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-rw-r--r--sound/soc/fsl/Kconfig32
-rw-r--r--sound/soc/fsl/Makefile7
-rw-r--r--sound/soc/fsl/efika-audio-fabric.c90
-rw-r--r--sound/soc/fsl/fsl_ssi.c11
-rw-r--r--sound/soc/fsl/mpc5200_dma.c564
-rw-r--r--sound/soc/fsl/mpc5200_dma.h80
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c329
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.h15
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c754
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.h12
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c90
11 files changed, 1277 insertions, 707 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9fc908283371..5dbebf82249c 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,8 @@
config SND_SOC_OF_SIMPLE
tristate
+
+config SND_MPC52xx_DMA
+ tristate
# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers
# for the SSI and the Elo DMA controller. You will still need to select
@@ -22,7 +25,34 @@ config SND_SOC_MPC8610_HPCD
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
- select SND_SOC_OF_SIMPLE
+ select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+ tristate "Freescale MPC5200 PSC in AC97 mode driver"
+ depends on PPC_MPC52xx && PPC_BESTCOMM
+ select AC97_BUS
+ select SND_MPC52xx_DMA
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+config SND_MPC52xx_SOC_PCM030
+ tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
+ depends on PPC_MPC5200_SIMPLE && BROKEN
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for sound on the Phytec pcm030
+ baseboard.
+
+config SND_MPC52xx_SOC_EFIKA
+ tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
+ depends on PPC_EFIKA && BROKEN
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_STAC9766
+ help
+ Say Y if you want to add support for sound on the Efika.
+
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f85134c86387..a83a73967ec6 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,5 +10,12 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
+
+# MPC5200 Machine Support
+obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
+obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
new file mode 100644
index 000000000000..85b0e7569504
--- /dev/null
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Efika driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/stac9766.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link efika_fabric_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 Analog",
+ .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+ .name = "AC97",
+ .stream_name = "AC97 IEC958",
+ .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int efika_fabric_init(void)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ if (!machine_is_compatible("bplan,efika"))
+ return -ENODEV;
+
+ card.platform = &mpc5200_audio_dma_platform;
+ card.name = "Efika";
+ card.dai_link = efika_fabric_dai;
+ card.num_links = ARRAY_SIZE(efika_fabric_dai);
+
+ device.card = &card;
+ device.codec_dev = &soc_codec_dev_stac9766;
+
+ pdev = platform_device_alloc("soc-audio", 1);
+ if (!pdev) {
+ pr_err("efika_fabric_init: platform_device_alloc() failed\n");
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(pdev, &device);
+ device.dev = &pdev->dev;
+
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("efika_fabric_init: platform_device_add() failed\n");
+ return -ENODEV;
+ }
+ return 0;
+}
+
+module_init(efika_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 3711d8454d96..93f0f38a32c9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *first_runtime =
ssi_private->first_stream->runtime;
- if (!first_runtime->rate || !first_runtime->sample_bits) {
+ if (!first_runtime->sample_bits) {
dev_err(substream->pcm->card->dev,
- "set sample rate and size in %s stream first\n",
+ "set sample size in %s stream first\n",
substream->stream == SNDRV_PCM_STREAM_PLAYBACK
? "capture" : "playback");
return -EAGAIN;
}
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- first_runtime->rate, first_runtime->rate);
-
/* If we're in synchronous mode, then we need to constrain
* the sample size as well. We don't support independent sample
* rates in asynchronous mode.
@@ -674,7 +670,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
ssi_private->dev = ssi_info->dev;
ssi_private->asynchronous = ssi_info->asynchronous;
- ssi_private->dev->driver_data = fsl_ssi_dai;
+ dev_set_drvdata(ssi_private->dev, fsl_ssi_dai);
/* Initialize the the device_attribute structure */
dev_attr->attr.name = "ssi-stats";
@@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
fsl_ssi_dai->name = ssi_private->name;
fsl_ssi_dai->id = ssi_info->id;
fsl_ssi_dai->dev = ssi_info->dev;
+ fsl_ssi_dai->symmetric_rates = 1;
ret = snd_soc_register_dai(fsl_ssi_dai);
if (ret != 0) {
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 000000000000..efec33a1c5bd
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,564 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+
+#include <sound/soc.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
+{
+ struct psc_dma *psc_dma = _psc_dma;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 isr;
+
+ isr = in_be16(&regs->mpc52xx_psc_isr);
+
+ /* Playback underrun error */
+ if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+ psc_dma->stats.underrun_count++;
+
+ /* Capture overrun error */
+ if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+ psc_dma->stats.overrun_count++;
+
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue. Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
+{
+ struct bcom_bd *bd;
+
+ /* Prepare and enqueue the next buffer descriptor */
+ bd = bcom_prepare_next_buffer(s->bcom_task);
+ bd->status = s->period_bytes;
+ bd->data[0] = s->period_next_pt;
+ bcom_submit_next_buffer(s->bcom_task, NULL);
+
+ /* Update for next period */
+ s->period_next_pt += s->period_bytes;
+ if (s->period_next_pt >= s->period_end)
+ s->period_next_pt = s->period_start;
+}
+
+static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
+{
+ while (s->appl_ptr < s->runtime->control->appl_ptr) {
+
+ if (bcom_queue_full(s->bcom_task))
+ return;
+
+ s->appl_ptr += s->period_size;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
+{
+ struct psc_dma_stream *s = _psc_dma_stream;
+
+ spin_lock(&s->psc_dma->lock);
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ s->period_current_pt += s->period_bytes;
+ if (s->period_current_pt >= s->period_end)
+ s->period_current_pt = s->period_start;
+ }
+ psc_dma_bcom_enqueue_tx(s);
+ spin_unlock(&s->psc_dma->lock);
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream)
+{
+ struct psc_dma_stream *s = _psc_dma_stream;
+
+ spin_lock(&s->psc_dma->lock);
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ s->period_current_pt += s->period_bytes;
+ if (s->period_current_pt >= s->period_end)
+ s->period_current_pt = s->period_start;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+ spin_unlock(&s->psc_dma->lock);
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+static int psc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+/**
+ * psc_dma_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct psc_dma_stream *s;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 imr;
+ unsigned long flags;
+ int i;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
+ " stream_id=%i\n",
+ substream, cmd, substream->pstr->stream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ s->period_bytes = frames_to_bytes(runtime,
+ runtime->period_size);
+ s->period_start = virt_to_phys(runtime->dma_area);
+ s->period_end = s->period_start +
+ (s->period_bytes * runtime->periods);
+ s->period_next_pt = s->period_start;
+ s->period_current_pt = s->period_start;
+ s->period_size = runtime->period_size;
+ s->active = 1;
+
+ /* track appl_ptr so that we have a better chance of detecting
+ * end of stream and not over running it.
