From 1f93a7dfd1f8d5ff7a5c53246c7534fe2332d6f4 Mon Sep 17 00:00:00 2001
From: hc <hc@nodka.com>
Date: Mon, 11 Dec 2023 02:46:07 +0000
Subject: [PATCH] add audio
---
kernel/sound/soc/qcom/qdsp6/q6asm-dai.c | 1010 ++++++++++++++++++++++++++++++++++++++++++++++++++++------
1 files changed, 893 insertions(+), 117 deletions(-)
diff --git a/kernel/sound/soc/qcom/qdsp6/q6asm-dai.c b/kernel/sound/soc/qcom/qdsp6/q6asm-dai.c
index c1a7d37..84cf190 100644
--- a/kernel/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/kernel/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -8,9 +8,10 @@
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
-#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
+#include <linux/spinlock.h>
+#include <sound/compress_driver.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/of_device.h>
@@ -31,6 +32,15 @@
#define CAPTURE_MIN_PERIOD_SIZE 320
#define SID_MASK_DEFAULT 0xF
+/* Default values used if user space does not set */
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
+#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+
+#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1)
+#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
+
enum stream_state {
Q6ASM_STREAM_IDLE = 0,
Q6ASM_STREAM_STOPPED,
@@ -39,24 +49,39 @@
struct q6asm_dai_rtd {
struct snd_pcm_substream *substream;
+ struct snd_compr_stream *cstream;
+ struct snd_codec codec;
+ struct snd_dma_buffer dma_buffer;
+ spinlock_t lock;
phys_addr_t phys;
unsigned int pcm_size;
unsigned int pcm_count;
unsigned int pcm_irq_pos; /* IRQ position */
unsigned int periods;
+ unsigned int bytes_sent;
+ unsigned int bytes_received;
+ unsigned int copied_total;
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
+ uint32_t next_track_stream_id;
+ bool next_track;
+ uint32_t stream_id;
uint16_t session_id;
enum stream_state state;
+ uint32_t initial_samples_drop;
+ uint32_t trailing_samples_drop;
+ bool notify_on_drain;
};
struct q6asm_dai_data {
+ struct snd_soc_dai_driver *dais;
+ int num_dais;
long long int sid;
};
-static struct snd_pcm_hardware q6asm_dai_hardware_capture = {
- .info = (SNDRV_PCM_INFO_MMAP |
+static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
@@ -78,7 +103,7 @@
};
static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
- .info = (SNDRV_PCM_INFO_MMAP |
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
@@ -123,7 +148,6 @@
.rate_max = 48000, \
}, \
.name = "MultiMedia"#num, \
- .probe = fe_dai_probe, \
.id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
}
@@ -139,8 +163,23 @@
.mask = 0,
};
+static const struct snd_compr_codec_caps q6asm_compr_caps = {
+ .num_descriptors = 1,
+ .descriptor[0].max_ch = 2,
+ .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000 },
+ .descriptor[0].num_sample_rates = 13,
+ .descriptor[0].bit_rate[0] = 320,
+ .descriptor[0].bit_rate[1] = 128,
+ .descriptor[0].num_bitrates = 2,
+ .descriptor[0].profiles = 0,
+ .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
+ .descriptor[0].formats = 0,
+};
+
static void event_handler(uint32_t opcode, uint32_t token,
- uint32_t *payload, void *priv)
+ void *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_pcm_substream *substream = prtd->substream;
@@ -148,8 +187,8 @@
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- q6asm_write_async(prtd->audio_client,
- prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
prtd->state = Q6ASM_STREAM_STOPPED;
@@ -158,8 +197,8 @@
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_write_async(prtd->audio_client,
- prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
break;
}
@@ -167,7 +206,7 @@
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
break;
default:
@@ -175,21 +214,22 @@
}
}
-static int q6asm_dai_prepare(struct snd_pcm_substream *substream)
+static int q6asm_dai_prepare(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
struct q6asm_dai_rtd *prtd = runtime->private_data;
- struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
int ret, i;
- pdata = snd_soc_component_get_drvdata(c);
+ pdata = snd_soc_component_get_drvdata(component);
if (!pdata)
return -EINVAL;
