From 926a01ce1ef5e27281af0270e4476979c0522954 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 22939142dd23..1f5d4872d623 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 6c941c8556dd9269be621cd8159fc60e955a91b3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 22939142dd23..1f5d4872d623 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 681b84e17747e1c208e8e1acc54cc5e612da84d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:00 +0100 Subject: sound: pcm: add vmalloc buffer helper functions There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 38 ++++++++++++++++++++++++++++++++++ sound/core/pcm_memory.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a79f16b..0ad2d28f2360 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -905,6 +905,44 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags); +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset); +#if 0 /* for kernel-doc */ +/** + * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is + * contiguous in kernel virtual space, but not in physical memory. Use this + * if the buffer is accessed by kernel code but not by device DMA. + * + * Returns 1 if the buffer was changed, 0 if not changed, or a negative error + * code. + */ +static int snd_pcm_lib_alloc_vmalloc_buffer + (struct snd_pcm_substream *substream, size_t size); +/** + * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses + * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. + */ +static int snd_pcm_lib_alloc_vmalloc_32_buffer + (struct snd_pcm_substream *substream, size_t size); +#endif +#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO) +#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO) + #ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index caa7796bc2f5..d9727c74b2e1 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -434,3 +434,57 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_pcm_lib_free_pages); + +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = __vmalloc(size, gfp_flags, PAGE_KERNEL); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); + +/** + * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + */ +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} +EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); + +/** + * snd_pcm_lib_get_vmalloc_page - map vmalloc buffer offset to page struct + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + * @offset: offset in the buffer + * + * This function is to be used as the page callback in the PCM ops. + */ +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} +EXPORT_SYMBOL(snd_pcm_lib_get_vmalloc_page); -- cgit v1.2.3 From ad8decb7f5dfd556e4a8400e37b127cd20d8e4c5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 19:01:50 +0100 Subject: ALSA: jazz16: Add support for Media Vision Jazz16 chipset This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/sb.h | 1 + sound/isa/Kconfig | 16 ++ sound/isa/sb/Makefile | 2 + sound/isa/sb/jazz16.c | 385 +++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/sb/sb8_main.c | 117 ++++++++++++-- sound/isa/sb/sb_common.c | 3 + sound/isa/sb/sb_mixer.c | 3 + 7 files changed, 511 insertions(+), 16 deletions(-) create mode 100644 sound/isa/sb/jazz16.c (limited to 'include') diff --git a/include/sound/sb.h b/include/sound/sb.h index 4e62ee1e4115..95353542256a 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 194af3b01e13..755a0a5f0e3f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -239,6 +239,22 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16 card and compatibles" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB8_DSP + help + Say Y here to include support for soundcards based on the + Media Vision Jazz16 chipset: digital chip MVD1216 (Jazz16), + codec MVA416 (CS4216) and mixer MVA514 (ICS2514). + Media Vision's Jazz16 cards were sold under names Pro Sonic 16, + Premium 3-D and Pro 3-D. There were also OEMs cards with the + Jazz16 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index faeffceb01b7..af3669681788 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o obj-$(CONFIG_SND_ES968) += snd-es968.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 000000000000..d52966b75846 --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,385 @@ + +/* + * jazz16.c - driver for Media Vision Jazz16 based soundcards. + * Copyright (C) 2009 Krzysztof Helt + * Based on patches posted by Rask Ingemann Lambertsen and Rene Herman. + * Based on OSS Sound Blaster driver. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include + +#define PFX "jazz16: " + +MODULE_DESCRIPTION("Media Vision Jazz16"); +MODULE_SUPPORTED_DEVICE("{{Media Vision ??? }," + "{RTL,RTL3000}}"); + +MODULE_AUTHOR("Krzysztof Helt "); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ +static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Media Vision Jazz16 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for jazz16 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); +module_param_array(dma8, int, NULL, 0444); +MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); +module_param_array(dma16, int, NULL, 0444); +MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); + +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_DSP_GET_JAZZ_BRD_REV 0xfa +#define SB_JAZZ16_SET_DMAINTR 0xfb +#define SB_DSP_GET_JAZZ_MODEL 0xfe + +struct snd_card_jazz16 { + struct snd_sb *chip; +}; + +static irqreturn_t jazz16_interrupt(int irq, void *chip) +{ + return snd_sb8dsp_interrupt(chip); +} + +static int __devinit jazz16_configure_ports(unsigned long port, + unsigned long mpu_port, int idx) +{ + unsigned char val; + + if (!request_region(0x201, 1, "jazz16 config")) { + snd_printk(KERN_ERR "config port region is already in use.\n"); + return -EBUSY; + } + outb(SB_JAZZ16_WAKEUP - idx, 0x201); + udelay(100); + outb(SB_JAZZ16_SET_PORTS + idx, 0x201); + udelay(100); + val = port & 0x70; + val |= (mpu_port & 0x30) >> 4; + outb(val, 0x201); + + release_region(0x201, 1); + return 0; +} + +static int __devinit jazz16_detect_board(unsigned long port, + unsigned long mpu_port) +{ + int err; + int val; + struct snd_sb chip; + + if (!request_region(port, 0x10, "jazz16")) { + snd_printk(KERN_ERR "I/O port region is already in use.\n"); + return -EBUSY; + } + /* just to call snd_sbdsp_command/reset/get_byte() */ + chip.port = port; + + err = snd_sbdsp_reset(&chip); + if (err < 0) + for (val = 0; val < 4; val++) { + err = jazz16_configure_ports(port, mpu_port, val); + if (err < 0) + break; + + err = snd_sbdsp_reset(&chip); + if (!err) + break; + } + if (err < 0) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_BRD_REV)) { + err = -EBUSY; + goto err_unmap; + } + val = snd_sbdsp_get_byte(&chip); + if (val >= 0x30) + snd_sbdsp_get_byte(&chip); + + if ((val & 0xf0) != 0x10) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_MODEL)) { + err = -EBUSY; + goto err_unmap; + } + snd_sbdsp_get_byte(&chip); + err = snd_sbdsp_get_byte(&chip); + snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); + + err = 0; + +err_unmap: + release_region(port, 0x10); + return err; +} + +static int __devinit jazz16_configure_board(struct snd_sb *chip, int mpu_irq) +{ + static unsigned char jazz_irq_bits[] = { 0, 0, 2, 3, 0, 1, 0, 4, + 0, 2, 5, 0, 0, 0, 0, 6 }; + static unsigned char jazz_dma_bits[] = { 0, 1, 0, 2, 0, 3, 0, 4 }; + + if (jazz_dma_bits[chip->dma8] == 0 || + jazz_dma_bits[chip->dma16] == 0 || + jazz_irq_bits[chip->irq] == 0) + return -EINVAL; + + if (!snd_sbdsp_command(chip, SB_JAZZ16_SET_DMAINTR)) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_dma_bits[chip->dma8] | + (jazz_dma_bits[chip->dma16] << 4))) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_irq_bits[chip->irq] | + (jazz_irq_bits[mpu_irq] << 4))) + return -EBUSY; + + return 0; +} + +static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please specify port\n"); + return 0; + } + if (dma16[dev] != SNDRV_AUTO_DMA && + dma16[dev] != 5 && dma16[dev] != 7) { + snd_printk(KERN_ERR "dma16 must be 5 or 7"); + return 0; + } + return 1; +} + +static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) +{ + struct snd_card *card; + struct snd_card_jazz16 *jazz16; + struct snd_sb *chip; + struct snd_opl3 *opl3; + static int possible_irqs[] = {2, 3, 5, 7, 9, 10, 15, -1}; + static int possible_dmas8[] = {1, 3, -1}; + static int possible_dmas16[] = {5, 7, -1}; + int err, xirq, xdma8, xdma16, xmpu_port, xmpu_irq; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_jazz16), &card); + if (err < 0) + return err; + + jazz16 = card->private_data; + + xirq = irq[dev]; + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_free; + } + } + xdma8 = dma8[dev]; + if (xdma8 == SNDRV_AUTO_DMA) { + xdma8 = snd_legacy_find_free_dma(possible_dmas8); + if (xdma8 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA8\n"); + err = -EBUSY; + goto err_free; + } + } + xdma16 = dma16[dev]; + if (xdma16 == SNDRV_AUTO_DMA) { + xdma16 = snd_legacy_find_free_dma(possible_dmas16); + if (xdma16 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA16\n"); + err = -EBUSY; + goto err_free; + } + } + + xmpu_port = mpu_port[dev]; + if (xmpu_port == SNDRV_AUTO_PORT) + xmpu_port = 0; + err = jazz16_detect_board(port[dev], xmpu_port); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + goto err_free; + } + err = snd_sbdsp_create(card, port[dev], irq[dev], + jazz16_interrupt, + dma8[dev], dma16[dev], + SB_HW_JAZZ16, + &chip); + if (err < 0) + goto err_free; + + xmpu_irq = mpu_irq[dev]; + if (xmpu_irq == SNDRV_AUTO_IRQ || mpu_port[dev] == SNDRV_AUTO_PORT) + xmpu_irq = 0; + err = jazz16_configure_board(chip, xmpu_irq); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + goto err_free; + } + + jazz16->chip = chip; + + strcpy(card->driver, "jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, + "Media Vision Jazz16 at 0x%lx, irq %d, dma8 %d, dma16 %d", + port[dev], xirq, xdma8, xdma16); + + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto err_free; + err = snd_sbmixer_new(chip); + if (err < 0) + goto err_free; + + err = snd_opl3_create(card, chip->port, chip->port + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); + else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_free; + } + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); + } + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_free; + + dev_set_drvdata(devptr, card); + return 0; + +err_free: + snd_card_free(card); + return err; +} + +static int __devexit snd_jazz16_remove(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + + dev_set_drvdata(devptr, NULL); + snd_card_free(card); + return 0; +} + +#ifdef CONFIG_PM +static int snd_jazz16_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + snd_sbmixer_suspend(chip); + return 0; +} + +static int snd_jazz16_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_sbdsp_reset(chip); + snd_sbmixer_resume(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), +#ifdef CONFIG_PM + .suspend = snd_jazz16_suspend, + .resume = snd_jazz16_resume, +#endif + .driver = { + .name = "jazz16" + }, +}; + +static int __init alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init(alsa_card_jazz16_init) +module_exit(alsa_card_jazz16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 658d55769c9c..3222aed5fac6 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -106,9 +106,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_CAPTURE_16) + return -EBUSY; + else + chip->mode |= SB_MODE_PLAYBACK_16; + } + chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -133,11 +145,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_PLAYBACK_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_PLAYBACK_8; + dma = chip->dma8; + } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); - if (runtime->channels > 1) { + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); @@ -147,15 +169,14 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); runtime->dma_area[0] = 0x80; - snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE); + snd_dma_program(dma, runtime->dma_addr, 1, DMA_MODE_WRITE); /* force interrupt */ - chip->mode = SB_MODE_HALT; snd_sbdsp_command(chip, SB_DSP_OUTPUT); snd_sbdsp_command(chip, 0); snd_sbdsp_command(chip, 0); } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save output filter status and turn it off */ @@ -168,13 +189,15 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -212,7 +235,6 @@ static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT; return 0; } @@ -234,9 +256,21 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_PLAYBACK_16) + return -EBUSY; + else + chip->mode |= SB_MODE_CAPTURE_16; + } + chip->capture_format = SB_DSP_LO_INPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -262,14 +296,24 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_CAPTURE_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_CAPTURE_8; + dma = chip->dma8; + } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save input filter status and turn it off */ @@ -282,13 +326,15 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_INPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -328,7 +374,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT; return 0; } @@ -339,13 +384,21 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) snd_sb_ack_8bit(chip); switch (chip->mode) { - case SB_MODE_PLAYBACK_8: /* ok.. playback is active */ + case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ + case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); break; + case SB_MODE_CAPTURE_16: + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; runtime = substream->runtime; @@ -361,10 +414,15 @@ static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *subs { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_PLAYBACK_8) + if (chip->mode & SB_MODE_PLAYBACK_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_PLAYBACK_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -372,10 +430,15 @@ static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *subst { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_CAPTURE_8) + if (chip->mode & SB_MODE_CAPTURE_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_CAPTURE_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -446,6 +509,13 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; + runtime->hw.rate_min = 4000; + runtime->hw.rate_max = 50000; + runtime->hw.channels_max = 2; + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -468,6 +538,14 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3 || chip->dma16 >= 0) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -480,6 +558,10 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + chip->mode &= ~SB_MODE_PLAYBACK; + else + chip->mode &= ~SB_MODE_CAPTURE; spin_unlock_irqrestore(&chip->open_lock, flags); return 0; } @@ -515,6 +597,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc = 64 * 1024; if (rpcm) *rpcm = NULL; @@ -527,9 +610,11 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + if (chip->dma8 > 3 || chip->dma16 >= 0) + max_prealloc = 128 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 27a651502251..eae6c1c0eff9 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -170,6 +170,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_CS5530: str = "16 (CS5530)"; break; + case SB_HW_JAZZ16: + str = "Pro (Jazz16)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 8cfc41fbe368..6496822c1808 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -779,6 +779,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -929,6 +930,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -955,6 +957,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: -- cgit v1.2.3 From 41116e926cb92292fa4fcbe888ae8133fa0038e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 09:00:14 +0100 Subject: ALSA: cs46xx - Fix suspend/resume with new DSP Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- include/sound/cs46xx_dsp_spos.h | 6 ++++-- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 42 +++++++++++++++++++++++++++++++++---- sound/pci/cs46xx/dsp_spos.h | 4 ++++ sound/pci/cs46xx/dsp_spos_scb_lib.c | 33 +++++++++++++---------------- 5 files changed, 62 insertions(+), 25 deletions(-) (limited to 'include') diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index 7c44667e79a6..49b03c9e5e55 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -118,9 +118,11 @@ struct dsp_scb_descriptor { struct snd_info_entry *proc_info; int ref_count; - spinlock_t lock; - int deleted; + u16 volume[2]; + unsigned int deleted :1; + unsigned int updated :1; + unsigned int volume_set :1; }; struct dsp_task_descriptor { diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1be96ead4244..e6b4a879ae2e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3597,7 +3597,7 @@ static struct cs_card_type __devinitdata cards[] = { #ifdef CONFIG_PM static unsigned int saved_regs[] = { BA0_ACOSV, - BA0_ASER_FADDR, + /*BA0_ASER_FADDR,*/ BA0_ASER_MASTER, BA1_PVOL, BA1_CVOL, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f4f0c8f5dad7..3e5ca8fb519f 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -298,6 +298,9 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); +#ifdef CONFIG_PM + kfree(ins->scbs[i].data); +#endif } kfree(ins->code.data); @@ -974,13 +977,11 @@ static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * nam index = find_free_scb_index (ins); + memset(&ins->scbs[index], 0, sizeof(ins->scbs[index])); strcpy(ins->scbs[index].scb_name, name); ins->scbs[index].address = dest; ins->scbs[index].index = index; - ins->scbs[index].proc_info = NULL; ins->scbs[index].ref_count = 1; - ins->scbs[index].deleted = 0; - spin_lock_init(&ins->scbs[index].lock); desc = (ins->scbs + index); ins->scbs[index].scb_symbol = add_symbol (chip, name, dest, SYMBOL_PARAMETER); @@ -1022,17 +1023,29 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return desc; } +#define SCB_BYTES (0x10 * 4) + struct dsp_scb_descriptor * cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest) { struct dsp_scb_descriptor * desc; +#ifdef CONFIG_PM + /* copy the data for resume */ + scb_data = kmemdup(scb_data, SCB_BYTES, GFP_KERNEL); + if (!scb_data) + return NULL; +#endif + desc = _map_scb (chip,name,dest); if (desc) { desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); +#ifdef CONFIG_PM + kfree(scb_data); +#endif } return desc; @@ -1988,7 +2001,28 @@ int cs46xx_dsp_resume(struct snd_cs46xx * chip) continue; _dsp_create_scb(chip, s->data, s->address); } - + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + if (s->updated) + cs46xx_dsp_spos_update_scb(chip, s); + if (s->volume_set) + cs46xx_dsp_scb_set_volume(chip, s, + s->volume[0], s->volume[1]); + } + if (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) { + cs46xx_dsp_enable_spdif_hw(chip); + snd_cs46xx_poke(chip, (ins->ref_snoop_scb->address + 2) << 2, + (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10); + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) + cs46xx_poke_via_dsp(chip, SP_SPDOUT_CSUV, + ins->spdif_csuv_stream); + } + if (chip->dsp_spos_instance->spdif_status_in) { + cs46xx_poke_via_dsp(chip, SP_ASER_COUNTDOWN, 0x80000005); + cs46xx_poke_via_dsp(chip, SP_SPDIN_CONTROL, 0x800003ff); + } return 0; } #endif diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index f9e169d33c03..ca47a8114c7f 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -212,6 +212,7 @@ static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip, (scb->address + SCBsubListPtr) << 2, (scb->sub_list_ptr->address << 0x10) | (scb->next_scb_ptr->address)); + scb->updated = 1; } static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, @@ -222,6 +223,9 @@ static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val); snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val); + scb->volume_set = 1; + scb->volume[0] = left; + scb->volume[1] = right; } #endif /* __DSP_SPOS_H__ */ #endif /* CONFIG_SND_CS46XX_NEW_DSP */ diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index dd7c41b037b4..00b148a10239 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -115,7 +115,6 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - unsigned long flags; if ( scb->parent_scb_ptr ) { /* unlink parent SCB */ @@ -153,8 +152,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor scb->next_scb_ptr = ins->the_null_scb; } - spin_lock_irqsave(&chip->reg_lock, flags); - /* update parent first entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr); @@ -162,7 +159,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor cs46xx_dsp_spos_update_scb(chip,scb); scb->parent_scb_ptr = NULL; - spin_unlock_irqrestore(&chip->reg_lock, flags); } } @@ -197,9 +193,9 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * goto _end; #endif - spin_lock_irqsave(&scb->lock, flags); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,scb); - spin_unlock_irqrestore(&scb->lock, flags); + spin_unlock_irqrestore(&chip->reg_lock, flags); cs46xx_dsp_proc_free_scb_desc(scb); if (snd_BUG_ON(!scb->scb_symbol)) @@ -207,6 +203,10 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * remove_symbol (chip,scb->scb_symbol); ins->scbs[scb->index].deleted = 1; +#ifdef CONFIG_PM + kfree(ins->scbs[scb->index].data); + ins->scbs[scb->index].data = NULL; +#endif if (scb->index < ins->scb_highest_frag_index) ins->scb_highest_frag_index = scb->index; @@ -1508,20 +1508,17 @@ int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, chip->dsp_spos_instance->npcm_channels <= 0)) return -EIO; - spin_lock(&pcm_channel->src_scb->lock); - + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } - spin_lock_irqsave(&chip->reg_lock, flags); pcm_channel->unlinked = 1; - spin_unlock_irqrestore(&chip->reg_lock, flags); _dsp_unlink_scb (chip,pcm_channel->pcm_reader_scb); + spin_unlock_irqrestore(&chip->reg_lock, flags); - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1533,10 +1530,10 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb; unsigned long flags; - spin_lock(&pcm_channel->src_scb->lock); + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked == 0) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } @@ -1552,8 +1549,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, snd_BUG_ON(pcm_channel->pcm_reader_scb->parent_scb_ptr); pcm_channel->pcm_reader_scb->parent_scb_ptr = parent_scb; - spin_lock_irqsave(&chip->reg_lock, flags); - /* update SCB entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,pcm_channel->pcm_reader_scb); @@ -1562,8 +1557,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, pcm_channel->unlinked = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); - - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1596,13 +1589,17 @@ cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * s int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) { + unsigned long flags; + if (snd_BUG_ON(!src->parent_scb_ptr)) return -EINVAL; /* mute SCB */ cs46xx_dsp_scb_set_volume (chip,src,0,0); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,src); + spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -- cgit v1.2.3 From 4757968dbff3d43f373f08de973014a9bd41ef0a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 28 Dec 2009 16:15:03 +0100 Subject: ALSA: Release v1.0.22.1 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 1f5d4872d623..7fed23442db8 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.22" +#define CONFIG_SND_VERSION "1.0.22.1" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 11:47:57 +0100 Subject: ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 6 +++ sound/core/pcm.c | 4 ++ sound/core/pcm_lib.c | 140 ++++++++++++++++++++++++++++++++++++++++++--------- 3 files changed, 127 insertions(+), 23 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a79f16b..4e18a6dbe690 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -262,6 +262,8 @@ struct snd_pcm_hw_constraint_list { unsigned int mask; }; +struct snd_pcm_hwptr_log; + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -340,6 +342,10 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif + +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + struct snd_pcm_hwptr_log *hwptr_log; +#endif }; struct snd_pcm_group { /* keep linked substreams */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6884ae031f6f..df57a0e30bf2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -921,6 +921,10 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + if (runtime->hwptr_log) + kfree(runtime->hwptr_log); +#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9621236b2fef..1990afb8a735 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,34 +126,34 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} + #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ #define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ #define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ #define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ +#define XRUN_DEBUG_LOG (1<<5) /* show last 10 positions on err */ +#define XRUN_DEBUG_LOGONCE (1<<6) /* do above only once */ #ifdef CONFIG_SND_PCM_XRUN_DEBUG + #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) -#else -#define xrun_debug(substream, mask) 0 -#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ dump_stack(); \ } while (0) -static void pcm_debug_name(struct snd_pcm_substream *substream, - char *name, size_t len) -{ - snprintf(name, len, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} - static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -169,6 +169,102 @@ static void xrun(struct snd_pcm_substream *substream) } } +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +#define XRUN_LOG_CNT 10 + +struct hwptr_log_entry { + unsigned long jiffies; + snd_pcm_uframes_t pos; + snd_pcm_uframes_t period_size; + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t old_hw_ptr; + snd_pcm_uframes_t hw_ptr_base; + snd_pcm_uframes_t hw_ptr_interrupt; +}; + +struct snd_pcm_hwptr_log { + unsigned int idx; + unsigned int hit: 1; + struct hwptr_log_entry entries[XRUN_LOG_CNT]; +}; + +static void xrun_log(struct snd_pcm_substream *substream, + snd_pcm_uframes_t pos) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hwptr_log *log = runtime->hwptr_log; + struct hwptr_log_entry *entry; + + if (log == NULL) { + log = kzalloc(sizeof(*log), GFP_ATOMIC); + if (log == NULL) + return; + runtime->hwptr_log = log; + } else { + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + } + entry = &log->entries[log->idx]; + entry->jiffies = jiffies; + entry->pos = pos; + entry->period_size = runtime->period_size; + entry->buffer_size = runtime->buffer_size;; + entry->old_hw_ptr = runtime->status->hw_ptr; + entry->hw_ptr_base = runtime->hw_ptr_base; + entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; + log->idx = (log->idx + 1) % XRUN_LOG_CNT; +} + +static void xrun_log_show(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hwptr_log *log = substream->runtime->hwptr_log; + struct hwptr_log_entry *entry; + char name[16]; + unsigned int idx; + int cnt; + + if (log == NULL) + return; + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + pcm_debug_name(substream, name, sizeof(name)); + for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { + entry = &log->entries[idx]; + if (entry->period_size == 0) + break; + snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, entry->jiffies, (unsigned long)entry->pos, + (unsigned long)entry->period_size, + (unsigned long)entry->buffer_size, + (unsigned long)entry->old_hw_ptr, + (unsigned long)entry->hw_ptr_base, + (unsigned long)entry->hw_ptr_interrupt); + idx++; + idx %= XRUN_LOG_CNT; + } + log->hit = 1; +} + +#else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ + +#define xrun_debug(substream, mask) 0 +#define xrun(substream) do { } while (0) +#define hw_ptr_error(substream, fmt, args...) do { } while (0) +#define xrun_log(substream, pos) do { } while (0) +#define xrun_log_show(substream) do { } while (0) + +#endif + static snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) @@ -182,6 +278,7 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, if (printk_ratelimit()) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " "buffer size = 0x%lx, period size = 0x%lx\n", name, pos, runtime->buffer_size, @@ -190,6 +287,8 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, pos = 0; } pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); return pos; } @@ -220,16 +319,6 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -#define hw_ptr_error(substream, fmt, args...) \ - do { \ - if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ - if (printk_ratelimit()) { \ - snd_printd("PCM: " fmt, ##args); \ - } \ - dump_stack_on_xrun(substream); \ - } \ - } while (0) - static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -270,6 +359,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", @@ -315,6 +405,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! [Q] " "(pos=%ld, delta=%ld, period=%ld, " @@ -334,6 +425,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { + xrun_log_show(substream); hw_ptr_error(substream, "Lost interrupts? " "(stream=%i, delta=%ld, intr_ptr=%ld)\n", @@ -397,6 +489,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) if (delta < 0) { delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value [2] " "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", @@ -416,6 +509,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) goto no_jiffies_check; delta -= runtime->delay; if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", -- cgit v1.2.3 From f240406babfe1526998e10583ea5eccc2676a433 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jan 2010 17:19:34 +0100 Subject: ALSA: pcm_lib - cleanup & merge hw_ptr update functions Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 - include/sound/pcm_oss.h | 2 +- sound/core/oss/pcm_oss.c | 32 ++++-- sound/core/pcm_lib.c | 279 ++++++++++++++++------------------------------- sound/core/pcm_native.c | 2 - 5 files changed, 121 insertions(+), 195 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e18a6dbe690..fe1b131842be 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,7 +271,6 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ - snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index cc4e226f35fd..760c969d885d 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -61,7 +61,7 @@ struct snd_pcm_oss_runtime { struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; #endif - unsigned int prev_hw_ptr_interrupt; + unsigned int prev_hw_ptr_period; }; struct snd_pcm_oss_file { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d9c96353121a..255ad910077a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -632,6 +632,13 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +static inline +snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) +{ + snd_pcm_uframes_t ptr = runtime->status->hw_ptr; + return ptr - (ptr % runtime->period_size); +} + /* define extended formats in the recent OSS versions (if any) */ /* linear formats */ #define AFMT_S32_LE 0x00001000 @@ -1102,7 +1109,7 @@ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) return err; } runtime->oss.prepare = 0; - runtime->oss.prev_hw_ptr_interrupt = 0; + runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; runtime->oss.buffer_used = 0; @@ -1950,7 +1957,8 @@ static int snd_pcm_oss_get_caps(struct snd_pcm_oss_file *pcm_oss_file) return result; } -static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, snd_pcm_uframes_t hw_ptr) +static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, + snd_pcm_uframes_t hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t appl_ptr; @@ -1986,7 +1994,8 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr if (runtime->oss.trigger) goto _skip1; if (atomic_read(&psubstream->mmap_count)) - snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt); + snd_pcm_oss_simulate_fill(psubstream, + get_hw_ptr_period(runtime)); runtime->oss.trigger = 1; runtime->start_threshold = 1; cmd = SNDRV_PCM_IOCTL_START; @@ -2105,11 +2114,12 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size); if (atomic_read(&substream->mmap_count)) { snd_pcm_sframes_t n; - n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt; + delay = get_hw_ptr_period(runtime); + n = delay - runtime->oss.prev_hw_ptr_period; if (n < 0) n += runtime->boundary; info.blocks = n / runtime->period_size; - runtime->oss.prev_hw_ptr_interrupt = delay; + runtime->oss.prev_hw_ptr_period = delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_pcm_oss_simulate_fill(substream, delay); info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX; @@ -2673,18 +2683,22 @@ static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_playback_avail(runtime) >= + runtime->oss.period_frames; } static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_capture_avail(runtime) >= + runtime->oss.period_frames; } static unsigned int snd_pcm_oss_poll(struct file *file, poll_table * wait) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1990afb8a735..70a4f7428d78 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -172,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + xrun_log_show(substream); \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -188,7 +189,6 @@ struct hwptr_log_entry { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t old_hw_ptr; snd_pcm_uframes_t hw_ptr_base; - snd_pcm_uframes_t hw_ptr_interrupt; }; struct snd_pcm_hwptr_log { @@ -220,7 +220,6 @@ static void xrun_log(struct snd_pcm_substream *substream, entry->buffer_size = runtime->buffer_size;; entry->old_hw_ptr = runtime->status->hw_ptr; entry->hw_ptr_base = runtime->hw_ptr_base; - entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; log->idx = (log->idx + 1) % XRUN_LOG_CNT; } @@ -241,14 +240,13 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + "hwptr=%ld/%ld\n", name, entry->jiffies, (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, - (unsigned long)entry->hw_ptr_base, - (unsigned long)entry->hw_ptr_interrupt); + (unsigned long)entry->hw_ptr_base); idx++; idx %= XRUN_LOG_CNT; } @@ -265,33 +263,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static snd_pcm_uframes_t -snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) -{ - snd_pcm_uframes_t pos; - - pos = substream->ops->pointer(substream); - if (pos == SNDRV_PCM_POS_XRUN) - return pos; /* XRUN */ - if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - xrun_log_show(substream); - snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " - "buffer size = 0x%lx, period size = 0x%lx\n", - name, pos, runtime->buffer_size, - runtime->period_size); - } - pos = 0; - } - pos -= pos % runtime->min_align; - if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); - return pos; -} - static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { @@ -319,72 +290,88 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + unsigned int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_ptr_interrupt, hw_base; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t hdelta, delta; unsigned long jdelta; old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); + pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; } - if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); + if (pos >= runtime->buffer_size) { + if (printk_ratelimit()) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); + snd_printd(KERN_ERR "BUG: %s, pos = %ld, " + "buffer size = %ld, period size = %ld\n", + name, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } + pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; - hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - delta = new_hw_ptr - hw_ptr_interrupt; - if (hw_ptr_interrupt >= runtime->boundary) { - hw_ptr_interrupt -= runtime->boundary; - if (hw_base < runtime->boundary / 2) - /* hw_base was already lapped; recalc delta */ - delta = new_hw_ptr - hw_ptr_interrupt; - } - if (delta < 0) { - if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value " - "(stream=%i, pos=%ld, intr_ptr=%ld)\n", - substream->stream, (long)pos, - (long)hw_ptr_interrupt); -#if 1 - /* simply skipping the hwptr update seems more - * robust in some cases, e.g. on VMware with - * inaccurate timer source - */ - return 0; /* skip this update */ -#else - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; -#endif - } else { + if (in_interrupt) { + /* we know that one period was processed */ + /* delta = "expected next hw_ptr" for in_interrupt != 0 */ + delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + + runtime->period_size; + if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; + goto __delta; } } + /* new_hw_ptr might be lower than old_hw_ptr in case when */ + /* pointer crosses the end of the ring buffer */ + if (new_hw_ptr < old_hw_ptr) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + __delta: + delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary; + if (xrun_debug(substream, in_interrupt ? + XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("%s_update: %s: pos=%u/%u/%u, " + "hwptr=%ld/%ld/%ld/%ld\n", + in_interrupt ? "period" : "hwptr", + name, + (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr, + (unsigned long)runtime->hw_ptr_base); + } + /* something must be really wrong */ + if (delta >= runtime->buffer_size) { + hw_ptr_error(substream, + "Unexpected hw_pointer value %s" + "(stream=%i, pos=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "[P]", + substream->stream, (long)pos, + (long)new_hw_ptr, (long)old_hw_ptr); + return 0; + } /* Do jiffies check only in xrun_debug mode */ if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) @@ -396,7 +383,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) */ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; - hdelta = new_hw_ptr - old_hw_ptr; + hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; @@ -405,45 +392,49 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); - xrun_log_show(substream); + /* move new_hw_ptr according jiffies not pos variable */ + new_hw_ptr = old_hw_ptr; + /* use loop to avoid checks for delta overflows */ + /* the delta value is small or zero in most cases */ + while (delta > 0) { + new_hw_ptr += runtime->period_size; + if (new_hw_ptr >= runtime->boundary) + new_hw_ptr -= runtime->boundary; + delta--; + } + /* align hw_base to buffer_size */ + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); + delta = 0; hw_ptr_error(substream, - "hw_ptr skipping! [Q] " + "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " - "jdelta=%lu/%lu/%lu)\n", + "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", + in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta); - hw_ptr_interrupt = runtime->hw_ptr_interrupt + - runtime->period_size * delta; - if (hw_ptr_interrupt >= runtime->boundary) - hw_ptr_interrupt -= runtime->boundary; - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; + ((hdelta * HZ) / runtime->rate), delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { - xrun_log_show(substream); hw_ptr_error(substream, - "Lost interrupts? " - "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + "Lost interrupts? %s" + "(stream=%i, delta=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "", substream->stream, (long)delta, - (long)hw_ptr_interrupt); - /* rebase hw_ptr_interrupt */ - hw_ptr_interrupt = - new_hw_ptr - new_hw_ptr % runtime->period_size; + (long)new_hw_ptr, + (long)old_hw_ptr); } - runtime->hw_ptr_interrupt = hw_ptr_interrupt; + + if (runtime->status->hw_ptr == new_hw_ptr) + return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; @@ -456,83 +447,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) /* CAUTION: call it with irq disabled */ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; - snd_pcm_sframes_t delta; - unsigned long jdelta; - - old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); - if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); - return -EPIPE; - } - if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); - } - - hw_base = runtime->hw_ptr_base; - new_hw_ptr = hw_base + pos; - - delta = new_hw_ptr - old_hw_ptr; - jdelta = jiffies - runtime->hw_ptr_jiffies; - if (delta < 0) { - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value [2] " - "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", - substream->stream, (long)pos, - (long)old_hw_ptr, jdelta); - return 0; - } - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - } - /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) - goto no_jiffies_check; - if (delta < runtime->delay) - goto no_jiffies_check; - delta -= runtime->delay; - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { - xrun_log_show(substream); - hw_ptr_error(substream, - "hw_ptr skipping! " - "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", - (long)pos, (long)delta, - (long)runtime->period_size, jdelta, - ((delta * HZ) / runtime->rate)); - return 0; - } - no_jiffies_check: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - - runtime->hw_ptr_base = hw_base; - runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_hw_ptr0(substream, 0); } /** @@ -1744,7 +1659,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || - snd_pcm_update_hw_ptr_interrupt(substream) < 0) + snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; if (substream->timer_running) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a12e11..8e777f71717c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1247,8 +1247,6 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; - runtime->hw_ptr_interrupt = runtime->status->hw_ptr - - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.2.3 From 1250932e48d3b698415b1f04775433cf1da688d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 7 Jan 2010 15:36:31 +0100 Subject: ALSA: pcm_lib - optimize wake_up() calls for PCM I/O As noted by pl bossart , the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 3 +++ sound/core/pcm_lib.c | 30 ++++++++++++++++++++---------- sound/core/pcm_native.c | 6 ++++-- 3 files changed, 27 insertions(+), 12 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fe1b131842be..e26fb3c58037 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,6 +311,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ + unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ wait_queue_head_t sleep; struct fasync_struct *fasync; @@ -839,6 +840,8 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 70a4f7428d78..a63226232ef4 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -263,8 +263,8 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -285,7 +285,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + if (!runtime->nowake && avail >= runtime->control->avail_min) wake_up(&runtime->sleep); return 0; } @@ -441,7 +441,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_state(substream, runtime); } /* CAUTION: call it with irq disabled */ @@ -1792,6 +1792,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1813,15 +1814,17 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -1850,8 +1853,10 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } @@ -2009,6 +2014,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2037,15 +2043,17 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -2068,8 +2076,10 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8e777f71717c..27284f628361 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, struct snd_pcm_sw_params *params) { struct snd_pcm_runtime *runtime; + int err; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -540,6 +541,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->silence_threshold > runtime->buffer_size) return -EINVAL; } + err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; runtime->period_step = params->period_step; @@ -553,10 +555,10 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - wake_up(&runtime->sleep); + err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); - return 0; + return err; } static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, -- cgit v1.2.3 From d1458279bf9c575a52fd22818ca19c463f380aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:16:52 +0100 Subject: ALSA: Add snd_pci_quirk_lookup_id() Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 +++ sound/core/misc.c | 32 +++++++++++++++++++++++++++----- 2 files changed, 30 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index a61499c22b0b..89e0ac17f44a 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -458,5 +458,8 @@ struct snd_pci_quirk { const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); +const struct snd_pci_quirk * +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list); #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 23a032c6d487..3da4f92427d8 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -101,8 +101,9 @@ EXPORT_SYMBOL_GPL(__snd_printk); #ifdef CONFIG_PCI #include /** - * snd_pci_quirk_lookup - look up a PCI SSID quirk list - * @pci: pci_dev handle + * snd_pci_quirk_lookup_id - look up a PCI SSID quirk list + * @vendor: PCI SSV id + * @device: PCI SSD id * @list: quirk list, terminated by a null entry * * Look through the given quirk list and finds a matching entry @@ -112,18 +113,39 @@ EXPORT_SYMBOL_GPL(__snd_printk); * Returns the matched entry pointer, or NULL if nothing matched. */ const struct snd_pci_quirk * -snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; for (q = list; q->subvendor; q++) { - if (q->subvendor != pci->subsystem_vendor) + if (q->subvendor != vendor) continue; if (!q->subdevice || - (pci->subsystem_device & q->subdevice_mask) == q->subdevice) + (device & q->subdevice_mask) == q->subdevice) return q; } return NULL; } +EXPORT_SYMBOL(snd_pci_quirk_lookup_id); + +/** + * snd_pci_quirk_lookup - look up a PCI SSID quirk list + * @pci: pci_dev handle + * @list: quirk list, terminated by a null entry + * + * Look through the given quirk list and finds a matching entry + * with the same PCI SSID. When subdevice is 0, all subdevice + * values may match. + * + * Returns the matched entry pointer, or NULL if nothing matched. + */ +const struct snd_pci_quirk * +snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +{ + return snd_pci_quirk_lookup_id(pci->subsystem_vendor, + pci->subsystem_device, + list); +} EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif -- cgit v1.2.3 From c32d977b8157bf67cdf47729ce7dd054a26eb534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:58:57 +0100 Subject: ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++++ sound/core/pcm_native.c | 9 +++++++++ sound/drivers/vx/vx_pcm.c | 2 ++ sound/mips/sgio2audio.c | 3 +++ sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 + sound/usb/ua101.c | 2 ++ sound/usb/usbaudio.c | 2 ++ 7 files changed, 23 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d4ca2aae50d..aabf48bb8ee6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1021,6 +1021,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area); +#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached + static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { *max = dma < 4 ? 