+ */
+ s->runtime = runtime;
+ s->appl_ptr = s->runtime->control->appl_ptr -
+ (runtime->period_size * runtime->periods);
+
+ /* Fill up the bestcomm bd queue and enable DMA.
+ * This will begin filling the PSC's fifo.
+ */
+ spin_lock_irqsave(&psc_dma->lock, flags);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ for (i = 0; i < runtime->periods; i++)
+ if (!bcom_queue_full(s->bcom_task))
+ psc_dma_bcom_enqueue_next_buffer(s);
+ } else {
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ psc_dma_bcom_enqueue_tx(s);
+ }
+
+ bcom_enable(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ s->active = 0;
+
+ spin_lock_irqsave(&psc_dma->lock, flags);
+ bcom_disable(s->bcom_task);
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ else
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ break;
+
+ default:
+ dev_dbg(psc_dma->dev, "invalid command\n");
+ return -EINVAL;
+ }
+
+ /* Update interrupt enable settings */
+ imr = 0;
+ if (psc_dma->playback.active)
+ imr |= MPC52xx_PSC_IMR_TXEMP;
+ if (psc_dma->capture.active)
+ imr |= MPC52xx_PSC_IMR_ORERR;
+ out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);
+
+ return 0;
+}
+
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_dma_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .period_bytes_max = 1024 * 1024,
+ .period_bytes_min = 32,
+ .periods_min = 2,
+ .periods_max = 256,
+ .buffer_bytes_max = 2 * 1024 * 1024,
+ .fifo_size = 512,
+};
+
+static int psc_dma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+ int rc;
+
+ dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
+
+ rc = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (rc < 0) {
+ dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+ return rc;
+ }
+
+ s->stream = substream;
+ return 0;
+}
+
+static int psc_dma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+
+ dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ if (!psc_dma->playback.active &&
+ !psc_dma->capture.active) {
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
+ }
+ s->stream = NULL;
+ return 0;
+}
+
+static snd_pcm_uframes_t
+psc_dma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+ dma_addr_t count;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ count = s->period_current_pt - s->period_start;
+
+ return bytes_to_frames(substream->runtime, count);
+}
+
+static int
+psc_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static struct snd_pcm_ops psc_dma_ops = {
+ .open = psc_dma_open,
+ .close = psc_dma_close,
+ .hw_free = psc_dma_hw_free,
+ .ioctl = snd_pcm_lib_ioctl,
+ .pointer = psc_dma_pointer,
+ .trigger = psc_dma_trigger,
+ .hw_params = psc_dma_hw_params,
+};
+
+static u64 psc_dma_dmamask = 0xffffffff;
+static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ size_t size = psc_dma_hardware.buffer_bytes_max;
+ int rc = 0;
+
+ dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
+ card, dai, pcm);
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &psc_dma_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (pcm->streams[0].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[0].substream->dma_buffer);
+ if (rc)
+ goto playback_alloc_err;
+ }
+
+ if (pcm->streams[1].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[1].substream->dma_buffer);
+ if (rc)
+ goto capture_alloc_err;
+ }
+
+ if (rtd->socdev->card->codec->ac97)
+ rtd->socdev->card->codec->ac97->private_data = psc_dma;
+
+ return 0;
+
+ capture_alloc_err:
+ if (pcm->streams[0].substream)
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+
+ playback_alloc_err:
+ dev_err(card->dev, "Cannot allocate buffer(s)\n");
+
+ return -ENOMEM;
+}
+
+static void psc_dma_free(struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_pcm_substream *substream;
+ int stream;
+
+ dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+struct snd_soc_platform mpc5200_audio_dma_platform = {
+ .name = "mpc5200-psc-audio",
+ .pcm_ops = &psc_dma_ops,
+ .pcm_new = &psc_dma_new,
+ .pcm_free = &psc_dma_free,
+};
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform);
+
+int mpc5200_audio_dma_create(struct of_device *op)
+{
+ phys_addr_t fifo;
+ struct psc_dma *psc_dma;
+ struct resource res;
+ int size, irq, rc;
+ const __be32 *prop;
+ void __iomem *regs;
+
+ /* Fetch the registers and IRQ of the PSC */
+ irq = irq_of_parse_and_map(op->node, 0);
+ if (of_address_to_resource(op->node, 0, &res)) {
+ dev_err(&op->dev, "Missing reg property\n");
+ return -ENODEV;
+ }
+ regs = ioremap(res.start, 1 + res.end - res.start);
+ if (!regs) {
+ dev_err(&op->dev, "Could not map registers\n");
+ return -ENODEV;
+ }
+
+ /* Allocate and initialize the driver private data */
+ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+ if (!psc_dma) {
+ iounmap(regs);
+ return -ENOMEM;
+ }
+
+ /* Get the PSC ID */
+ prop = of_get_property(op->node, "cell-index", &size);