if (!prtd || !prtd->audio_client) {
- pr_err("%s: private data null or audio client freed\n",
+ dev_err(dev, "%s: private data null or audio client freed\n",
__func__);
return -EINVAL;
}
@@ -199,7 +239,7 @@
/* rate and channels are sent to audio driver */
if (prtd->state) {
/* clear the previous setup if any */
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6routing_stream_close(soc_prtd->dai_link->id,
@@ -212,58 +252,70 @@
prtd->periods);
if (ret < 0) {
- pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
ret);
return -ENOMEM;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ 0, prtd->bits_per_sample, false);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
}
if (ret < 0) {
- pr_err("%s: q6asm_open_write failed\n", __func__);
- q6asm_audio_client_free(prtd->audio_client);
- prtd->audio_client = NULL;
- return -ENOMEM;
+ dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
+ goto open_err;
}
prtd->session_id = q6asm_get_session_id(prtd->audio_client);
ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
prtd->session_id, substream->stream);
if (ret) {
- pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
- return ret;
+ dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
+ goto routing_err;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_media_format_block_multi_ch_pcm(
- prtd->audio_client, runtime->rate,
- runtime->channels, NULL,
+ prtd->audio_client, prtd->stream_id,
+ runtime->rate, runtime->channels, NULL,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
- runtime->rate, runtime->channels,
- prtd->bits_per_sample);
+ prtd->stream_id,
+ runtime->rate,
+ runtime->channels,
+ prtd->bits_per_sample);
/* Queue the buffers */
for (i = 0; i < runtime->periods; i++)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
}
if (ret < 0)
- pr_info("%s: CMD Format block failed\n", __func__);
+ dev_info(dev, "%s: CMD Format block failed\n", __func__);
+ else
+ prtd->state = Q6ASM_STREAM_RUNNING;
- prtd->state = Q6ASM_STREAM_RUNNING;
+ return ret;
- return 0;
+routing_err:
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
+open_err:
+ q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+
+ return ret;
}
-static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int q6asm_dai_trigger(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
@@ -273,15 +325,18 @@
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
break;
default:
ret = -EINVAL;
@@ -291,23 +346,23 @@
return ret;
}
-static int q6asm_dai_open(struct snd_pcm_substream *substream)
+static int q6asm_dai_open(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
- struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
struct q6asm_dai_rtd *prtd;
struct q6asm_dai_data *pdata;
- struct device *dev = c->dev;
+ struct device *dev = component->dev;
int ret = 0;
int stream_id;
stream_id = cpu_dai->driver->id;
- pdata = snd_soc_component_get_drvdata(c);
+ pdata = snd_soc_component_get_drvdata(component);
if (!pdata) {
- pr_err("Drv data not found ..\n");
+ dev_err(dev, "Drv data not found ..\n");
return -EINVAL;
}
@@ -320,11 +375,14 @@
(q6asm_cb)event_handler, prtd, stream_id,
LEGACY_PCM_MODE);
if (IS_ERR(prtd->audio_client)) {
- pr_info("%s: Could not allocate memory\n", __func__);
+ dev_info(dev, "%s: Could not allocate memory\n", __func__);
ret = PTR_ERR(prtd->audio_client);
kfree(prtd);
return ret;
}
+
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = q6asm_dai_hardware_playback;
@@ -335,12 +393,12 @@
SNDRV_PCM_HW_PARAM_RATE,
&constraints_sample_rates);
if (ret < 0)
- pr_info("snd_pcm_hw_constraint_list failed\n");
+ dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
- pr_info("snd_pcm_hw_constraint_integer failed\n");
+ dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = snd_pcm_hw_constraint_minmax(runtime,
@@ -348,21 +406,21 @@
PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
if (ret < 0) {
- pr_err("constraint for buffer bytes min max ret = %d\n",
- ret);
+ dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
+ ret);
}
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
if (ret < 0) {