64 * 1024 : 128 * 1024; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5df0d21f18b3..88fff44702a4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3176,6 +3176,15 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ +/* mmap callback with pgprot_noncached */ +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); + return snd_pcm_default_mmap(substream, area); +} +EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); + /* * mmap DMA buffer */ diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index c8385d26a16f..35a2f71a6af5 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -905,6 +905,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1125,6 +1126,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9b486beeb932..6aff217379d9 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -691,6 +691,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -703,6 +704,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -715,6 +717,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0afa683c900e..0d668f471620 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -277,6 +277,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 16dc7bd5e120..4f4ccdf70dd0 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -911,6 +911,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -923,6 +924,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4ada98e16309..b8e0b8fda607 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1997,6 +1997,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -2009,6 +2010,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; -- cgit v1.2.3 From c91a988dc6551c66418690e36b2a23cdb0255da8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 10:32:15 +0100 Subject: ALSA: pcm_core: Fix wake_up() optimization This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 5 +++-- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 20 ++++++++++---------- sound/core/pcm_native.c | 3 +++ 4 files changed, 17 insertions(+), 12 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e26fb3c58037..3bc9bca771ec 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,8 +311,9 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ - wait_queue_head_t sleep; + unsigned int twake: 1; /* do transfer (!poll) wakeup */ + wait_queue_head_t sleep; /* poll sleep */ + wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; /* -- private section -- */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index df57a0e30bf2..0d428d0896db 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,6 +894,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5417f7dce834..e2a817eac2a9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -285,8 +285,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (!runtime->nowake && avail >= runtime->control->avail_min) - wake_up(&runtime->sleep); + if (avail >= runtime->control->avail_min) + wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); return 0; } @@ -1692,7 +1692,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, long tout; init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); + add_wait_queue(&runtime->tsleep, &wait); for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; @@ -1735,7 +1735,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, break; } _endloop: - remove_wait_queue(&runtime->sleep, &wait); + remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; } @@ -1794,7 +1794,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1816,7 +1816,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -1855,7 +1855,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); @@ -2016,7 +2016,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2045,7 +2045,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -2078,7 +2078,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 27284f628361..56ec35e8510b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -919,6 +919,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) runtime->status->state = state; } wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_stop = { @@ -1004,6 +1005,7 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) SNDRV_TIMER_EVENT_MPAUSE, &runtime->trigger_tstamp); wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; if (substream->timer) @@ -1061,6 +1063,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_suspend = { -- cgit v1.2.3 From e7636925789b042ff9d98c51d48392e8c5549480 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 26 Jan 2010 17:08:24 +0100 Subject: ALSA: pcm_lib - return back hw_ptr_interrupt Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 + sound/core/oss/pcm_oss.c | 3 +-- sound/core/pcm_lib.c | 7 +++++-- sound/core/pcm_native.c | 2 ++ 4 files changed, 9 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3bc9bca771ec..13bc83ca35fb 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,6 +271,7 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 255ad910077a..82d4e3329b3d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -635,8 +635,7 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) static inline snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t ptr = runtime->status->hw_ptr; - return ptr - (ptr % runtime->period_size); + return runtime->hw_ptr_interrupt; } /* define extended formats in the recent OSS versions (if any) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e2a817eac2a9..aa54195ef3b0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -325,8 +325,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ - delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) - + runtime->period_size; + delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -437,6 +436,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { + runtime->hw_ptr_interrupt = new_hw_ptr - + (new_hw_ptr % runtime->period_size); + } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 56ec35e8510b..7a002db512b4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1252,6 +1252,8 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; + runtime->hw_ptr_interrupt = runtime->status->hw_ptr - + runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.2.3 From 28e1b773083d349d5223f586a39fa30f5d0f1c36 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:09 +0100 Subject: ALSA: usbaudio: parse USB descriptors with structs In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 32 +++++++- sound/usb/usbaudio.c | 198 ++++++++++++++++++++++++++++------------------ sound/usb/usbmixer.c | 37 +++++---- 3 files changed, 168 insertions(+), 99 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index eaf9dffe0a01..44f82d8e09c5 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -81,7 +81,7 @@ /* Terminal Control Selectors */ /* 4.3.2 Class-Specific AC Interface Descriptor */ -struct uac_ac_header_descriptor { +struct uac_ac_header_descriptor_v1 { __u8 bLength; /* 8 + n */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* UAC_MS_HEADER */ @@ -95,7 +95,7 @@ struct uac_ac_header_descriptor { /* As above, but more useful for defining your own descriptors: */ #define DECLARE_UAC_AC_HEADER_DESCRIPTOR(n) \ -struct uac_ac_header_descriptor_##n { \ +struct uac_ac_header_descriptor_v1_##n { \ __u8 bLength; \ __u8 bDescriptorType; \ __u8 bDescriptorSubtype; \ @@ -131,7 +131,7 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 /* 4.3.2.2 Output Terminal Descriptor */ -struct uac_output_terminal_descriptor { +struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ __u8 bDescriptorSubtype; /* OUTPUT_TERMINAL descriptor subtype */ @@ -171,7 +171,7 @@ struct uac_feature_unit_descriptor_##ch { \ } __attribute__ ((packed)) /* 4.5.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor { +struct uac_as_header_descriptor_v1 { __u8 bLength; /* in bytes: 7 */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* AS_GENERAL */ @@ -232,6 +232,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +/* Formats - Audio Data Format Type I Codes */ + +struct uac_format_type_ii_discrete_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __le16 wMaxBitRate; + __le16 wSamplesPerFrame; + __u8 bSamFreqType; + __u8 tSamFreq[][3]; +} __attribute__((packed)); + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 @@ -253,6 +266,17 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_FILL_MAX 0x80 /* A.10.2 Feature Unit Control Selectors */ + +struct uac_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + __u8 bControlSize; + __u8 controls[0]; /* variable length */ +} __attribute__((packed)); + #define UAC_FU_CONTROL_UNDEFINED 0x00 #define UAC_MUTE_CONTROL 0x01 #define UAC_VOLUME_CONTROL 0x02 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c6b9c8cac59e..f833dea60180 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -2421,15 +2423,17 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @fmt: the format type descriptor */ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; int sample_width, sample_bytes; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt[6]; - sample_bytes = fmt[5]; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + switch (format) { case 0: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", @@ -2442,7 +2446,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ - switch (fmt[5]) { + switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; break; @@ -2463,8 +2467,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", - chip->dev->devnum, fp->iface, - fp->altsetting, sample_width, sample_bytes); + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); break; } break; @@ -2564,11 +2568,12 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - if (fmt[3] == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2590,23 +2595,27 