_info, &cfg->pcm_instances[i], + &card_info->pcm_instances[i]); + if (ret < 0) + goto fail; + } + + strncpy(card->driver, XENSND_DRIVER_NAME, sizeof(card->driver)); + strncpy(card->shortname, cfg->name_short, sizeof(card->shortname)); + strncpy(card->longname, cfg->name_long, sizeof(card->longname)); + + ret = snd_card_register(card); + if (ret < 0) + goto fail; + + return 0; + +fail: + snd_card_free(card); + return ret; +} + +void xen_snd_front_alsa_fini(struct xen_snd_front_info *front_info) +{ + struct xen_snd_front_card_info *card_info; + struct snd_card *card; + + card_info = front_info->card_info; + if (!card_info) + return; + + card = card_info->card; + if (!card) + return; + + dev_dbg(&front_info->xb_dev->dev, "Removing virtual sound card %d\n", + card->number); + snd_card_free(card); + + /* Card_info will be freed when destroying front_info->xb_dev->dev. */ + card_info->card = NULL; +} diff --git a/sound/xen/xen_snd_front_alsa.h b/sound/xen/xen_snd_front_alsa.h new file mode 100644 index 000000000000..18abd9eec967 --- /dev/null +++ b/sound/xen/xen_snd_front_alsa.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_ALSA_H +#define __XEN_SND_FRONT_ALSA_H + +struct xen_snd_front_info; + +int xen_snd_front_alsa_init(struct xen_snd_front_info *front_info); + +void xen_snd_front_alsa_fini(struct xen_snd_front_info *front_info); + +void xen_snd_front_alsa_handle_cur_pos(struct xen_snd_front_evtchnl *evtchnl, + u64 pos_bytes); + +#endif /* __XEN_SND_FRONT_ALSA_H */ diff --git a/sound/xen/xen_snd_front_cfg.c b/sound/xen/xen_snd_front_cfg.c new file mode 100644 index 000000000000..eda077c8087a --- /dev/null +++ b/sound/xen/xen_snd_front_cfg.c @@ -0,0 +1,519 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <xen/xenbus.h> + +#include <xen/interface/io/sndif.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_cfg.h" + +/* Maximum number of supported streams. */ +#define VSND_MAX_STREAM 8 + +struct cfg_hw_sample_rate { + const char *name; + unsigned int mask; + unsigned int value; +}; + +static const struct cfg_hw_sample_rate CFG_HW_SUPPORTED_RATES[] = { + { .name = "5512", .mask = SNDRV_PCM_RATE_5512, .value = 5512 }, + { .name = "8000", .mask = SNDRV_PCM_RATE_8000, .value = 8000 }, + { .name = "11025", .mask = SNDRV_PCM_RATE_11025, .value = 11025 }, + { .name = "16000", .mask = SNDRV_PCM_RATE_16000, .value = 16000 }, + { .name = "22050", .mask = SNDRV_PCM_RATE_22050, .value = 22050 }, + { .name = "32000", .mask = SNDRV_PCM_RATE_32000, .value = 32000 }, + { .name = "44100", .mask = SNDRV_PCM_RATE_44100, .value = 44100 }, + { .name = "48000", .mask = SNDRV_PCM_RATE_48000, .value = 48000 }, + { .name = "64000", .mask = SNDRV_PCM_RATE_64000, .value = 64000 }, + { .name = "96000", .mask = SNDRV_PCM_RATE_96000, .value = 96000 }, + { .name = "176400", .mask = SNDRV_PCM_RATE_176400, .value = 176400 }, + { .name = "192000", .mask = SNDRV_PCM_RATE_192000, .value = 192000 }, +}; + +struct cfg_hw_sample_format { + const char *name; + u64 mask; +}; + +static const struct cfg_hw_sample_format CFG_HW_SUPPORTED_FORMATS[] = { + { + .name = XENSND_PCM_FORMAT_U8_STR, + .mask = SNDRV_PCM_FMTBIT_U8 + }, + { + .name = XENSND_PCM_FORMAT_S8_STR, + .mask = SNDRV_PCM_FMTBIT_S8 + }, + { + .name = XENSND_PCM_FORMAT_U16_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U16_LE + }, + { + .name = XENSND_PCM_FORMAT_U16_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U16_BE + }, + { + .name = XENSND_PCM_FORMAT_S16_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S16_LE + }, + { + .name = XENSND_PCM_FORMAT_S16_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S16_BE + }, + { + .name = XENSND_PCM_FORMAT_U24_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U24_LE + }, + { + .name = XENSND_PCM_FORMAT_U24_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U24_BE + }, + { + .name = XENSND_PCM_FORMAT_S24_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S24_LE + }, + { + .name = XENSND_PCM_FORMAT_S24_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S24_BE + }, + { + .name = XENSND_PCM_FORMAT_U32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U32_LE + }, + { + .name = XENSND_PCM_FORMAT_U32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U32_BE + }, + { + .name = XENSND_PCM_FORMAT_S32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S32_LE + }, + { + .name = XENSND_PCM_FORMAT_S32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S32_BE + }, + { + .name = XENSND_PCM_FORMAT_A_LAW_STR, + .mask = SNDRV_PCM_FMTBIT_A_LAW + }, + { + .name = XENSND_PCM_FORMAT_MU_LAW_STR, + .mask = SNDRV_PCM_FMTBIT_MU_LAW + }, + { + .name = XENSND_PCM_FORMAT_F32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT_LE + }, + { + .name = XENSND_PCM_FORMAT_F32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT_BE + }, + { + .name = XENSND_PCM_FORMAT_F64_LE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT64_LE + }, + { + .name = XENSND_PCM_FORMAT_F64_BE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT64_BE + }, + { + .name = XENSND_PCM_FORMAT_IEC958_SUBFRAME_LE_STR, + .mask = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE + }, + { + .name = XENSND_PCM_FORMAT_IEC958_SUBFRAME_BE_STR, + .mask = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE + }, + { + .name = XENSND_PCM_FORMAT_IMA_ADPCM_STR, + .mask = SNDRV_PCM_FMTBIT_IMA_ADPCM + }, + { + .name = XENSND_PCM_FORMAT_MPEG_STR, + .mask = SNDRV_PCM_FMTBIT_MPEG + }, + { + .name = XENSND_PCM_FORMAT_GSM_STR, + .mask = SNDRV_PCM_FMTBIT_GSM + }, +}; + +static void cfg_hw_rates(char *list, unsigned int len, + const char *path, struct snd_pcm_hardware *pcm_hw) +{ + char *cur_rate; + unsigned int cur_mask; + unsigned int cur_value; + unsigned int rates; + unsigned int rate_min; + unsigned int rate_max; + int i; + + rates = 0; + rate_min = -1; + rate_max = 0; + while ((cur_rate = strsep(&list, XENSND_LIST_SEPARATOR))) { + for (i = 0; i < ARRAY_SIZE(CFG_HW_SUPPORTED_RATES); i++) + if (!strncasecmp(cur_rate, + CFG_HW_SUPPORTED_RATES[i].name, + XENSND_SAMPLE_RATE_MAX_LEN)) { + cur_mask = CFG_HW_SUPPORTED_RATES[i].mask; + cur_value = CFG_HW_SUPPORTED_RATES[i].value; + rates |= cur_mask; + if (rate_min > cur_value) + rate_min = cur_value; + if (rate_max < cur_value) + rate_max = cur_value; + } + } + + if (rates) { + pcm_hw->rates = rates; + pcm_hw->rate_min = rate_min; + pcm_hw->rate_max = rate_max; + } +} + +static void cfg_formats(char *list, unsigned int len, + const char *path, struct snd_pcm_hardware *pcm_hw) +{ + u64 formats; + char *cur_format; + int i; + + formats = 0; + while ((cur_format = strsep(&list, XENSND_LIST_SEPARATOR))) { + for (i = 0; i < ARRAY_SIZE(CFG_HW_SUPPORTED_FORMATS); i++) + if (!strncasecmp(cur_format, + CFG_HW_SUPPORTED_FORMATS[i].name, + XENSND_SAMPLE_FORMAT_MAX_LEN)) + formats |= CFG_HW_SUPPORTED_FORMATS[i].mask; + } + + if (formats) + pcm_hw->formats = formats; +} + +#define MAX_BUFFER_SIZE (64 * 1024) +#define MIN_PERIOD_SIZE 64 +#define MAX_PERIOD_SIZE MAX_BUFFER_SIZE +#define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE) +#define USE_RATE (SNDRV_PCM_RATE_CONTINUOUS | \ + SNDRV_PCM_RATE_8000_48000) +#define USE_RATE_MIN 5512 +#define USE_RATE_MAX 48000 +#define USE_CHANNELS_MIN 1 +#define USE_CHANNELS_MAX 2 +#define USE_PERIODS_MIN 2 +#define USE_PERIODS_MAX (MAX_BUFFER_SIZE / MIN_PERIOD_SIZE) + +static const struct snd_pcm_hardware SND_DRV_PCM_HW_DEFAULT = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = USE_FORMATS, + .rates = USE_RATE, + .rate_min = USE_RATE_MIN, + .rate_max = USE_RATE_MAX, + .channels_min = USE_CHANNELS_MIN, + .channels_max = USE_CHANNELS_MAX, + .buffer_bytes_max = MAX_BUFFER_SIZE, + .period_bytes_min = MIN_PERIOD_SIZE, + .period_bytes_max = MAX_PERIOD_SIZE, + .periods_min = USE_PERIODS_MIN, + .periods_max = USE_PERIODS_MAX, + .fifo_size = 0, +}; + +static void cfg_read_pcm_hw(const char *path, + struct snd_pcm_hardware *parent_pcm_hw, + struct snd_pcm_hardware *pcm_hw) +{ + char *list; + int val; + size_t buf_sz; + unsigned int len; + + /* Inherit parent's PCM HW and read overrides from XenStore. */ + if (parent_pcm_hw) + *pcm_hw = *parent_pcm_hw; + else + *pcm_hw = SND_DRV_PCM_HW_DEFAULT; + + val = xenbus_read_unsigned(path, XENSND_FIELD_CHANNELS_MIN, 0); + if (val) + pcm_hw->channels_min = val; + + val = xenbus_read_unsigned(path, XENSND_FIELD_CHANNELS_MAX, 0); + if (val) + pcm_hw->channels_max = val; + + list = xenbus_read(XBT_NIL, path, XENSND_FIELD_SAMPLE_RATES, &len); + if (!IS_ERR(list)) { + cfg_hw_rates(list, len, path, pcm_hw); + kfree(list); + } + + list = xenbus_read(XBT_NIL, path, XENSND_FIELD_SAMPLE_FORMATS, &len); + if (!IS_ERR(list)) { + cfg_formats(list, len, path, pcm_hw); + kfree(list); + } + + buf_sz = xenbus_read_unsigned(path, XENSND_FIELD_BUFFER_SIZE, 0); + if (buf_sz) + pcm_hw->buffer_bytes_max = buf_sz; + + /* Update configuration to match new values. */ + if (pcm_hw->channels_min > pcm_hw->channels_max) + pcm_hw->channels_min = pcm_hw->channels_max; + + if (pcm_hw->rate_min > pcm_hw->rate_max) + pcm_hw->rate_min = pcm_hw->rate_max; + + pcm_hw->period_bytes_max = pcm_hw->buffer_bytes_max; + + pcm_hw->periods_max = pcm_hw->period_bytes_max / + pcm_hw->period_bytes_min; +} + +static int cfg_get_stream_type(const char *path, int index, + int *num_pb, int *num_cap) +{ + char *str = NULL; + char *stream_path; + int ret; + + *num_pb = 0; + *num_cap = 0; + stream_path = kasprintf(GFP_KERNEL, "%s/%d", path, index); + if (!stream_path) { + ret = -ENOMEM; + goto fail; + } + + str = xenbus_read(XBT_NIL, stream_path, XENSND_FIELD_TYPE, NULL); + if (IS_ERR(str)) { + ret = PTR_ERR(str); + str = NULL; + goto fail; + } + + if (!strncasecmp(str, XENSND_STREAM_TYPE_PLAYBACK, + sizeof(XENSND_STREAM_TYPE_PLAYBACK))) { + (*num_pb)++; + } else if (!strncasecmp(str, XENSND_STREAM_TYPE_CAPTURE, + sizeof(XENSND_STREAM_TYPE_CAPTURE))) { + (*num_cap)++; + } else { + ret = -EINVAL; + goto fail; + } + ret = 0; + +fail: + kfree(stream_path); + kfree(str); + return ret; +} + +static int cfg_stream(struct xen_snd_front_info *front_info, + struct xen_front_cfg_pcm_instance *pcm_instance, + const char *path, int index, int *cur_pb, int *cur_cap, + int *stream_cnt) +{ + char *str = NULL; + char *stream_path; + struct xen_front_cfg_stream *stream; + int ret; + + stream_path = devm_kasprintf(&front_info->xb_dev->dev, + GFP_KERNEL, "%s/%d", path, index); + if (!stream_path) { + ret = -ENOMEM; + goto fail; + } + + str = xenbus_read(XBT_NIL, stream_path, XENSND_FIELD_TYPE, NULL); + if (IS_ERR(str)) { + ret = PTR_ERR(str); + str = NULL; + goto fail; + } + + if (!strncasecmp(str, XENSND_STREAM_TYPE_PLAYBACK, + sizeof(XENSND_STREAM_TYPE_PLAYBACK))) { + stream = &pcm_instance->streams_pb[(*cur_pb)++]; + } else if (!strncasecmp(str, XENSND_STREAM_TYPE_CAPTURE, + sizeof(XENSND_STREAM_TYPE_CAPTURE))) { + stream = &pcm_instance->streams_cap[(*cur_cap)++]; + } else { + ret = -EINVAL; + goto fail; + } + + /* Get next stream index. */ + stream->index = (*stream_cnt)++; + stream->xenstore_path = stream_path; + /* + * Check XenStore if PCM HW configuration exists for this stream + * and update if so, e.g. we inherit all values from device's PCM HW, + * but can still override some of the values for the stream. + */ + cfg_read_pcm_hw(stream->xenstore_path, + &pcm_instance->pcm_hw, &stream->pcm_hw); + ret = 0; + +fail: + kfree(str); + return ret; +} + +static int cfg_device(struct xen_snd_front_info *front_info, + struct xen_front_cfg_pcm_instance *pcm_instance, + struct snd_pcm_hardware *parent_pcm_hw, + const char *path, int node_index, int *stream_cnt) +{ + char *str; + char *device_path; + int ret, i, num_streams; + int num_pb, num_cap; + int cur_pb, cur_cap; + char node[3]; + + device_path = kasprintf(GFP_KERNEL, "%s/%d", path, node_index); + if (!device_path) + return -ENOMEM; + + str = xenbus_read(XBT_NIL, device_path, XENSND_FIELD_DEVICE_NAME, NULL); + if (!IS_ERR(str)) { + strlcpy(pcm_instance->name, str, sizeof(pcm_instance->name)); + kfree(str); + } + + pcm_instance->device_id = node_index; + + /* + * Check XenStore if PCM HW configuration exists for this device + * and update if so, e.g. we inherit all values from card's PCM HW, + * but can still override some of the values for the device. + */ + cfg_read_pcm_hw(device_path, parent_pcm_hw, &pcm_instance->pcm_hw); + + /* Find out how many streams were configured in Xen store. */ + num_streams = 0; + do { + snprintf(node, sizeof(node), "%d", num_streams); + if (!xenbus_exists(XBT_NIL, device_path, node)) + break; + + num_streams++; + } while (num_streams < VSND_MAX_STREAM); + + pcm_instance->num_streams_pb = 0; + pcm_instance->num_streams_cap = 0; + /* Get number of playback and capture streams. */ + for (i = 0; i < num_streams; i++) { + ret = cfg_get_stream_type(device_path, i, &num_pb, &num_cap); + if (ret < 0) + goto fail; + + pcm_instance->num_streams_pb += num_pb; + pcm_instance->num_streams_cap += num_cap; + } + + if (pcm_instance->num_streams_pb) { + pcm_instance->streams_pb = + devm_kcalloc(&front_info->xb_dev->dev, + pcm_instance->num_streams_pb, + sizeof(struct xen_front_cfg_stream), + GFP_KERNEL); + if (!pcm_instance->streams_pb) { + ret = -ENOMEM; + goto fail; + } + } + + if (pcm_instance->num_streams_cap) { + pcm_instance->streams_cap = + devm_kcalloc(&front_info->xb_dev->dev, + pcm_instance->num_streams_cap, + sizeof(struct xen_front_cfg_stream), + GFP_KERNEL); + if (!pcm_instance->streams_cap) { + ret = -ENOMEM; + goto fail; + } + } + + cur_pb = 0; + cur_cap = 0; + for (i = 0; i < num_streams; i++) { + ret = cfg_stream(front_info, pcm_instance, device_path, i, + &cur_pb, &cur_cap, stream_cnt); + if (ret < 0) + goto fail; + } + ret = 0; + +fail: + kfree(device_path); + return ret; +} + +int xen_snd_front_cfg_card(struct xen_snd_front_info *front_info, + int *stream_cnt) +{ + struct xenbus_device *xb_dev = front_info->xb_dev; + struct xen_front_cfg_card *cfg = &front_info->cfg; + int ret, num_devices, i; + char node[3]; + + *stream_cnt = 0; + num_devices = 0; + do { + snprintf(node, sizeof(node), "%d", num_devices); + if (!xenbus_exists(XBT_NIL, xb_dev->nodename, node)) + break; + + num_devices++; + } while (num_devices < SNDRV_PCM_DEVICES); + + if (!num_devices) { + dev_warn(&xb_dev->dev, + "No devices configured for sound card at %s\n", + xb_dev->nodename); + return -ENODEV; + } + + /* Start from default PCM HW configuration for the card. */ + cfg_read_pcm_hw(xb_dev->nodename, NULL, &cfg->pcm_hw); + + cfg->pcm_instances = + devm_kcalloc(&front_info->xb_dev->dev, num_devices, + sizeof(struct xen_front_cfg_pcm_instance), + GFP_KERNEL); + if (!cfg->pcm_instances) + return -ENOMEM; + + for (i = 0; i < num_devices; i++) { + ret = cfg_device(front_info, &cfg->pcm_instances[i], + &cfg->pcm_hw, xb_dev->nodename, i, stream_cnt); + if (ret < 0) + return ret; + } + cfg->num_pcm_instances = num_devices; + return 0; +} + diff --git a/sound/xen/xen_snd_front_cfg.h b/sound/xen/xen_snd_front_cfg.h new file mode 100644 index 000000000000..2353fcc74889 --- /dev/null +++ b/sound/xen/xen_snd_front_cfg.h @@ -0,0 +1,46 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_CFG_H +#define __XEN_SND_FRONT_CFG_H + +#include <sound/core.h> +#include <sound/pcm.h> + +struct xen_snd_front_info; + +struct xen_front_cfg_stream { + int index; + char *xenstore_path; + struct snd_pcm_hardware pcm_hw; +}; + +struct xen_front_cfg_pcm_instance { + char name[80]; + int device_id; + struct snd_pcm_hardware pcm_hw; + int num_streams_pb; + struct xen_front_cfg_stream *streams_pb; + int num_streams_cap; + struct xen_front_cfg_stream *streams_cap; +}; + +struct xen_front_cfg_card { + char name_short[32]; + char name_long[80]; + struct snd_pcm_hardware pcm_hw; + int num_pcm_instances; + struct xen_front_cfg_pcm_instance *pcm_instances; +}; + +int xen_snd_front_cfg_card(struct xen_snd_front_info *front_info, + int *stream_cnt); + +#endif /* __XEN_SND_FRONT_CFG_H */ diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c new file mode 100644 index 000000000000..102d6e096cc8 --- /dev/null +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -0,0 +1,494 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <xen/events.h> +#include <xen/grant_table.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_cfg.h" +#include "xen_snd_front_evtchnl.h" + +static irqreturn_t evtchnl_interrupt_req(int irq, void *dev_id) +{ + struct xen_snd_front_evtchnl *channel = dev_id; + struct xen_snd_front_info *front_info = channel->front_info; + struct xensnd_resp *resp; + RING_IDX i, rp; + + if (unlikely(channel->state != EVTCHNL_STATE_CONNECTED)) + return IRQ_HANDLED; + + mutex_lock(&channel->ring_io_lock); + +again: + rp = channel->u.req.ring.sring->rsp_prod; + /* Ensure we see queued responses up to rp. */ + rmb(); + + /* + * Assume that the backend is trusted to always write sane values + * to the ring counters, so no overflow checks on frontend side + * are required. + */ + for (i = channel->u.req.ring.rsp_cons; i != rp; i++) { + resp = RING_GET_RESPONSE(&channel->u.req.ring, i); + if (resp->id != channel->evt_id) + continue; + switch (resp->operation) { + case XENSND_OP_OPEN: + /* fall through */ + case XENSND_OP_CLOSE: + /* fall through */ + case XENSND_OP_READ: + /* fall through */ + case XENSND_OP_WRITE: + /* fall through */ + case XENSND_OP_TRIGGER: + channel->u.req.resp_status = resp->status; + complete(&channel->u.req.completion); + break; + case XENSND_OP_HW_PARAM_QUERY: + channel->u.req.resp_status = resp->status; + channel->u.req.resp.hw_param = + resp->resp.hw_param; + complete(&channel->u.req.completion); + break; + + default: + dev_err(&front_info->xb_dev->dev, + "Operation %d is not supported\n", + resp->operation); + break; + } + } + + channel->u.req.ring.rsp_cons = i; + if (i != channel->u.req.ring.req_prod_pvt) { + int more_to_do; + + RING_FINAL_CHECK_FOR_RESPONSES(&channel->u.req.ring, + more_to_do); + if (more_to_do) + goto