- pr_err("constraint for period bytes step ret = %d\n",
+ dev_err(dev, "constraint for period bytes step ret = %d\n",
ret);
}
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
if (ret < 0) {
- pr_err("constraint for buffer bytes step ret = %d\n",
+ dev_err(dev, "constraint for buffer bytes step ret = %d\n",
ret);
}
@@ -383,15 +441,17 @@
return 0;
}
-static int q6asm_dai_close(struct snd_pcm_substream *substream)
+static int q6asm_dai_close(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
struct q6asm_dai_rtd *prtd = runtime->private_data;
if (prtd->audio_client) {
if (prtd->state)
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
@@ -404,7 +464,8 @@
return 0;
}
-static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream)
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
@@ -416,22 +477,21 @@
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
-static int q6asm_dai_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
+static int q6asm_dai_mmap(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
{
-
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
- struct device *dev = c->dev;
+ struct device *dev = component->dev;
return dma_mmap_coherent(dev, vma,
runtime->dma_area, runtime->dma_addr,
runtime->dma_bytes);
}
-static int q6asm_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+static int q6asm_dai_hw_params(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
@@ -451,26 +511,693 @@
return 0;
}
-static struct snd_pcm_ops q6asm_dai_ops = {
- .open = q6asm_dai_open,
- .hw_params = q6asm_dai_hw_params,
- .close = q6asm_dai_close,
- .ioctl = snd_pcm_lib_ioctl,
- .prepare = q6asm_dai_prepare,
- .trigger = q6asm_dai_trigger,
- .pointer = q6asm_dai_pointer,
- .mmap = q6asm_dai_mmap,
+static void compress_event_handler(uint32_t opcode, uint32_t token,
+ void *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_compr_stream *substream = prtd->cstream;
+ unsigned long flags;
+ u32 wflags = 0;
+ uint64_t avail;
+ uint32_t bytes_written, bytes_to_write;
+ bool is_last_buffer = false;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (!prtd->bytes_sent) {
+ q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->initial_samples_drop);
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
+ prtd->bytes_sent += prtd->pcm_count;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->notify_on_drain) {
+ if (substream->partial_drain) {
+ /*
+ * Close old stream and make it stale, switch
+ * the active stream now!
+ */
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id,
+ CMD_CLOSE);
+ /*
+ * vaild stream ids start from 1, So we are
+ * toggling this between 1 and 2.
+ */
+ prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ }
+
+ snd_compr_drain_notify(prtd->cstream);
+ prtd->notify_on_drain = false;
+
+ } else {
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
+ prtd->copied_total += bytes_written;
+ snd_compr_fragment_elapsed(substream);
+
+ if (prtd->state != Q6ASM_STREAM_RUNNING) {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ avail = prtd->bytes_received - prtd->bytes_sent;
+ if (avail > prtd->pcm_count) {
+ bytes_to_write = prtd->pcm_count;
+ } else {
+ if (substream->partial_drain || prtd->notify_on_drain)
+ is_last_buffer = true;
+ bytes_to_write = avail;
+ }
+
+ if (bytes_to_write) {
+ if (substream->partial_drain && is_last_buffer) {
+ wflags |= ASM_LAST_BUFFER_FLAG;
+ q6asm_stream_remove_trailing_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->trailing_samples_drop);
+ }
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+
+ prtd->bytes_sent += bytes_to_write;
+ }
+
+ if (prtd->notify_on_drain && is_last_buffer)
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id, CMD_EOS);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_compr_open(struct snd_soc_component *component,
+ struct snd_compr_stream *stream)
+{
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ struct q6asm_dai_rtd *prtd;
+ int stream_id, size, ret;
+
+ stream_id = cpu_dai->driver->id;
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata) {
+ dev_err(dev, "Drv data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (!prtd)