+ if (!prop || size < sizeof *prop)
+ return -ENODEV;
+
+ spin_lock_init(&psc_dma->lock);
+ psc_dma->id = be32_to_cpu(*prop);
+ psc_dma->irq = irq;
+ psc_dma->psc_regs = regs;
+ psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+ psc_dma->dev = &op->dev;
+ psc_dma->playback.psc_dma = psc_dma;
+ psc_dma->capture.psc_dma = psc_dma;
+ snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
+
+ /* Find the address of the fifo data registers and setup the
+ * DMA tasks */
+ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+ psc_dma->capture.bcom_task =
+ bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
+ psc_dma->playback.bcom_task =
+ bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
+ if (!psc_dma->capture.bcom_task ||
+ !psc_dma->playback.bcom_task) {
+ dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+ iounmap(regs);
+ kfree(psc_dma);
+ return -ENODEV;
+ }
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ /* reset receiver */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
+ /* reset transmitter */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
+ /* reset error */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
+ /* reset mode */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
+
+ /* Set up mode register;
+ * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+ * Second write: register Normal mode for non loopback
+ */
+ out_8(&psc_dma->psc_regs->mode, 0);
+ out_8(&psc_dma->psc_regs->mode, 0);
+
+ /* Set the TX and RX fifo alarm thresholds */
+ out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+ out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
+
+ /* Lookup the IRQ numbers */
+ psc_dma->playback.irq =
+ bcom_get_task_irq(psc_dma->playback.bcom_task);
+ psc_dma->capture.irq =
+ bcom_get_task_irq(psc_dma->capture.bcom_task);
+
+ rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+ "psc-dma-status", psc_dma);
+ rc |= request_irq(psc_dma->capture.irq,
+ &psc_dma_bcom_irq_rx, IRQF_SHARED,
+ "psc-dma-capture", &psc_dma->capture);
+ rc |= request_irq(psc_dma->playback.irq,
+ &psc_dma_bcom_irq_tx, IRQF_SHARED,
+ "psc-dma-playback", &psc_dma->playback);
+ if (rc) {
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq,
+ &psc_dma->capture);
+ free_irq(psc_dma->playback.irq,
+ &psc_dma->playback);
+ return -ENODEV;
+ }
+
+ /* Save what we've done so it can be found again later */
+ dev_set_drvdata(&op->dev, psc_dma);
+
+ /* Tell the ASoC OF helpers about it */
+ return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
+
+int mpc5200_audio_dma_destroy(struct of_device *op)
+{
+ struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
+
+ dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
+
+ snd_soc_unregister_platform(&mpc5200_audio_dma_platform);
+
+ bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+ bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
+
+ /* Release irqs */
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+
+ iounmap(psc_dma->psc_regs);
+ kfree(psc_dma);
+ dev_set_drvdata(&op->dev, NULL);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 000000000000..2000803f06a7
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,80 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+#define PSC_STREAM_NAME_LEN 32
+
+/**
+ * psc_ac97_stream - Data specific to a single stream (playback or capture)
+ * @active: flag indicating if the stream is active
+ * @psc_dma: pointer back to parent psc_dma data structure
+ * @bcom_task: bestcomm task structure
+ * @irq: irq number for bestcomm task
+ * @period_start: physical address of start of DMA region
+ * @period_end: physical address of end of DMA region
+ * @period_next_pt: physical address of next DMA buffer to enqueue
+ * @period_bytes: size of DMA period in bytes
+ */
+struct psc_dma_stream {
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_uframes_t appl_ptr;
+
+ int active;
+ struct psc_dma *psc_dma;
+ struct bcom_task *bcom_task;
+ int irq;
+ struct snd_pcm_substream *stream;
+ dma_addr_t period_start;
+ dma_addr_t period_end;
+ dma_addr_t period_next_pt;
+ dma_addr_t period_current_pt;
+ int period_bytes;
+ int period_size;
+};
+
+/**
+ * psc_dma - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ * with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_dma {
+ char name[32];
+ struct mpc52xx_psc __iomem *psc_regs;
+ struct mpc52xx_psc_fifo __iomem *fifo_regs;
+ unsigned int irq;
+ struct device *dev;
+ spinlock_t lock;
+ u32 sicr;
+ uint sysclk;
+ int imr;
+ int id;
+ unsigned int slots;
+
+ /* per-stream data */
+ struct psc_dma_stream playback;
+ struct psc_dma_stream capture;
+
+ /* Statistics */
+ struct {
+ unsigned long overrun_count;
+ unsigned long underrun_count;
+ } stats;
+};
+
+int mpc5200_audio_dma_create(struct of_device *op);
+int mpc5200_audio_dma_destroy(struct of_device *op);