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * if (pcm_format < 0) return -1; } + fp->format = pcm_format; - fp->channels = fmt[4]; + fp->channels = fmt->bNrChannels; + if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt, 7); + return parse_audio_format_rates(chip, fp, fmt_raw, 7); } /* - * prase the format type II descriptor + * parse the format type II descriptor */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int brate, framesize; + struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + switch (format) { case USB_AUDIO_FORMAT_AC3: /* FIXME: there is no AC3 format defined yet */ @@ -2622,20 +2631,25 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; } + fp->channels = 1; - brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */ - framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */ + + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */ + return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream) + int format, void *fmt_raw, int stream) { int err; + /* we only parse the common header of all format types here, + * so it is safe to take a type_i struct */ + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt[3]) { + switch (fmt->bFormatType) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt); @@ -2645,10 +2659,10 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -2659,7 +2673,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt->bFormatType == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2708,6 +2722,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { + struct uac_as_header_descriptor_v1 *as; + alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); /* skip invalid one */ @@ -2726,7 +2742,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2734,20 +2750,21 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); - if (!fmt) { + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 7) { + if (as->bLength < sizeof(*as)) { snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } - format = (fmt[6] << 8) | fmt[5]; /* remember the format value */ + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2875,6 +2892,65 @@ static void snd_usb_stream_disconnect(struct list_head *head) } } +static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface = usb_ifnum_to_if(dev, interface); + + if (!iface) { + snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + if (usb_interface_claimed(iface)) { + snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { + snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + + return 0; + } + + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", + dev->devnum, ctrlif, interface, altsd->bInterfaceClass); + /* skip non-supported classes */ + return -EINVAL; + } + + if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { + snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + return -EINVAL; + } + + if (! parse_audio_endpoints(chip, interface)) { + usb_set_interface(dev, interface, 0); /* reset the current interface */ + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return -EINVAL; + } + + return 0; +} + /* * parse audio control descriptor and create pcm/midi streams */ @@ -2882,69 +2958,36 @@ 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 *iface; - unsigned char *p1; - int i, j; + struct uac_ac_header_descriptor_v1 *h1; + void *control_header; + int i; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { + control_header = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, HEADER); + + if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + + h1 = control_header; + + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); return -EINVAL; } - /* - * parse all USB audio streaming interfaces - */ - for (i = 0; i < p1[7]; i++) { - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - j = p1[8 + i]; - iface = usb_ifnum_to_if(dev, j); - if (!iface) { - snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", - dev->devnum, ctrlif, j); - continue; - } - if (usb_interface_claimed(iface)) { - snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j); - continue; - } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || - altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); - if (err < 0) { - snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); - continue; - } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - continue; - } - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { - snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass); - /* skip non-supported classes */ - continue; - } - if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); - continue; - } - if (! parse_audio_endpoints(chip, j)) { - usb_set_interface(dev, j, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - } + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; } + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + return 0; } @@ -3607,7 +3650,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev, ifnum = get_iface_desc(alts)->bInterfaceNumber; id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); - if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) goto __err_val; diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 35b4830fb0c4..11636a6112d5 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include #include @@ -1086,29 +1088,30 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, * * most of controlls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) { int channels, i, j; struct usb_audio_term iterm; unsigned int master_bits, first_ch_bits; int err, csize; + struct uac_feature_unit_descriptor *ftr = _ftr; - if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) { + if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } /* parse the source unit */ - if ((err = parse_audio_unit(state, ftr[4])) < 0) + if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0) return err; /* determine the input source type and name */ - if (check_input_term(state, ftr[4], &iterm) < 0) + if (check_input_term(state, ftr->bSourceID, &iterm) < 0) return -EINVAL; - channels = (ftr[0] - 7) / csize - 1; + channels = (ftr->bLength - 7) / csize - 1; - master_bits = snd_usb_combine_bytes(ftr + 6, csize); + master_bits = snd_usb_combine_bytes(ftr->controls, csize); /* master configuration quirks */ switch (state->chip->usb_id) { case USB_ID(0x08bb, 0x2702): @@ -1119,21 +1122,21 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig break; } if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); + first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize); else first_ch_bits = 0; /* check all control types */ for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize); + unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid); + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid); if (master_bits & (1 << i)) - build_feature_ctl(state, ftr, 0, i, &iterm, unitid); + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid); } return 0; @@ -1780,7 +1783,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { - unsigned char *desc; + struct uac_output_terminal_descriptor_v1 *desc; struct mixer_build state; int err; const struct usbmix_ctl_map *map; @@ -1805,13 +1808,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) desc = NULL; while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { - if (desc[0] < 9) + if (desc->bLength < 9) continue; /* invalid descriptor? */ - set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */ - state.oterm.id = desc[3]; - state.oterm.type = combine_word(&desc[4]); - state.oterm.name = desc[8]; - err = parse_audio_unit(&state, desc[7]); + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; } -- cgit v1.2.3 From 8fee4aff8c89c229593b76a6ab172a9cad24b412 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:10 +0100 Subject: ALSA: usbaudio: introduce new types for audio class v2 This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 57 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 19 +++++++++++++--- sound/usb/usbmixer.c | 14 ++++++------ 3 files changed, 80 insertions(+), 10 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 44f82d8e09c5..fb1a97bf943d 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -25,6 +25,9 @@ #define USB_SUBCLASS_AUDIOSTREAMING 0x02 #define USB_SUBCLASS_MIDISTREAMING 0x03 +#define UAC_VERSION_1 0x00 +#define UAC_VERSION_2 0x20 + /* A.5 Audio Class-Specific AC Interface Descriptor Subtypes */ #define UAC_HEADER 0x01 #define UAC_INPUT_TERMINAL 0x02 @@ -180,6 +183,19 @@ struct uac_as_header_descriptor_v1 { __le16 wFormatTag; /* The Audio Data Format */ } __attribute__ ((packed)); +struct uac_as_header_descriptor_v2 { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __u8 bmControls; + __u8 bFormatType; + __u32 bmFormats; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; +} __attribute__((packed)); + #define UAC_DT_AS_HEADER_SIZE 7 /* Formats - A.1.1 Audio Data Format Type I Codes */ @@ -232,6 +248,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +struct uac_format_type_i_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bSubslotSize; + __u8 bFormatType; + __u8 bBitResolution; + __u8 bHeaderLength; + __u8 bControlSize; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - Audio Data Format Type I Codes */ struct uac_format_type_ii_discrete_descriptor { @@ -245,11 +274,26 @@ struct uac_format_type_ii_discrete_descriptor { __u8 tSamFreq[][3]; } __attribute__((packed)); +struct uac_format_type_ii_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __u16 wMaxBitRate; + __u16 wSamplesPerFrame; + __u8 bHeaderLength; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 #define UAC_FORMAT_TYPE_II 0x2 #define UAC_FORMAT_TYPE_III 0x3 +#define UAC_EXT_FORMAT_TYPE_I 0x81 +#define UAC_EXT_FORMAT_TYPE_II 0x82 +#define UAC_EXT_FORMAT_TYPE_III 0x83 struct uac_iso_endpoint_descriptor { __u8 bLength; /* in bytes: 7 */ @@ -265,6 +309,19 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_PITCH_CONTROL 0x02 #define UAC_EP_CS_ATTR_FILL_MAX 0x80 +/* Audio class v2.0: CLOCK_SOURCE descriptor */ + +struct uac_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __u8 bmControls; + __u8 bAssocTerminal; + __u8 iClockSource; +} __attribute__((packed)); + /* A.10.2 Feature Unit Control Selectors */ struct uac_feature_unit_descriptor { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea48fc5f..4f482939e8e8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,8 +36,17 @@ #define MIXER_UNIT 0x04 #define SELECTOR_UNIT 0x05 #define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT 0x07 -#define EXTENSION_UNIT 0x08 +#define PROCESSING_UNIT_V1 0x07 +#define EXTENSION_UNIT_V1 0x08 + +/* audio class v2 */ +#define EFFECT_UNIT 0x07 +#define PROCESSING_UNIT_V2 0x08 +#define EXTENSION_UNIT_V2 0x09 +#define CLOCK_SOURCE 0x0a +#define CLOCK_SELECTOR 0x0b +#define CLOCK_MULTIPLIER 0x0c +#define SAMPLE_RATE_CONVERTER 0x0d #define AS_GENERAL 0x01 #define FORMAT_TYPE 0x02 @@ -60,7 +69,7 @@ #define EP_CS_ATTR_PITCH_CONTROL 0x02 #define EP_CS_ATTR_FILL_MAX 0x80 -/* Audio Class specific Request Codes */ +/* Audio Class specific Request Codes (v1) */ #define SET_CUR 0x01 #define GET_CUR 0x81 @@ -74,6 +83,10 @@ #define GET_MEM 0x85 #define GET_STAT 0xff +/* Audio Class specific Request Codes (v2) */ +#define CS_CUR 0x01 +#define CS_RANGE 0x02 + /* Terminal Control Selectors */ #define COPY_PROTECT_CONTROL 0x01 diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 11636a6112d5..ca7949598191 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit) + if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -607,9 +607,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; case MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT: - case EXTENSION_UNIT: + case PROCESSING_UNIT_V1: + case EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -1747,9 +1747,9 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); -- cgit v1.2.3 From de48c7bc6f93c6c8e0be8612c9d72a2dc92eaa01 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:13 +0100 Subject: ALSA: usbaudio: consolidate header files Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 31 +++++++++++- sound/usb/usbaudio.c | 125 +++++++++++++++++++++++----------------------- sound/usb/usbaudio.h | 100 ------------------------------------- sound/usb/usbmidi.c | 10 ++-- sound/usb/usbmixer.c | 62 +++++++++++------------ sound/usb/usbquirks.h | 34 ++++++------- sound/usb/usx2y/us122l.c | 6 ++- 7 files changed, 150 insertions(+), 218 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index fb1a97bf943d..6bb293684eb8 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -35,8 +35,17 @@ #define UAC_MIXER_UNIT 0x04 #define UAC_SELECTOR_UNIT 0x05 #define UAC_FEATURE_UNIT 0x06 -#define UAC_PROCESSING_UNIT 0x07 -#define UAC_EXTENSION_UNIT 0x08 +#define UAC_PROCESSING_UNIT_V1 0x07 +#define UAC_EXTENSION_UNIT_V1 0x08 + +/* UAC v2.0 types */ +#define UAC_EFFECT_UNIT 0x07 +#define UAC_PROCESSING_UNIT_V2 0x08 +#define UAC_EXTENSION_UNIT_V2 0x09 +#define UAC_CLOCK_SOURCE 0x0a +#define UAC_CLOCK_SELECTOR 0x0b +#define UAC_CLOCK_MULTIPLIER 0x0c +#define UAC_SAMPLE_RATE_CONVERTER 0x0d /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 @@ -69,6 +78,10 @@ #define UAC_GET_STAT 0xff +/* Audio class v2.0 handles all the parameter calls differently */ +#define UAC2_CS_CUR 0x01 +#define UAC2_CS_RANGE 0x02 + /* MIDI - A.1 MS Class-Specific Interface Descriptor Subtypes */ #define UAC_MS_HEADER 0x01 #define UAC_MIDI_IN_JACK 0x02 @@ -133,6 +146,10 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_MICROPHONE_ARRAY 0x205 #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 +/* Terminals - control selectors */ + +#define UAC_TERMINAL_CS_COPY_PROTECT_CONTROL 0x01 + /* 4.3.2.2 Output Terminal Descriptor */ struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ @@ -263,6 +280,9 @@ struct uac_format_type_i_ext_descriptor { /* Formats - Audio Data Format Type I Codes */ +#define UAC_FORMAT_TYPE_II_MPEG 0x1001 +#define UAC_FORMAT_TYPE_II_AC3 0x1002 + struct uac_format_type_ii_discrete_descriptor { __u8 bLength; __u8 bDescriptorType; @@ -285,6 +305,13 @@ struct uac_format_type_ii_ext_descriptor { __u8 bSideBandProtocol; } __attribute__((packed)); +/* type III */ +#define UAC_FORMAT_TYPE_III_IEC1937_AC3 0x2001 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG1_LAYER1 0x2002 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_NOEXT 0x2003 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_EXT 0x2004 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER1_LS 0x2005 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER23_LS 0x2006 /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 411a6cf43c21..c539f7fe292f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include @@ -598,7 +599,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; - if (subs->fmt_type == USB_FORMAT_TYPE_II) { + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ @@ -1106,7 +1107,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri u->packets = (i + 1) * total_packs / subs->nurbs - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == USB_FORMAT_TYPE_II) + if (subs->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) @@ -1182,7 +1183,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned if (i >= fp->nr_rates) continue; } - attr = fp->ep_attr & EP_ATTR_MASK; + attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; if (! found) { found = fp; cur_attr = attr; @@ -1194,14 +1195,14 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned * M-audio audiophile USB. */ if (attr != cur_attr) { - if ((attr == EP_ATTR_ASYNC && + if ((attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (attr == EP_ATTR_ADAPTIVE && + (attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) continue; - if ((cur_attr == EP_ATTR_ASYNC && + if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (cur_attr == EP_ATTR_ADAPTIVE && + (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { found = fp; cur_attr = attr; @@ -1231,11 +1232,11 @@ static int init_usb_pitch(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has pitch control, enable it */ - if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) { + if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) { data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -1254,21 +1255,21 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has sampling rate control, set it */ - if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) { + if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) { int crate; data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -1386,9 +1387,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * descriptors which fool us. if it has only one EP, * assume it as adaptive-out or sync-in. */ - attr = fmt->ep_attr & EP_ATTR_MASK; - if (((is_playback && attr == EP_ATTR_ASYNC) || - (! is_playback && attr == EP_ATTR_ADAPTIVE)) && + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -1428,7 +1429,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } /* always fill max packet size */ - if (fmt->attributes & EP_CS_ATTR_FILL_MAX) + if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) subs->fill_max = 1; if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0) @@ -1886,7 +1887,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = fp->channels; if (runtime->hw.channels_max < fp->channels) runtime->hw.channels_max = fp->channels; - if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) { + if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { /* FIXME: there might be more than one audio formats... */ runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; @@ -2120,7 +2121,7 @@ static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { } /* Terminating entry */ }; @@ -2159,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, fp->endpoint & USB_DIR_IN ? "IN" : "OUT", - sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]); + sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", fp->rate_min, fp->rate_max); @@ -2471,11 +2472,11 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, pcm_format = -1; switch (format) { - case 0: /* some devices don't define this correctly... */ + case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); /* fall-through */ - case USB_AUDIO_FORMAT_PCM: + case UAC_FORMAT_TYPE_I_PCM: if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -2509,7 +2510,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, break; } break; - case USB_AUDIO_FORMAT_PCM8: + case UAC_FORMAT_TYPE_I_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; /* Dallas DS4201 workaround: it advertises U8 format, but really @@ -2517,13 +2518,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x04fa, 0x4201)) pcm_format = SNDRV_PCM_FORMAT_S8; break; - case USB_AUDIO_FORMAT_IEEE_FLOAT: + case UAC_FORMAT_TYPE_I_IEEE_FLOAT: pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; break; - case USB_AUDIO_FORMAT_ALAW: + case UAC_FORMAT_TYPE_I_ALAW: pcm_format = SNDRV_PCM_FORMAT_A_LAW; break; - case USB_AUDIO_FORMAT_MU_LAW: + case UAC_FORMAT_TYPE_I_MULAW: pcm_format = SNDRV_PCM_FORMAT_MU_LAW; break; default: @@ -2551,7 +2552,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int nr_rates = fmt[offset]; if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", chip->dev->devnum, fp->iface, fp->altsetting); return -1; } @@ -2614,7 +2615,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, int i, nr_rates, data_size, ret = 0; /* get the number of sample rates first by only fetching 2 bytes */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); @@ -2632,7 +2633,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, } /* now get the full information */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, data, data_size, 1000); @@ -2682,7 +2683,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; - if (fmt->bFormatType == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2745,12 +2746,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; switch (format) { - case USB_AUDIO_FORMAT_AC3: + case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ // fp->format = SNDRV_PCM_FORMAT_AC3; fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ break; - case USB_AUDIO_FORMAT_MPEG: + case UAC_FORMAT_TYPE_II_MPEG: fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: @@ -2793,11 +2794,11 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp int err; switch (fmt[3]) { - case USB_FORMAT_TYPE_I: - case USB_FORMAT_TYPE_III: + case UAC_FORMAT_TYPE_I: + case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); break; - case USB_FORMAT_TYPE_II: + case UAC_FORMAT_TYPE_II: err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: @@ -2816,7 +2817,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt[3] == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2871,7 +2872,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING && + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) @@ -2895,16 +2896,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) switch (protocol) { case UAC_VERSION_1: { struct uac_as_header_descriptor_v1 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2915,16 +2916,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac_as_header_descriptor_v2 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2942,15 +2943,15 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } @@ -2972,7 +2973,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Creamware Noah has this descriptor after the 2nd endpoint */ if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { + if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); @@ -3006,12 +3007,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Optoplay sets the sample rate attribute although * it seems not supporting it in fact. */ - fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE; + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= EP_CS_ATTR_SAMPLE_RATE; + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is @@ -3020,11 +3021,11 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * plantronics headset and Griffin iMic have set adaptive-in * although it's really not... */ - fp->ep_attr &= ~EP_ATTR_MASK; + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= EP_ATTR_ADAPTIVE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; else - fp->ep_attr |= EP_ATTR_SYNC; + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; } @@ -3094,7 +3095,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { int err = snd_usbmidi_create(chip->card, iface, &chip->midi_list, NULL); if (err < 0) { @@ -3109,7 +3110,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, interface, altsd->bInterfaceClass); /* skip non-supported classes */ @@ -3145,12 +3146,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, HEADER); + NULL, UAC_HEADER); altsd = get_iface_desc(host_iface); protocol = altsd->bInterfaceProtocol; if (!control_header) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); return -EINVAL; } @@ -3164,7 +3165,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); return -EINVAL; } @@ -3190,7 +3191,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * clock selectors and sample rate conversion units. */ cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, CLOCK_SOURCE); + NULL, UAC_CLOCK_SOURCE); if (!cs) { snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); @@ -3302,7 +3303,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -3394,7 +3395,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, { static const struct audioformat ua1000_format = { .