+ return -ENOMEM;
+
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
+ prtd->cstream = stream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)compress_event_handler,
+ prtd, stream_id, LEGACY_PCM_MODE);
+ if (IS_ERR(prtd->audio_client)) {
+ dev_err(dev, "Could not allocate memory\n");
+ ret = PTR_ERR(prtd->audio_client);
+ goto free_prtd;
+ }
+
+ size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
+ COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &prtd->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ goto free_client;
+ }
+
+ if (pdata->sid < 0)
+ prtd->phys = prtd->dma_buffer.addr;
+ else
+ prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
+
+ snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
+ spin_lock_init(&prtd->lock);
+ runtime->private_data = prtd;
+
+ return 0;
+
+free_client:
+ q6asm_audio_client_free(prtd->audio_client);
+free_prtd:
+ kfree(prtd);
+
+ return ret;
+}
+
+static int q6asm_dai_compr_free(struct snd_soc_component *component,
+ struct snd_compr_stream *stream)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+
+ if (prtd->audio_client) {
+ if (prtd->state) {
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
+ if (prtd->next_track_stream_id) {
+ q6asm_cmd(prtd->audio_client,
+ prtd->next_track_stream_id,
+ CMD_CLOSE);
+ }
+ }
+
+ snd_dma_free_pages(&prtd->dma_buffer);
+ q6asm_unmap_memory_regions(stream->direction,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ }
+ q6routing_stream_close(rtd->dai_link->id, stream->direction);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_codec *codec,
+ int stream_id)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct q6asm_flac_cfg flac_cfg;
+ struct q6asm_wma_cfg wma_cfg;
+ struct q6asm_alac_cfg alac_cfg;
+ struct q6asm_ape_cfg ape_cfg;
+ unsigned int wma_v9 = 0;
+ struct device *dev = component->dev;
+ int ret;
+ union snd_codec_options *codec_options;
+ struct snd_dec_flac *flac;
+ struct snd_dec_wma *wma;
+ struct snd_dec_alac *alac;
+ struct snd_dec_ape *ape;
+
+ codec_options = &(prtd->codec.options);
+
+ memcpy(&prtd->codec, codec, sizeof(*codec));
+
+ switch (codec->id) {
+ case SND_AUDIOCODEC_FLAC:
+
+ memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
+ flac = &codec_options->flac_d;
+
+ flac_cfg.ch_cfg = codec->ch_in;
+ flac_cfg.sample_rate = codec->sample_rate;
+ flac_cfg.stream_info_present = 1;
+ flac_cfg.sample_size = flac->sample_size;
+ flac_cfg.min_blk_size = flac->min_blk_size;
+ flac_cfg.max_blk_size = flac->max_blk_size;
+ flac_cfg.max_frame_size = flac->max_frame_size;
+ flac_cfg.min_frame_size = flac->min_frame_size;
+
+ ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+ stream_id,
+ &flac_cfg);
+ if (ret < 0) {
+ dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_WMA:
+ wma = &codec_options->wma_d;
+
+ memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+ wma_cfg.sample_rate = codec->sample_rate;
+ wma_cfg.num_channels = codec->ch_in;
+ wma_cfg.bytes_per_sec = codec->bit_rate / 8;
+ wma_cfg.block_align = codec->align;
+ wma_cfg.bits_per_sample = prtd->bits_per_sample;
+ wma_cfg.enc_options = wma->encoder_option;
+ wma_cfg.adv_enc_options = wma->adv_encoder_option;
+ wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+ if (wma_cfg.num_channels == 1)
+ wma_cfg.channel_mask = 4; /* Mono Center */
+ else if (wma_cfg.num_channels == 2)
+ wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+ else
+ return -EINVAL;
+
+ /* check the codec profile */
+ switch (codec->profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ wma_cfg.fmtag = 0x161;
+ wma_v9 = 1;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10:
+ wma_cfg.fmtag = 0x166;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ wma_cfg.fmtag = 0x162;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ wma_cfg.fmtag = 0x163;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ wma_cfg.fmtag = 0x167;
+ break;
+
+ default:
+ dev_err(dev, "Unknown WMA profile:%x\n",
+ codec->profile);
+ return -EIO;
+ }
+
+ if (wma_v9)
+ ret = q6asm_stream_media_format_block_wma_v9(
+ prtd->audio_client, stream_id,