+
+extern struct snd_soc_platform mpc5200_audio_dma_platform;
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 000000000000..794a247b3eb5
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,329 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/time.h>
+#include <asm/delay.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ int status;
+ unsigned int val;
+
+ /* Wait for command send status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (rdy)\n");
+ return -ENODEV;
+ }
+ /* Send the read */
+ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+ /* Wait for the answer */
+ status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_DATA_VAL), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 read (val) %x\n",
+ in_be16(&psc_dma->psc_regs->sr_csr.status));
+ return -ENODEV;
+ }
+ /* Get the data */
+ val = in_be32(&psc_dma->psc_regs->ac97_data);
+ if (((val >> 24) & 0x7f) != reg) {
+ pr_err("reg echo error on ac97 read\n");
+ return -ENODEV;
+ }
+ val = (val >> 8) & 0xffff;
+
+ return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ int status;
+
+ /* Wait for command status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (write)\n");
+ return;
+ }
+ /* Write data */
+ out_be32(&psc_dma->psc_regs->ac97_cmd,
+ ((reg & 0x7f) << 24) | (val << 8));
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+ udelay(3);
+ out_be32(&regs->sicr, psc_dma->sicr);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Do a cold reset */
+ out_8(&regs->op1, MPC52xx_PSC_OP_RES);
+ udelay(10);
+ out_8(&regs->op0, MPC52xx_PSC_OP_RES);
+ udelay(50);
+ psc_ac97_warm_reset(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = psc_ac97_read,
+ .write = psc_ac97_write,
+ .reset = psc_ac97_cold_reset,
+ .warm_reset = psc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i channels=%i"
+ " rate=%i format=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params),
+ params_channels(params), params_rate(params),
+ params_format(params));
+
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x00000100;
+ else
+ psc_dma->slots |= 0x00000300;
+ } else {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x01000000;
+ else
+ psc_dma->slots |= 0x03000000;
+ }
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+
+ return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+
+ if (params_channels(params) == 1)
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+ else
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+
+ return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ psc_dma->slots &= 0xFFFF0000;
+ else
+ psc_dma->slots &= 0x0000FFFF;
+
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ break;
+ }
+ return 0;
+}
+
+static int psc_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Go */
+ out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+ .hw_params = psc_ac97_hw_analog_params,
+ .trigger = psc_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+ .hw_params = psc_ac97_hw_digital_params,
+};
+
+struct snd_soc_dai psc_ac97_dai[] = {
+{
+ .name = "AC97",
+ .ac97_control = 1,
+ .probe = psc_ac97_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .ops = &psc_ac97_analog_ops,
+},
+{
+ .name = "SPDIF",
+ .ac97_control = 1,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+ },
+ .ops = &psc_ac97_digital_ops,
+} };
+EXPORT_SYMBOL_GPL(psc_ac97_dai);
+
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_ac97_of_probe(struct of_device *op,
+ const struct of_device_id *match)
+{
+ int rc, i;
+ struct snd_ac97 ac97;
+ struct mpc52xx_psc __iomem *regs;
+
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].dev = &op->dev;
+
+ rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+ if (rc != 0) {
+ dev_err(&op->dev, "Failed to register DAI\n");
+ return rc;
+ }
+
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
+ ac97.private_data = psc_dma;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].private_data = psc_dma;
+
+ psc_dma->imr = 0;
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+ /* Configure the serial interface mode to AC97 */
+ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+ out_be32(&regs->sicr, psc_dma->sicr);
+
+ /* No slots active */
+ out_be32(&regs->ac97_slots, 0x00000000);
+
+ return 0;
+}
+
+static int __devexit psc_ac97_of_remove(struct of_device *op)
+{
+ return mpc5200_audio_dma_destroy(op);
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_ac97_match[] __devinitdata = {
+ { .compatible = "fsl,mpc5200-psc-ac97", },
+ { .compatible = "fsl,mpc5200b-psc-ac97", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct of_platform_driver psc_ac97_driver = {
+ .match_table = psc_ac97_match,
+ .probe = psc_ac97_of_probe,
+ .remove = __devexit_p(psc_ac97_of_remove),
+ .driver = {
+ .name = "mpc5200-psc-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in AC97 mode.