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .attributes = 0, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 26daf68631eb..6b016d4aac6b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,106 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - -/* - */ - -#define USB_SUBCLASS_AUDIO_CONTROL 0x01 -#define USB_SUBCLASS_AUDIO_STREAMING 0x02 -#define USB_SUBCLASS_MIDI_STREAMING 0x03 -#define USB_SUBCLASS_VENDOR_SPEC 0xff - -#define HEADER 0x01 -#define INPUT_TERMINAL 0x02 -#define OUTPUT_TERMINAL 0x03 -#define MIXER_UNIT 0x04 -#define SELECTOR_UNIT 0x05 -#define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT_V1 0x07 -#define EXTENSION_UNIT_V1 0x08 - -/* audio class v2 */ -#define EFFECT_UNIT 0x07 -#define PROCESSING_UNIT_V2 0x08 -#define EXTENSION_UNIT_V2 0x09 -#define CLOCK_SOURCE 0x0a -#define CLOCK_SELECTOR 0x0b -#define CLOCK_MULTIPLIER 0x0c -#define SAMPLE_RATE_CONVERTER 0x0d - -#define AS_GENERAL 0x01 -#define FORMAT_TYPE 0x02 -#define FORMAT_SPECIFIC 0x03 - -#define EP_GENERAL 0x01 - -#define MS_GENERAL 0x01 -#define MIDI_IN_JACK 0x02 -#define MIDI_OUT_JACK 0x03 - -/* endpoint attributes */ -#define EP_ATTR_MASK 0x0c -#define EP_ATTR_ASYNC 0x04 -#define EP_ATTR_ADAPTIVE 0x08 -#define EP_ATTR_SYNC 0x0c - -/* cs endpoint attributes */ -#define EP_CS_ATTR_SAMPLE_RATE 0x01 -#define EP_CS_ATTR_PITCH_CONTROL 0x02 -#define EP_CS_ATTR_FILL_MAX 0x80 - -/* Audio Class specific Request Codes (v1) */ - -#define SET_CUR 0x01 -#define GET_CUR 0x81 -#define SET_MIN 0x02 -#define GET_MIN 0x82 -#define SET_MAX 0x03 -#define GET_MAX 0x83 -#define SET_RES 0x04 -#define GET_RES 0x84 -#define SET_MEM 0x05 -#define GET_MEM 0x85 -#define GET_STAT 0xff - -/* Audio Class specific Request Codes (v2) */ -#define CS_CUR 0x01 -#define CS_RANGE 0x02 - -/* Terminal Control Selectors */ - -#define COPY_PROTECT_CONTROL 0x01 - -/* Endpoint Control Selectors */ - -#define SAMPLING_FREQ_CONTROL 0x01 -#define PITCH_CONTROL 0x02 - -/* Format Types */ -#define USB_FORMAT_TYPE_I 0x01 -#define USB_FORMAT_TYPE_II 0x02 -#define USB_FORMAT_TYPE_III 0x03 - -/* type I */ -#define USB_AUDIO_FORMAT_PCM 0x01 -#define USB_AUDIO_FORMAT_PCM8 0x02 -#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03 -#define USB_AUDIO_FORMAT_ALAW 0x04 -#define USB_AUDIO_FORMAT_MU_LAW 0x05 - -/* type II */ -#define USB_AUDIO_FORMAT_MPEG 0x1001 -#define USB_AUDIO_FORMAT_AC3 0x1002 - -/* type III */ -#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001 -#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006 - - /* maximum number of endpoints per interface */ #define MIDI_MAX_ENDPOINTS 2 diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index b2da478a0fae..2c59afd99611 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -1540,7 +1542,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostif->extralen >= 7 && ms_header->bLength >= 7 && ms_header->bDescriptorType == USB_DT_CS_INTERFACE && - ms_header->bDescriptorSubtype == HEADER) + ms_header->bDescriptorSubtype == UAC_HEADER) snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n", ms_header->bcdMSC[1], ms_header->bcdMSC[0]); else @@ -1556,7 +1558,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostep->extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != MS_GENERAL) + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -1768,9 +1770,9 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; cs_desc += cs_desc[0]) { if (cs_desc[1] == USB_DT_CS_INTERFACE) { - if (cs_desc[2] == MIDI_IN_JACK) + if (cs_desc[2] == UAC_MIDI_IN_JACK) endpoint->in_cables = (endpoint->in_cables << 1) | 1; - else if (cs_desc[2] == MIDI_OUT_JACK) + else if (cs_desc[2] == UAC_MIDI_OUT_JACK) endpoint->out_cables = (endpoint->out_cables << 1) | 1; } } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 42bb95c739a8..8e8f871b74ca 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) + if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -407,14 +407,14 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) { - return get_ctl_value(cval, GET_CUR, validx, value); + return get_ctl_value(cval, UAC_GET_CUR, validx, value); } /* channel = 0: master, 1 = first channel */ static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -468,14 +468,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) { - return set_ctl_value(cval, SET_CUR, validx, value); + return set_ctl_value(cval, UAC_SET_CUR, validx, value); } static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; - err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) return err; @@ -605,13 +605,13 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm if (term_only) return 0; switch (iterm->type >> 16) { - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; - case MIXER_UNIT: + case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: return sprintf(name, "Unit %d", iterm->id); @@ -650,22 +650,22 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ while ((p1 = find_audio_control_unit(state, id)) != NULL) { term->id = id; switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: term->type = combine_word(p1 + 4); term->channels = p1[7]; term->chconfig = combine_word(p1 + 8); term->name = p1[11]; return 0; - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: id = p1[4]; break; /* continue to parse */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: term->type = p1[2] << 16; /* virtual type */ term->channels = p1[5 + p1[4]]; term->chconfig = combine_word(p1 + 6 + p1[4]); term->name = p1[p1[0] - 1]; return 0; - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: /* call recursively to retrieve the channel info */ if (check_input_term(state, p1[5], term) < 0) return -ENODEV; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT_V1: - case EXTENSION_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -752,23 +752,23 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; } } - if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || - get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { + if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || + get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->id, cval->mixer->ctrlif, cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { - if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) + if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1097,7 +1097,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void struct uac_feature_unit_descriptor *ftr = _ftr; if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); + snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } @@ -1739,17 +1739,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: return 0; /* NOP */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: return parse_audio_selector_unit(state, unitid, p1); - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); @@ -1779,7 +1779,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) /* * create mixer controls * - * walk through all OUTPUT_TERMINAL descriptors to search for mixers + * walk through all UAC_OUTPUT_TERMINAL descriptors to search for mixers */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { @@ -1807,7 +1807,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } desc = NULL; - while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { + while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) { if (desc->bLength < 9) continue; /* invalid descriptor? */ set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ @@ -2047,7 +2047,7 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) } mixer->rc_setup_packet->bRequestType = USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; - mixer->rc_setup_packet->bRequest = GET_MEM; + mixer->rc_setup_packet->bRequest = UAC_GET_MEM; mixer->rc_setup_packet->wValue = cpu_to_le16(0); mixer->rc_setup_packet->wIndex = cpu_to_le16(0); mixer->rc_setup_packet->wLength = cpu_to_le16(len); @@ -2170,7 +2170,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s: ", jacks[i].name); err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), - GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | + UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3, 100); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index fc1d2cd6ccc3..f06faf7917b9 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -91,7 +91,7 @@ .idVendor = 0x046d, .idProduct = 0x0850, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -100,7 +100,7 @@ .idVendor = 0x046d, .idProduct = 0x08ae, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -109,7 +109,7 @@ .idVendor = 0x046d, .idProduct = 0x08c6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -118,7 +118,7 @@ .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -127,7 +127,7 @@ .idVendor = 0x046d, .idProduct = 0x08f5, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -136,7 +136,7 @@ .idVendor = 0x046d, .idProduct = 0x08f6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { USB_DEVICE(0x046d, 0x0990), @@ -301,7 +301,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_FILL_MAX, + .attributes = UAC_EP_CS_ATTR_FILL_MAX, .endpoint = 0x81, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -2108,7 +2108,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2122,7 +2122,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2136,7 +2136,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2150,7 +2150,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2164,7 +2164,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2178,7 +2178,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2192,7 +2192,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2206,7 +2206,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-850", @@ -2238,7 +2238,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_SAMPLE_RATE, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x02, .ep_attr = 0x01, .maxpacksize = 0x130, @@ -2268,7 +2268,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .match_flags = USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING, + .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_MIDI_STANDARD_INTERFACE diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb29666d26..44deb21b1777 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,8 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include +#include #include #include #include @@ -315,9 +317,9 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate) data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000); if (err < 0) snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n", dev->devnum, rate, ep); -- cgit v1.2.3