+ &wma_cfg);
+ else
+ ret = q6asm_stream_media_format_block_wma_v10(
+ prtd->audio_client, stream_id,
+ &wma_cfg);
+ if (ret < 0) {
+ dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_ALAC:
+ memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+ alac = &codec_options->alac_d;
+
+ alac_cfg.sample_rate = codec->sample_rate;
+ alac_cfg.avg_bit_rate = codec->bit_rate;
+ alac_cfg.bit_depth = prtd->bits_per_sample;
+ alac_cfg.num_channels = codec->ch_in;
+
+ alac_cfg.frame_length = alac->frame_length;
+ alac_cfg.pb = alac->pb;
+ alac_cfg.mb = alac->mb;
+ alac_cfg.kb = alac->kb;
+ alac_cfg.max_run = alac->max_run;
+ alac_cfg.compatible_version = alac->compatible_version;
+ alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+ switch (codec->ch_in) {
+ case 1:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
+ break;
+ case 2:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
+ break;
+ }
+ ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+ stream_id,
+ &alac_cfg);
+ if (ret < 0) {
+ dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_APE:
+ memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+ ape = &codec_options->ape_d;
+
+ ape_cfg.sample_rate = codec->sample_rate;
+ ape_cfg.num_channels = codec->ch_in;
+ ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+ ape_cfg.compatible_version = ape->compatible_version;
+ ape_cfg.compression_level = ape->compression_level;
+ ape_cfg.format_flags = ape->format_flags;
+ ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+ ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+ ape_cfg.total_frames = ape->total_frames;
+ ape_cfg.seek_table_present = ape->seek_table_present;
+
+ ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+ stream_id,
+ &ape_cfg);
+ if (ret < 0) {
+ dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_params *params)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ int dir = stream->direction;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ int ret;
+
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ dev_err(dev, "private data null or audio client freed\n");
+ return -EINVAL;
+ }
+
+ prtd->periods = runtime->fragments;
+ prtd->pcm_count = runtime->fragment_size;
+ prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+ prtd->bits_per_sample = 16;
+
+ if (dir == SND_COMPRESS_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
+ params->codec.profile, prtd->bits_per_sample,
+ true);
+
+ if (ret < 0) {
+ dev_err(dev, "q6asm_open_write failed\n");
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return ret;
+ }
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, dir);
+ if (ret) {
+ dev_err(dev, "Stream reg failed ret:%d\n", ret);
+ return ret;
+ }
+
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ ¶ms->codec,
+ prtd->stream_id);
+ if (ret) {
+ dev_err(dev, "codec param setup failed ret:%d\n", ret);
+ return ret;
+ }
+
+ ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
+ return -ENOMEM;
+ }
+
+ prtd->state = Q6ASM_STREAM_RUNNING;
+
+ return 0;
+}
+
+static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_metadata *metadata)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ int ret = 0;
+
+ switch (metadata->key) {
+ case SNDRV_COMPRESS_ENCODER_PADDING:
+ prtd->trailing_samples_drop = metadata->value[0];
+ break;
+ case SNDRV_COMPRESS_ENCODER_DELAY:
+ prtd->initial_samples_drop = metadata->value[0];
+ if (prtd->next_track_stream_id) {
+ ret = q6asm_open_write(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->codec.id,
+ prtd->codec.profile,
+ prtd->bits_per_sample,
+ true);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ &prtd->codec,
+ prtd->next_track_stream_id);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+
+ ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->initial_samples_drop);
+ prtd->next_track_stream_id = 0;
+
+ }
+
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
+ struct snd_compr_stream *stream, int cmd)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
+ break;
+ case SND_COMPR_TRIGGER_NEXT_TRACK:
+ prtd->next_track = true;
+ prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ break;
+ case SND_COMPR_TRIGGER_DRAIN:
+ case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+ prtd->notify_on_drain = true;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_tstamp *tstamp)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ tstamp->copied_total = prtd->copied_total;
+ tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return 0;
+}
+
+static int q6asm_compr_copy(struct snd_soc_component *component,
+ struct snd_compr_stream *stream, char __user *buf,
+ size_t count)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ unsigned long flags;
+ u32 wflags = 0;
+ int avail, bytes_in_flight = 0;
+ void *dstn;
+ size_t copy;
+ u32 app_pointer;
+ u32 bytes_received;
+
+ bytes_received = prtd->bytes_received;
+
+ /**
+ * Make sure that next track data pointer is aligned at 32 bit boundary
+ * This is a Mandatory requirement from DSP data buffers alignment
+ */
+ if (prtd->next_track)
+ bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+ app_pointer = bytes_received/prtd->pcm_size;
+ app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
+ dstn = prtd->dma_buffer.area + app_pointer;
+
+ if (count < prtd->pcm_size - app_pointer) {
+ if (copy_from_user(dstn, buf, count))
+ return -EFAULT;
+ } else {
+ copy = prtd->pcm_size - app_pointer;
+ if (copy_from_user(dstn, buf, copy))
+ return -EFAULT;
+ if (copy_from_user(prtd->dma_buffer.area, buf + copy,
+ count - copy))
+ return -EFAULT;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+ if (prtd->next_track) {
+ prtd->next_track = false;
+ prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
+ prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+ }
+
+ prtd->bytes_received = bytes_received + count;
+
+ /* Kick off the data to dsp if its starving!! */
+ if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+ uint32_t bytes_to_write = prtd->pcm_count;
+
+ avail = prtd->bytes_received - prtd->bytes_sent;
+
+ if (avail < prtd->pcm_count)
+ bytes_to_write = avail;
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+ prtd->bytes_sent += bytes_to_write;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return count;
+}
+
+static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct vm_area_struct *vma)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct device *dev = component->dev;
+
+ return dma_mmap_coherent(dev, vma,
+ prtd->dma_buffer.area, prtd->dma_buffer.addr,
+ prtd->dma_buffer.bytes);
+}
+
+static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_caps *caps)
+{
+ caps->direction = SND_COMPRESS_PLAYBACK;
+ caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+ caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+ caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+ caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ caps->num_codecs = 5;
+ caps->codecs[0] = SND_AUDIOCODEC_MP3;
+ caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+ caps->codecs[2] = SND_AUDIOCODEC_WMA;
+ caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+ caps->codecs[4] = SND_AUDIOCODEC_APE;
+
+ return 0;
+}
+
+static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_codec_caps *codec)
+{
+ switch (codec->codec) {
+ case SND_AUDIOCODEC_MP3:
+ *codec = q6asm_compr_caps;
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_compress_ops q6asm_dai_compress_ops = {
+ .open = q6asm_dai_compr_open,
+ .free = q6asm_dai_compr_free,
+ .set_params = q6asm_dai_compr_set_params,
+ .set_metadata = q6asm_dai_compr_set_metadata,
+ .pointer = q6asm_dai_compr_pointer,
+ .trigger = q6asm_dai_compr_trigger,
+ .get_caps = q6asm_dai_compr_get_caps,
+ .get_codec_caps = q6asm_dai_compr_get_codec_caps,
+ .mmap = q6asm_dai_compr_mmap,
+ .copy = q6asm_compr_copy,
};
-static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
+static int q6asm_dai_pcm_new(struct snd_soc_component *component,
+ struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm_substream *psubstream, *csubstream;
- struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct snd_pcm *pcm = rtd->pcm;
struct device *dev;
int size, ret;
- dev = c->dev;