+ */
+static int __init psc_ac97_init(void)
+{
+ return of_register_platform_driver(&psc_ac97_driver);
+}
+module_init(psc_ac97_init);
+
+static void __exit psc_ac97_exit(void)
+{
+ of_unregister_platform_driver(&psc_ac97_driver);
+}
+module_exit(psc_ac97_exit);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 000000000000..4bc18c35c369
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,15 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+extern struct snd_soc_dai psc_ac97_dai[];
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 1111c710118a..ce8de90fb94a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -3,31 +3,21 @@
* ALSA SoC Digital Audio Interface (DAI) driver
*
* Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
*/
-#include <linux/init.h>
#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
-#include <linux/dma-mapping.h>
-#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/initval.h>
#include <sound/soc.h>
-#include <sound/soc-of-simple.h>
-#include <sysdev/bestcomm/bestcomm.h>
-#include <sysdev/bestcomm/gen_bd.h>
#include <asm/mpc52xx_psc.h>
-MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
-MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
-MODULE_LICENSE("GPL");
+#include "mpc5200_psc_i2s.h"
+#include "mpc5200_dma.h"
/**
* PSC_I2S_RATES: sample rates supported by the I2S
@@ -44,191 +34,17 @@ MODULE_LICENSE("GPL");
* PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
*/
#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
- SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
- SNDRV_PCM_FMTBIT_S32_BE)
-
-/**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
- * @active: flag indicating if the stream is active
- * @psc_i2s: pointer back to parent psc_i2s data structure
- * @bcom_task: bestcomm task structure
- * @irq: irq number for bestcomm task
- * @period_start: physical address of start of DMA region
- * @period_end: physical address of end of DMA region
- * @period_next_pt: physical address of next DMA buffer to enqueue
- * @period_bytes: size of DMA period in bytes
- */
-struct psc_i2s_stream {
- int active;
- struct psc_i2s *psc_i2s;
- struct bcom_task *bcom_task;
- int irq;
- struct snd_pcm_substream *stream;
- dma_addr_t period_start;
- dma_addr_t period_end;
- dma_addr_t period_next_pt;
- dma_addr_t period_current_pt;
- int period_bytes;
-};
-
-/**
- * psc_i2s - Private driver data
- * @name: short name for this device ("PSC0", "PSC1", etc)
- * @psc_regs: pointer to the PSC's registers
- * @fifo_regs: pointer to the PSC's FIFO registers
- * @irq: IRQ of this PSC
- * @dev: struct device pointer
- * @dai: the CPU DAI for this device
- * @sicr: Base value used in serial interface control register; mode is ORed
- * with this value.
- * @playback: Playback stream context data
- * @capture: Capture stream context data
- */
-struct psc_i2s {
- char name[32];
- struct mpc52xx_psc __iomem *psc_regs;
- struct mpc52xx_psc_fifo __iomem *fifo_regs;
- unsigned int irq;
- struct device *dev;
- struct snd_soc_dai dai;
- spinlock_t lock;
- u32 sicr;
-
- /* per-stream data */
- struct psc_i2s_stream playback;
- struct psc_i2s_stream capture;
-
- /* Statistics */
- struct {
- int overrun_count;
- int underrun_count;
- } stats;
-};
-
-/*
- * Interrupt handlers
- */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
-{
- struct psc_i2s *psc_i2s = _psc_i2s;
- struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
- u16 isr;
-
- isr = in_be16(&regs->mpc52xx_psc_isr);
-
- /* Playback underrun error */
- if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
- psc_i2s->stats.underrun_count++;
-
- /* Capture overrun error */
- if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
- psc_i2s->stats.overrun_count++;
-
- out_8(&regs->command, 4 << 4); /* reset the error status */
-
- return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- * @s: pointer to stream private data structure
- *
- * Enqueues another audio period buffer into the bestcomm queue.
- *
- * Note: The routine must only be called when there is space available in
- * the queue. Otherwise the enqueue will fail and the audio ring buffer
- * will get out of sync
- */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
-{
- struct bcom_bd *bd;
-
- /* Prepare and enqueue the next buffer descriptor */
- bd = bcom_prepare_next_buffer(s->bcom_task);
- bd->status = s->period_bytes;
- bd->data[0] = s->period_next_pt;
- bcom_submit_next_buffer(s->bcom_task, NULL);
-
- /* Update for next period */
- s->period_next_pt += s->period_bytes;
- if (s->period_next_pt >= s->period_end)
- s->period_next_pt = s->period_start;
-}
-
-/* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
-{
- struct psc_i2s_stream *s = _psc_i2s_stream;
-
- /* For each finished period, dequeue the completed period buffer
- * and enqueue a new one in it's place. */
- while (bcom_buffer_done(s->bcom_task)) {
- bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
- s->period_current_pt += s->period_bytes;
- if (s->period_current_pt >= s->period_end)
- s->period_current_pt = s->period_start;
- psc_i2s_bcom_enqueue_next_buffer(s);
- bcom_enable(s->bcom_task);
- }
-
- /* If the stream is active, then also inform the PCM middle layer
- * of the period finished event. */
- if (s->active)
- snd_pcm_period_elapsed(s->stream);
-
- return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_startup: create a new substream
- *
- * This is the first function called when a stream is opened.
- *
- * If this is the first stream open, then grab the IRQ and program most of
- * the PSC registers.