again; + } else { + channel->u.req.ring.sring->rsp_event = i + 1; + } + + mutex_unlock(&channel->ring_io_lock); + return IRQ_HANDLED; +} + +static irqreturn_t evtchnl_interrupt_evt(int irq, void *dev_id) +{ + struct xen_snd_front_evtchnl *channel = dev_id; + struct xensnd_event_page *page = channel->u.evt.page; + u32 cons, prod; + + if (unlikely(channel->state != EVTCHNL_STATE_CONNECTED)) + return IRQ_HANDLED; + + mutex_lock(&channel->ring_io_lock); + + prod = page->in_prod; + /* Ensure we see ring contents up to prod. */ + virt_rmb(); + if (prod == page->in_cons) + goto out; + + /* + * Assume that the backend is trusted to always write sane values + * to the ring counters, so no overflow checks on frontend side + * are required. + */ + for (cons = page->in_cons; cons != prod; cons++) { + struct xensnd_evt *event; + + event = &XENSND_IN_RING_REF(page, cons); + if (unlikely(event->id != channel->evt_id++)) + continue; + + switch (event->type) { + case XENSND_EVT_CUR_POS: + xen_snd_front_alsa_handle_cur_pos(channel, + event->op.cur_pos.position); + break; + } + } + + page->in_cons = cons; + /* Ensure ring contents. */ + virt_wmb(); + +out: + mutex_unlock(&channel->ring_io_lock); + return IRQ_HANDLED; +} + +void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *channel) +{ + int notify; + + channel->u.req.ring.req_prod_pvt++; + RING_PUSH_REQUESTS_AND_CHECK_NOTIFY(&channel->u.req.ring, notify); + if (notify) + notify_remote_via_irq(channel->irq); +} + +static void evtchnl_free(struct xen_snd_front_info *front_info, + struct xen_snd_front_evtchnl *channel) +{ + unsigned long page = 0; + + if (channel->type == EVTCHNL_TYPE_REQ) + page = (unsigned long)channel->u.req.ring.sring; + else if (channel->type == EVTCHNL_TYPE_EVT) + page = (unsigned long)channel->u.evt.page; + + if (!page) + return; + + channel->state = EVTCHNL_STATE_DISCONNECTED; + if (channel->type == EVTCHNL_TYPE_REQ) { + /* Release all who still waits for response if any. */ + channel->u.req.resp_status = -EIO; + complete_all(&channel->u.req.completion); + } + + if (channel->irq) + unbind_from_irqhandler(channel->irq, channel); + + if (channel->port) + xenbus_free_evtchn(front_info->xb_dev, channel->port); + + /* End access and free the page. */ + if (channel->gref != GRANT_INVALID_REF) + gnttab_end_foreign_access(channel->gref, 0, page); + else + free_page(page); + + memset(channel, 0, sizeof(*channel)); +} + +void xen_snd_front_evtchnl_free_all(struct xen_snd_front_info *front_info) +{ + int i; + + if (!front_info->evt_pairs) + return; + + for (i = 0; i < front_info->num_evt_pairs; i++) { + evtchnl_free(front_info, &front_info->evt_pairs[i].req); + evtchnl_free(front_info, &front_info->evt_pairs[i].evt); + } + + kfree(front_info->evt_pairs); + front_info->evt_pairs = NULL; +} + +static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, + struct xen_snd_front_evtchnl *channel, + enum xen_snd_front_evtchnl_type type) +{ + struct xenbus_device *xb_dev = front_info->xb_dev; + unsigned long page; + grant_ref_t gref; + irq_handler_t handler; + char *handler_name = NULL; + int ret; + + memset(channel, 0, sizeof(*channel)); + channel->type = type; + channel->index = index; + channel->front_info = front_info; + channel->state = EVTCHNL_STATE_DISCONNECTED; + channel->gref = GRANT_INVALID_REF; + page = get_zeroed_page(GFP_KERNEL); + if (!page) { + ret = -ENOMEM; + goto fail; + } + + handler_name = kasprintf(GFP_KERNEL, "%s-%s", XENSND_DRIVER_NAME, + type == EVTCHNL_TYPE_REQ ? + XENSND_FIELD_RING_REF : + XENSND_FIELD_EVT_RING_REF); + if (!handler_name) { + ret = -ENOMEM; + goto fail; + } + + mutex_init(&channel->ring_io_lock); + + if (type == EVTCHNL_TYPE_REQ) { + struct xen_sndif_sring *sring = (struct xen_sndif_sring *)page; + + init_completion(&channel->u.req.completion); + mutex_init(&channel->u.req.req_io_lock); + SHARED_RING_INIT(sring); + FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); + + ret = xenbus_grant_ring(xb_dev, sring, 1, &gref); + if (ret < 0) { + channel->u.req.ring.sring = NULL; + goto fail; + } + + handler = evtchnl_interrupt_req; + } else { + ret = gnttab_grant_foreign_access(xb_dev->otherend_id, + virt_to_gfn((void *)page), 0); + if (ret < 0) + goto fail; + + channel->u.evt.page = (struct xensnd_event_page *)page; + gref = ret; + handler = evtchnl_interrupt_evt; + } + + channel->gref = gref; + + ret = xenbus_alloc_evtchn(xb_dev, &channel->port); + if (ret < 0) + goto fail; + + ret = bind_evtchn_to_irq(channel->port); + if (ret < 0) { + dev_err(&xb_dev->dev, + "Failed to bind IRQ for domid %d port %d: %d\n", + front_info->xb_dev->otherend_id, channel->port, ret); + goto fail; + } + + channel->irq = ret; + + ret = request_threaded_irq(channel->irq, NULL, handler, + IRQF_ONESHOT, handler_name, channel); + if (ret < 0) { + dev_err(&xb_dev->dev, "Failed to request IRQ %d: %d\n", + channel->irq, ret); + goto fail; + } + + kfree(handler_name); + return 0; + +fail: + if (page) + free_page(page); + kfree(handler_name); + dev_err(&xb_dev->dev, "Failed to allocate ring: %d\n", ret); + return ret; +} + +int xen_snd_front_evtchnl_create_all(struct xen_snd_front_info *front_info, + int num_streams) +{ + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct device *dev = &front_info->xb_dev->dev; + int d, ret = 0; + + front_info->evt_pairs = + kcalloc(num_streams, + sizeof(struct xen_snd_front_evtchnl_pair), + GFP_KERNEL); + if (!front_info->evt_pairs) + return -ENOMEM; + + /* Iterate over devices and their streams and create event channels. */ + for (d = 0; d < cfg->num_pcm_instances; d++) { + struct xen_front_cfg_pcm_instance *pcm_instance; + int s, index; + + pcm_instance = &cfg->pcm_instances[d]; + + for (s = 0; s < pcm_instance->num_streams_pb; s++) { + index = pcm_instance->streams_pb[s].index; + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].req, + EVTCHNL_TYPE_REQ); + if (ret < 0) { + dev_err(dev, "Error allocating control channel\n"); + goto fail; + } + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].evt, + EVTCHNL_TYPE_EVT); + if (ret < 0) { + dev_err(dev, "Error allocating in-event channel\n"); + goto fail; + } + } + + for (s = 0; s < pcm_instance->num_streams_cap; s++) { + index = pcm_instance->streams_cap[s].index; + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].req, + EVTCHNL_TYPE_REQ); + if (ret < 0) { + dev_err(dev, "Error allocating control channel\n"); + goto fail; + } + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].evt, + EVTCHNL_TYPE_EVT); + if (ret < 0) { + dev_err(dev, "Error allocating in-event channel\n"); + goto fail; + } + } + } + + front_info->num_evt_pairs = num_streams; + return 0; + +fail: + xen_snd_front_evtchnl_free_all(front_info); + return ret; +} + +static int evtchnl_publish(struct xenbus_transaction xbt, + struct xen_snd_front_evtchnl *channel, + const char *path, const char *node_ring, + const char *node_chnl) +{ + struct xenbus_device *xb_dev = channel->front_info->xb_dev; + int ret; + + /* Write control channel ring reference. */ + ret = xenbus_printf(xbt, path, node_ring, "%u", channel->gref); + if (ret < 0) { + dev_err(&xb_dev->dev, "Error writing ring-ref: %d\n", ret); + return ret; + } + + /* Write event channel ring reference. */ + ret = xenbus_printf(xbt, path, node_chnl, "%u", channel->port); + if (ret < 0) { + dev_err(&xb_dev->dev, "Error writing event channel: %d\n", ret); + return ret; + } + + return 0; +} + +int xen_snd_front_evtchnl_publish_all(struct xen_snd_front_info *front_info) +{ + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct xenbus_transaction xbt; + int ret, d; + +again: + ret = xenbus_transaction_start(&xbt); + if (ret < 0) { + xenbus_dev_fatal(front_info->xb_dev, ret, + "starting transaction"); + return ret; + } + + for (d = 0; d < cfg->num_pcm_instances; d++) { + struct xen_front_cfg_pcm_instance *pcm_instance; + int s, index; + + pcm_instance = &cfg->pcm_instances[d]; + + for (s = 0; s < pcm_instance->num_streams_pb; s++) { + index = pcm_instance->streams_pb[s].index; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].req, + pcm_instance->streams_pb[s].xenstore_path, + XENSND_FIELD_RING_REF, + XENSND_FIELD_EVT_CHNL); + if (ret < 0) + goto fail; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].evt, + pcm_instance->streams_pb[s].xenstore_path, + XENSND_FIELD_EVT_RING_REF, + XENSND_FIELD_EVT_EVT_CHNL); + if (ret < 0) + goto fail; + } + + for (s = 0; s < pcm_instance->num_streams_cap; s++) { + index = pcm_instance->streams_cap[s].index; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].req, + pcm_instance->streams_cap[s].xenstore_path, + XENSND_FIELD_RING_REF, + XENSND_FIELD_EVT_CHNL); + if (ret < 0) + goto fail; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].evt, + pcm_instance->streams_cap[s].xenstore_path, + XENSND_FIELD_EVT_RING_REF, + XENSND_FIELD_EVT_EVT_CHNL); + if (ret < 0) + goto fail; + } + } + ret = xenbus_transaction_end(xbt, 0); + if (ret < 0) { + if (ret == -EAGAIN) + goto again; + + xenbus_dev_fatal(front_info->xb_dev, ret, + "completing transaction"); + goto fail_to_end; + } + return 0; +fail: + xenbus_transaction_end(xbt, 1); +fail_to_end: + xenbus_dev_fatal(front_info->xb_dev, ret, "writing XenStore"); + return ret; +} + +void xen_snd_front_evtchnl_pair_set_connected(struct xen_snd_front_evtchnl_pair *evt_pair, + bool is_connected) +{ + enum xen_snd_front_evtchnl_state state; + + if (is_connected) + state = EVTCHNL_STATE_CONNECTED; + else + state = EVTCHNL_STATE_DISCONNECTED; + + mutex_lock(&evt_pair->req.ring_io_lock); + evt_pair->req.state = state; + mutex_unlock(&evt_pair->req.ring_io_lock); + + mutex_lock(&evt_pair->evt.ring_io_lock); + evt_pair->evt.state = state; + mutex_unlock(&evt_pair->evt.ring_io_lock); +} + +void xen_snd_front_evtchnl_pair_clear(struct xen_snd_front_evtchnl_pair *evt_pair) +{ + mutex_lock(&evt_pair->req.ring_io_lock); + evt_pair->req.evt_next_id = 0; + mutex_unlock(&evt_pair->req.ring_io_lock); + + mutex_lock(&evt_pair->evt.ring_io_lock); + evt_pair->evt.evt_next_id = 0; + mutex_unlock(&evt_pair->evt.ring_io_lock); +} + diff --git a/sound/xen/xen_snd_front_evtchnl.h b/sound/xen/xen_snd_front_evtchnl.h new file mode 100644 index 000000000000..cbe51fd1ec15 --- /dev/null +++ b/sound/xen/xen_snd_front_evtchnl.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_EVTCHNL_H +#define __XEN_SND_FRONT_EVTCHNL_H + +#include <xen/interface/io/sndif.h> + +struct xen_snd_front_info; + +#ifndef GRANT_INVALID_REF +/* + * FIXME: usage of grant reference 0 as invalid grant reference: + * grant reference 0 is valid, but never exposed to a PV driver, + * because of the fact it is already in use/reserved by the PV console. + */ +#define GRANT_INVALID_REF 0 +#endif + +/* Timeout in ms to wait for backend to respond. */ +#define VSND_WAIT_BACK_MS 3000 + +enum xen_snd_front_evtchnl_state { + EVTCHNL_STATE_DISCONNECTED, + EVTCHNL_STATE_CONNECTED, +}; + +enum xen_snd_front_evtchnl_type { + EVTCHNL_TYPE_REQ, + EVTCHNL_TYPE_EVT, +}; + +struct xen_snd_front_evtchnl { + struct xen_snd_front_info *front_info; + int gref; + int port; + int irq; + int index; + /* State of the event channel. */ + enum xen_snd_front_evtchnl_state state; + enum xen_snd_front_evtchnl_type type; + /* Either response id or incoming event id. */ + u16 evt_id; + /* Next request id or next expected event id. */ + u16 evt_next_id; + /* Shared ring access lock. */ + struct mutex ring_io_lock; + union { + struct { + struct xen_sndif_front_ring ring; + struct completion completion; + /* Serializer for backend IO: request/response. */ + struct mutex req_io_lock; + + /* Latest response status. */ + int resp_status; + union { + struct xensnd_query_hw_param hw_param; + } resp; + } req; + struct { + struct xensnd_event_page *page; + /* This is needed to handle XENSND_EVT_CUR_POS event. */ + struct snd_pcm_substream *substream; + } evt; + } u; +}; + +struct xen_snd_front_evtchnl_pair { + struct xen_snd_front_evtchnl req; + struct xen_snd_front_evtchnl evt; +}; + +int xen_snd_front_evtchnl_create_all(struct xen_snd_front_info *front_info, + int num_streams); + +void xen_snd_front_evtchnl_free_all(struct xen_snd_front_info *front_info); + +int xen_snd_front_evtchnl_publish_all(struct xen_snd_front_info *front_info); + +void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *evtchnl); + +void xen_snd_front_evtchnl_pair_set_connected(struct xen_snd_front_evtchnl_pair *evt_pair, + bool is_connected); + +void xen_snd_front_evtchnl_pair_clear(struct xen_snd_front_evtchnl_pair *evt_pair); + +#endif /* __XEN_SND_FRONT_EVTCHNL_H */ diff --git a/sound/xen/xen_snd_front_shbuf.c b/sound/xen/xen_snd_front_shbuf.c new file mode 100644 index 000000000000..07ac176a41ba --- /dev/null +++ b/sound/xen/xen_snd_front_shbuf.c @@ -0,0 +1,194 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/kernel.