+ dev = component->dev;
size = q6asm_dai_hardware_playback.buffer_bytes_max;
psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
if (psubstream) {
@@ -494,10 +1221,11 @@
}
}
- return ret;
+ return 0;
}
-static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
+static void q6asm_dai_pcm_free(struct snd_soc_component *component,
+ struct snd_pcm *pcm)
{
struct snd_pcm_substream *substream;
int i;
@@ -512,47 +1240,42 @@
}
}
-static const struct snd_soc_dapm_route afe_pcm_routes[] = {
- {"MM_DL1", NULL, "MultiMedia1 Playback" },
- {"MM_DL2", NULL, "MultiMedia2 Playback" },
- {"MM_DL3", NULL, "MultiMedia3 Playback" },
- {"MM_DL4", NULL, "MultiMedia4 Playback" },
- {"MM_DL5", NULL, "MultiMedia5 Playback" },
- {"MM_DL6", NULL, "MultiMedia6 Playback" },
- {"MM_DL7", NULL, "MultiMedia7 Playback" },
- {"MM_DL7", NULL, "MultiMedia8 Playback" },
- {"MultiMedia1 Capture", NULL, "MM_UL1"},
- {"MultiMedia2 Capture", NULL, "MM_UL2"},
- {"MultiMedia3 Capture", NULL, "MM_UL3"},
- {"MultiMedia4 Capture", NULL, "MM_UL4"},
- {"MultiMedia5 Capture", NULL, "MM_UL5"},
- {"MultiMedia6 Capture", NULL, "MM_UL6"},
- {"MultiMedia7 Capture", NULL, "MM_UL7"},
- {"MultiMedia8 Capture", NULL, "MM_UL8"},
-
+static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
};
-
-static int fe_dai_probe(struct snd_soc_dai *dai)
-{
- struct snd_soc_dapm_context *dapm;
-
- dapm = snd_soc_component_get_dapm(dai->component);
- snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
- ARRAY_SIZE(afe_pcm_routes));
-
- return 0;
-}
-
static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.name = DRV_NAME,
- .ops = &q6asm_dai_ops,
- .pcm_new = q6asm_dai_pcm_new,
- .pcm_free = q6asm_dai_pcm_free,
-
+ .open = q6asm_dai_open,
+ .hw_params = q6asm_dai_hw_params,
+ .close = q6asm_dai_close,
+ .prepare = q6asm_dai_prepare,
+ .trigger = q6asm_dai_trigger,
+ .pointer = q6asm_dai_pointer,
+ .mmap = q6asm_dai_mmap,
+ .pcm_construct = q6asm_dai_pcm_new,
+ .pcm_destruct = q6asm_dai_pcm_free,
+ .compress_ops = &q6asm_dai_compress_ops,
+ .dapm_widgets = q6asm_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets),
};
-static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
Q6ASM_FEDAI_DRIVER(1),
Q6ASM_FEDAI_DRIVER(2),
Q6ASM_FEDAI_DRIVER(3),
@@ -562,6 +1285,54 @@
Q6ASM_FEDAI_DRIVER(7),
Q6ASM_FEDAI_DRIVER(8),
};
+
+static int of_q6asm_parse_dai_data(struct device *dev,
+ struct q6asm_dai_data *pdata)
+{
+ struct snd_soc_dai_driver *dai_drv;
+ struct snd_soc_pcm_stream empty_stream;
+ struct device_node *node;
+ int ret, id, dir, idx = 0;
+
+
+ pdata->num_dais = of_get_child_count(dev->of_node);
+ if (!pdata->num_dais) {
+ dev_err(dev, "No dais found in DT\n");
+ return -EINVAL;
+ }
+
+ pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
+ GFP_KERNEL);
+ if (!pdata->dais)
+ return -ENOMEM;
+
+ memset(&empty_stream, 0, sizeof(empty_stream));
+
+ for_each_child_of_node(dev->of_node, node) {
+ ret = of_property_read_u32(node, "reg", &id);
+ if (ret || id >= MAX_SESSIONS || id < 0) {
+ dev_err(dev, "valid dai id not found:%d\n", ret);
+ continue;
+ }
+
+ dai_drv = &pdata->dais[idx++];
+ *dai_drv = q6asm_fe_dais_template[id];
+
+ ret = of_property_read_u32(node, "direction", &dir);
+ if (ret)
+ continue;
+
+ if (dir == Q6ASM_DAI_RX)
+ dai_drv->capture = empty_stream;
+ else if (dir == Q6ASM_DAI_TX)
+ dai_drv->playback = empty_stream;
+
+ if (of_property_read_bool(node, "is-compress-dai"))
+ dai_drv->compress_new = snd_soc_new_compress;
+ }
+
+ return 0;
+}
static int q6asm_dai_probe(struct platform_device *pdev)
{
@@ -583,16 +1354,21 @@
dev_set_drvdata(dev, pdata);
+ rc = of_q6asm_parse_dai_data(dev, pdata);
+ if (rc)
+ return rc;
+
return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
- q6asm_fe_dais,
- ARRAY_SIZE(q6asm_fe_dais));
+ pdata->dais, pdata->num_dais);
}
+#ifdef CONFIG_OF
static const struct of_device_id q6asm_dai_device_id[] = {
{ .compatible = "qcom,q6asm-dais" },
{},
};
MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
+#endif
static struct platform_driver q6asm_dai_platform_driver = {
.driver = {
--
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