- */
-static int psc_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- int rc;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
-
- if (!psc_i2s->playback.active &&
- !psc_i2s->capture.active) {
- /* Setup the IRQs */
- rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
- "psc-i2s-status", psc_i2s);
- rc |= request_irq(psc_i2s->capture.irq,
- &psc_i2s_bcom_irq, IRQF_SHARED,
- "psc-i2s-capture", &psc_i2s->capture);
- rc |= request_irq(psc_i2s->playback.irq,
- &psc_i2s_bcom_irq, IRQF_SHARED,
- "psc-i2s-playback", &psc_i2s->playback);
- if (rc) {
- free_irq(psc_i2s->irq, psc_i2s);
- free_irq(psc_i2s->capture.irq,
- &psc_i2s->capture);
- free_irq(psc_i2s->playback.irq,
- &psc_i2s->playback);
- return -ENODEV;
- }
- }
-
- return 0;
-}
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
u32 mode;
- dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
" periods=%i buffer_size=%i buffer_bytes=%i\n",
__func__, substream, params_period_size(params),
params_period_bytes(params), params_periods(params),
@@ -248,175 +64,15 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
break;
default:
- dev_dbg(psc_i2s->dev, "invalid format\n");
- return -EINVAL;
- }
- out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- return 0;
-}
-
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- snd_pcm_set_runtime_buffer(substream, NULL);
- return 0;
-}
-
-/**
- * psc_i2s_trigger: start and stop the DMA transfer.
- *
- * This function is called by ALSA to start, stop, pause, and resume the DMA
- * transfer of data.
- */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct psc_i2s_stream *s;
- struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
- u16 imr;
- u8 psc_cmd;
- unsigned long flags;
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
- " stream_id=%i\n",
- substream, cmd, substream->pstr->stream);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- s->period_bytes = frames_to_bytes(runtime,
- runtime->period_size);
- s->period_start = virt_to_phys(runtime->dma_area);
- s->period_end = s->period_start +
- (s->period_bytes * runtime->periods);
- s->period_next_pt = s->period_start;
- s->period_current_pt = s->period_start;
- s->active = 1;
-
- /* First; reset everything */
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
- out_8(&regs->command, MPC52xx_PSC_RST_RX);
- out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
- } else {
- out_8(&regs->command, MPC52xx_PSC_RST_TX);
- out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
- }
-
- /* Next, fill up the bestcomm bd queue and enable DMA.
- * This will begin filling the PSC's fifo. */
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- bcom_gen_bd_rx_reset(s->bcom_task);
- else
- bcom_gen_bd_tx_reset(s->bcom_task);
- while (!bcom_queue_full(s->bcom_task))
- psc_i2s_bcom_enqueue_next_buffer(s);
- bcom_enable(s->bcom_task);
-
- /* Due to errata in the i2s mode; need to line up enabling
- * the transmitter with a transition on the frame sync
- * line */
-
- spin_lock_irqsave(&psc_i2s->lock, flags);
- /* first make sure it is low */
- while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
- ;
- /* then wait for the transition to high */
- while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
- ;
- /* Finally, enable the PSC.
- * Receiver must always be enabled; even when we only want
- * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
- psc_cmd = MPC52xx_PSC_RX_ENABLE;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
- psc_cmd |= MPC52xx_PSC_TX_ENABLE;
- out_8(&regs->command, psc_cmd);
- spin_unlock_irqrestore(&psc_i2s->lock, flags);
-
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- /* Turn off the PSC */
- s->active = 0;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
- if (!psc_i2s->playback.active) {
- out_8(&regs->command, 2 << 4); /* reset rx */
- out_8(&regs->command, 3 << 4); /* reset tx */
- out_8(&regs->command, 4 << 4); /* reset err */
- }
- } else {
- out_8(&regs->command, 3 << 4); /* reset tx */
- out_8(&regs->command, 4 << 4); /* reset err */
- if (!psc_i2s->capture.active)
- out_8(&regs->command, 2 << 4); /* reset rx */
- }
-
- bcom_disable(s->bcom_task);
- while (!bcom_queue_empty(s->bcom_task))
- bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-
- break;
-
- default:
- dev_dbg(psc_i2s->dev, "invalid command\n");
+ dev_dbg(psc_dma->dev, "invalid format\n");
return -EINVAL;
}
-
- /* Update interrupt enable settings */
- imr = 0;
- if (psc_i2s->playback.active)
- imr |= MPC52xx_PSC_IMR_TXEMP;
- if (psc_i2s->capture.active)
- imr |= MPC52xx_PSC_IMR_ORERR;
- out_be16(&regs->isr_imr.imr, imr);
+ out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
return 0;
}
/**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
- *
- * Shutdown the PSC if there are no other substreams open.
- */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
-
- /*
- * If this is the last active substream, disable the PSC and release
- * the IRQ.