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include "xen_snd_front_shbuf.h" + +grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf) +{ + if (!buf->grefs) + return GRANT_INVALID_REF; + + return buf->grefs[0]; +} + +void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf) +{ + memset(buf, 0, sizeof(*buf)); +} + +void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf) +{ + int i; + + if (buf->grefs) { + for (i = 0; i < buf->num_grefs; i++) + if (buf->grefs[i] != GRANT_INVALID_REF) + gnttab_end_foreign_access(buf->grefs[i], + 0, 0UL); + kfree(buf->grefs); + } + kfree(buf->directory); + free_pages_exact(buf->buffer, buf->buffer_sz); + xen_snd_front_shbuf_clear(buf); +} + +/* + * number of grant references a page can hold with respect to the + * xensnd_page_directory header + */ +#define XENSND_NUM_GREFS_PER_PAGE ((XEN_PAGE_SIZE - \ + offsetof(struct xensnd_page_directory, gref)) / \ + sizeof(grant_ref_t)) + +static void fill_page_dir(struct xen_snd_front_shbuf *buf, + int num_pages_dir) +{ + struct xensnd_page_directory *page_dir; + unsigned char *ptr; + int i, cur_gref, grefs_left, to_copy; + + ptr = buf->directory; + grefs_left = buf->num_grefs - num_pages_dir; + /* + * skip grant references at the beginning, they are for pages granted + * for the page directory itself + */ + cur_gref = num_pages_dir; + for (i = 0; i < num_pages_dir; i++) { + page_dir = (struct xensnd_page_directory *)ptr; + if (grefs_left <= XENSND_NUM_GREFS_PER_PAGE) { + to_copy = grefs_left; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; + } else { + to_copy = XENSND_NUM_GREFS_PER_PAGE; + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + } + + memcpy(&page_dir->gref, &buf->grefs[cur_gref], + to_copy * sizeof(grant_ref_t)); + + ptr += XEN_PAGE_SIZE; + grefs_left -= to_copy; + cur_gref += to_copy; + } +} + +static int grant_references(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + int num_pages_dir, int num_pages_buffer, + int num_grefs) +{ + grant_ref_t priv_gref_head; + unsigned long frame; + int ret, i, j, cur_ref; + int otherend_id; + + ret = gnttab_alloc_grant_references(num_grefs, &priv_gref_head); + if (ret) + return ret; + + buf->num_grefs = num_grefs; + otherend_id = xb_dev->otherend_id; + j = 0; + + for (i = 0; i < num_pages_dir; i++) { + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) { + ret = cur_ref; + goto fail; + } + + frame = xen_page_to_gfn(virt_to_page(buf->directory + + XEN_PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + for (i = 0; i < num_pages_buffer; i++) { + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) { + ret = cur_ref; + goto fail; + } + + frame = xen_page_to_gfn(virt_to_page(buf->buffer + + XEN_PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + gnttab_free_grant_references(priv_gref_head); + fill_page_dir(buf, num_pages_dir); + return 0; + +fail: + gnttab_free_grant_references(priv_gref_head); + return ret; +} + +static int alloc_int_buffers(struct xen_snd_front_shbuf *buf, + int num_pages_dir, int num_pages_buffer, + int num_grefs) +{ + buf->grefs = kcalloc(num_grefs, sizeof(*buf->grefs), GFP_KERNEL); + if (!buf->grefs) + return -ENOMEM; + + buf->directory = kcalloc(num_pages_dir, XEN_PAGE_SIZE, GFP_KERNEL); + if (!buf->directory) + goto fail; + + buf->buffer_sz = num_pages_buffer * XEN_PAGE_SIZE; + buf->buffer = alloc_pages_exact(buf->buffer_sz, GFP_KERNEL); + if (!buf->buffer) + goto fail; + + return 0; + +fail: + kfree(buf->grefs); + buf->grefs = NULL; + kfree(buf->directory); + buf->directory = NULL; + return -ENOMEM; +} + +int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + unsigned int buffer_sz) +{ + int num_pages_buffer, num_pages_dir, num_grefs; + int ret; + + xen_snd_front_shbuf_clear(buf); + + num_pages_buffer = DIV_ROUND_UP(buffer_sz, XEN_PAGE_SIZE); + /* number of pages the page directory consumes itself */ + num_pages_dir = DIV_ROUND_UP(num_pages_buffer, + XENSND_NUM_GREFS_PER_PAGE); + num_grefs = num_pages_buffer + num_pages_dir; + + ret = alloc_int_buffers(buf, num_pages_dir, + num_pages_buffer, num_grefs); + if (ret < 0) + return ret; + + ret = grant_references(xb_dev, buf, num_pages_dir, num_pages_buffer, + num_grefs); + if (ret < 0) + return ret; + + fill_page_dir(buf, num_pages_dir); + return 0; +} diff --git a/sound/xen/xen_snd_front_shbuf.h b/sound/xen/xen_snd_front_shbuf.h new file mode 100644 index 000000000000..d28e97c47b2c --- /dev/null +++ b/sound/xen/xen_snd_front_shbuf.h @@ -0,0 +1,36 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_SHBUF_H +#define __XEN_SND_FRONT_SHBUF_H + +#include <xen/grant_table.h> + +#include "xen_snd_front_evtchnl.h" + +struct xen_snd_front_shbuf { + int num_grefs; + grant_ref_t *grefs; + u8 *directory; + u8 *buffer; + size_t buffer_sz; +}; + +grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf); + +int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + unsigned int buffer_sz); + +void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf); + +void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf); + +#endif /* __XEN_SND_FRONT_SHBUF_H */ |