- */
- if (!psc_i2s->playback.active &&
- !psc_i2s->capture.active) {
-
- /* Disable all interrupts and reset the PSC */
- out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
- out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
- out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
- out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
- out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
-
- /* Release irqs */
- free_irq(psc_i2s->irq, psc_i2s);
- free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
- free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
- }
-}
-
-/**
* psc_i2s_set_sysclk: set the clock frequency and direction
*
* This function is called by the machine driver to tell us what the clock
@@ -433,8 +89,8 @@ static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
cpu_dai, dir);
return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
}
@@ -452,8 +108,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
*/
static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
cpu_dai, format);
return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
}
@@ -469,16 +125,13 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
* psc_i2s_dai_template: template CPU Digital Audio Interface
*/
static struct snd_soc_dai_ops psc_i2s_dai_ops = {
- .startup = psc_i2s_startup,
.hw_params = psc_i2s_hw_params,
- .hw_free = psc_i2s_hw_free,
- .shutdown = psc_i2s_shutdown,
- .trigger = psc_i2s_trigger,
.set_sysclk = psc_i2s_set_sysclk,
.set_fmt = psc_i2s_set_fmt,
};
-static struct snd_soc_dai psc_i2s_dai_template = {
+struct snd_soc_dai psc_i2s_dai[] = {{
+ .name = "I2S",
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -492,223 +145,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
.formats = PSC_I2S_FORMATS,
},
.ops = &psc_i2s_dai_ops,
-};
-
-/* ---------------------------------------------------------------------
- * The PSC I2S 'ASoC platform' driver
- *
- * Can be referenced by an 'ASoC machine' driver
- * This driver only deals with the audio bus; it doesn't have any
- * interaction with the attached codec
- */
-
-static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
- .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_BATCH,
- .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
- SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
- .rate_min = 8000,
- .rate_max = 48000,
- .channels_min = 2,
- .channels_max = 2,
- .period_bytes_max = 1024 * 1024,
- .period_bytes_min = 32,
- .periods_min = 2,
- .periods_max = 256,
- .buffer_bytes_max = 2 * 1024 * 1024,
- .fifo_size = 0,
-};
-
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
-
- s->stream = substream;
- return 0;
-}
-
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- s->stream = NULL;
- return 0;
-}
-
-static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
- dma_addr_t count;
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- count = s->period_current_pt - s->period_start;
-
- return bytes_to_frames(substream->runtime, count);
-}
-
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
- .open = psc_i2s_pcm_open,
- .close = psc_i2s_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .pointer = psc_i2s_pcm_pointer,
-};
-
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
-{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
- size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
- int rc = 0;
-
- dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
- card, dai, pcm);
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &psc_i2s_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = 0xffffffff;
-
- if (pcm->streams[0].substream) {
- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
- &pcm->streams[0].substream->dma_buffer);
- if (rc)
- goto playback_alloc_err;
- }
-
- if (pcm->streams[1].substream) {
- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
- &pcm->streams[1].substream->dma_buffer);
- if (rc)
- goto capture_alloc_err;
- }
-
- return 0;
-
- capture_alloc_err:
- if (pcm->streams[0].substream)
- snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
- playback_alloc_err:
- dev_err(card->dev, "Cannot allocate buffer(s)\n");
- return -ENOMEM;
-}
-
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
-{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
- struct snd_pcm_substream *substream;
- int stream;
-
- dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (substream) {
- snd_dma_free_pages(&substream->dma_buffer);
- substream->dma_buffer.area = NULL;
- substream->dma_buffer.addr = 0;
- }
- }
-}
-
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
- .name = "mpc5200-psc-audio",
- .pcm_ops = &psc_i2s_pcm_ops,
- .pcm_new = &psc_i2s_pcm_new,
- .pcm_free = &psc_i2s_pcm_free,
-};
-
-/* ---------------------------------------------------------------------
- * Sysfs attributes for debugging
- */
-
-static ssize_t psc_i2s_status_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-
- return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
- "tfnum=%i tfstat=0x%.4x\n",
- in_be16(&psc_i2s->psc_regs->sr_csr.status),
- in_be32(&psc_i2s->psc_regs->sicr),
- in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
- in_be16(&psc_i2s->fifo_regs->rfstat),
- in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
- in_be16(&psc_i2s->fifo_regs->tfstat));
-}
-
-static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
-{
- if (strcmp(name, "playback_underrun") == 0)
- return &psc_i2s->stats.underrun_count;
- if (strcmp(name, "capture_overrun") == 0)
- return &psc_i2s->stats.overrun_count;
-
- return NULL;
-}
-
-static ssize_t psc_i2s_stat_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
-
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
- return 0;
-
- return sprintf(buf, "%i\n", *attrib);
-}
-
-static ssize_t psc_i2s_stat_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf,
- size_t count)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
-
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
- return 0;
-
- *attrib = simple_strtoul(buf, NULL, 0);
- return count;
-}
-
-static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
- psc_i2s_stat_store);
-static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
- psc_i2s_stat_store);
+} };
+EXPORT_SYMBOL_GPL(psc_i2s_dai);
/* ---------------------------------------------------------------------
* OF platform bus binding code:
@@ -718,150 +156,65 @@ static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)
{
- phys_addr_t fifo;
- struct psc_i2s *psc_i2s;
- struct resource res;
- int size, psc_id, irq, rc;
- const __be32 *prop;
- void __iomem *regs;
-
- dev_dbg(&op->dev, "probing psc i2s device\n");
-
- /* Get the PSC ID */
- prop = of_get_property(op->node, "cell-index", &size);
- if (!prop || size < sizeof *prop)
- return -ENODEV;
- psc_id = be32_to_cpu(*prop);
-
- /* Fetch the registers and IRQ of the PSC */
- irq = irq_of_parse_and_map(op->node, 0);
- if (of_address_to_resource(op->node, 0, &res)) {
- dev_err(&op->dev, "Missing reg property\n");
- return -ENODEV;
- }
- regs = ioremap(res.start, 1 + res.end - res.start);
- if (!regs) {
- dev_err(&op->dev, "Could not map registers\n");
- return -ENODEV;
- }
+ int rc;
+ struct psc_dma *psc_dma;
+ struct mpc52xx_psc __iomem *regs;
- /* Allocate and initialize the driver private data */
- psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
- if (!psc_i2s) {
- iounmap(regs);
- return -ENOMEM;
- }
- spin_lock_init(&psc_i2s->lock);
- psc_i2s->irq = irq;
- psc_i2s->psc_regs = regs;
- psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
- psc_i2s->dev = &op->dev;
- psc_i2s->playback.psc_i2s = psc_i2s;
- psc_i2s->capture.psc_i2s = psc_i2s;
- snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
-
- /* Fill out the CPU DAI structure */
- memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
- psc_i2s->dai.private_data = psc_i2s;
- psc_i2s->dai.name = psc_i2s->name;
- psc_i2s->dai.id = psc_id;
-
- /* Find the address of the fifo data registers and setup the
- * DMA tasks */
- fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
- psc_i2s->capture.bcom_task =
- bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
- psc_i2s->playback.bcom_task =
- bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
- if (!psc_i2s->capture.bcom_task ||
- !psc_i2s->playback.bcom_task) {
- dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
- iounmap(regs);
- kfree(psc_i2s);
- return -ENODEV;
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+ if (rc != 0) {
+ pr_err("Failed to register DAI\n");
+ return 0;
}
- /* Disable all interrupts and reset the PSC */
- out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
- out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
- out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
- out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
- out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
/* Configure the serial interface mode; defaulting to CODEC8 mode */
- psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+ psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
MPC52xx_PSC_SICR_CLKPOL;
- if (of_get_property(op->node, "fsl,cellslave", NULL))
- psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
- MPC52xx_PSC_SICR_GENCLK;
- out_be32(&psc_i2s->psc_regs->sicr,
- psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+ out_be32(&psc_dma->psc_regs->sicr,
+ psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
/* Check for the codec handle. If it is not present then we
* are done */
if (!of_get_property(op->node, "codec-handle", NULL))
return 0;
- /* Set up mode register;
- * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
- * Second write: register Normal mode for non loopback
- */
- out_8(&psc_i2s->psc_regs->mode, 0);
- out_8(&psc_i2s->psc_regs->mode, 0);
-
- /* Set the TX and RX fifo alarm thresholds */
- out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
- out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
- out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
- out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
-
- /* Lookup the IRQ numbers */
- psc_i2s->playback.irq =
- bcom_get_task_irq(psc_i2s->playback.bcom_task);
- psc_i2s->capture.irq =
- bcom_get_task_irq(psc_i2s->capture.bcom_task);
-
- /* Save what we've done so it can be found again later */
- dev_set_drvdata(&op->dev, psc_i2s);
-
- /* Register the SYSFS files */
- rc = device_create_file(psc_i2s->dev, &dev_attr_status);
- rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
- rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
- if (rc)
- dev_info(psc_i2s->dev, "error creating sysfs files\n");
-
- snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
-
- /* Tell the ASoC OF helpers about it */
- of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
- &psc_i2s->dai);
+ /* Due to errata in the dma mode; need to line up enabling
+ * the transmitter with a transition on the frame sync
+ * line */
+
+ /* first make sure it is low */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+ ;
+ /* then wait for the transition to high */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+ ;
+ /* Finally, enable the PSC.
+ * Receiver must always be enabled; even when we only want
+ * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
+
+ /* Go */
+ out_8(&psc_dma->psc_regs->command,
+ MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
return 0;
+
}
static int __devexit psc_i2s_of_remove(struct of_device *op)
{
- struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
-
- dev_dbg(&op->dev, "psc_i2s_remove()\n");
-
- snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
-
- bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
- bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
-
- iounmap(psc_i2s->psc_regs);
- iounmap(psc_i2s->fifo_regs);
- kfree(psc_i2s);
- dev_set_drvdata(&op->dev, NULL);
-
- return 0;
+ return mpc5200_audio_dma_destroy(op);
}
/* Match table for of_platform binding */
static struct of_device_id psc_i2s_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-i2s", },
+ { .compatible = "fsl,mpc5200b-psc-i2s", },
{}
};
MODULE_DEVICE_TABLE(of, psc_i2s_match);
@@ -892,4 +245,7 @@ static void __exit psc_i2s_exit(void)
}
module_exit(psc_i2s_exit);
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h
new file mode 100644
index 000000000000..ce55e070fdf3
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.h
@@ -0,0 +1,12 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+
+extern struct snd_soc_dai psc_i2s_dai[];
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
new file mode 100644
index 000000000000..8766f7a3893d
--- /dev/null
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/wm9712.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link pcm030_fabric_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 Analog",
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+ .name = "AC97",
+ .stream_name = "AC97 IEC958",
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int pcm030_fabric_init(void)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ if (!machine_is_compatible("phytec,pcm030"))
+ return -ENODEV;
+
+ card.platform = &mpc5200_audio_dma_platform;
+ card.name = "pcm030";
+ card.dai_link = pcm030_fabric_dai;
+ card.num_links = ARRAY_SIZE(pcm030_fabric_dai);
+
+ device.card = &card;
+ device.codec_dev = &soc_codec_dev_wm9712;
+
+ pdev = platform_device_alloc("soc-audio", 1);
+ if (!pdev) {
+ pr_err("pcm030_fabric_init: platform_device_alloc() failed\n");
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(pdev, &device);
+ device.dev = &pdev->dev;
+
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+ return -ENODEV;
+ }
+ return 0;
+}
+
+module_init(pcm030_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
+MODULE_LICENSE("GPL");
+
generated by cgit v1.2.3 (git 2.39.0) at 2024-11-23 18:33:22 +0300