This patch adds MP3 playback support in q6asm dais, adding other codec
support should be pretty trivial.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx>
---
sound/soc/qcom/Kconfig | 1 +
sound/soc/qcom/qdsp6/q6asm-dai.c | 372 ++++++++++++++++++++++++++++++-
2 files changed, 372 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 2a4c912d1e48..ebf991bb546c 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -66,6 +66,7 @@ config SND_SOC_QDSP6_ASM
tristate
config SND_SOC_QDSP6_ASM_DAI
+ select SND_SOC_COMPRESS
tristate
config SND_SOC_QDSP6
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 86115de5c1b2..5b986b74dd36 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -10,6 +10,8 @@
#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>
@@ -30,6 +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 Q6ASM_DAI_TX_RX 0
+#define Q6ASM_DAI_TX 1
+#define Q6ASM_DAI_RX 2
+
enum stream_state {
Q6ASM_STREAM_IDLE = 0,
Q6ASM_STREAM_STOPPED,
@@ -38,11 +49,18 @@ enum stream_state {
struct q6asm_dai_rtd {
struct snd_pcm_substream *substream;
+ struct snd_compr_stream *cstream;
+ struct snd_compr_params codec_param;
+ 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;
@@ -137,6 +155,21 @@ static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.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)
{
@@ -460,6 +493,306 @@ static struct snd_pcm_ops q6asm_dai_ops = {
.mmap = q6asm_dai_mmap,
};
+static void compress_event_handler(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_compr_stream *substream = prtd->cstream;
+ unsigned long flags;
+ uint64_t avail;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (!prtd->bytes_sent) {
+ q6asm_write_async(prtd->audio_client, prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ prtd->bytes_sent += prtd->pcm_count;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ break;
+
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ prtd->copied_total += prtd->pcm_count;
+ 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) {
+ q6asm_write_async(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ prtd->bytes_sent += prtd->pcm_count;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
+{
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = c->dev;
+ struct q6asm_dai_rtd *prtd;
+ int stream_id, size, ret;
+
+ stream_id = cpu_dai->driver->id;
+ pdata = snd_soc_component_get_drvdata(c);
+ if (!pdata) {
+ dev_err(dev, "Drv data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (!prtd)
+ return -ENOMEM;
+
+ prtd->cstream = stream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)compress_event_handler,
+ prtd, stream_id, LEGACY_PCM_MODE);
+ if (!prtd->audio_client) {
+ dev_err(dev, "Could not allocate memory\n");
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+ 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");
+ return ret;
+ }
+
+ 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;
+}
+
+static int q6asm_dai_compr_free(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, 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_params(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;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ int dir = stream->direction;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = c->dev;
+ int ret;
+
+ memcpy(&prtd->codec_param, params, sizeof(*params));
+
+ pdata = snd_soc_component_get_drvdata(c);
+ 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, params->codec.id,
+ prtd->bits_per_sample);
+
+ 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_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_trigger(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, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_compr_pointer(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_dai_compr_ack(struct snd_compr_stream *stream,
+ size_t count)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ prtd->bytes_received += count;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return count;
+}
+
+static int q6asm_dai_compr_mmap(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 snd_soc_pcm_runtime *rtd = stream->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ struct device *dev = c->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_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 = 1;
+ caps->codecs[0] = SND_AUDIOCODEC_MP3;
+
+ return 0;
+}
+
+static int q6asm_dai_compr_get_codec_caps(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_compr_ops q6asm_dai_compr_ops = {
+ .open = q6asm_dai_compr_open,
+ .free = q6asm_dai_compr_free,
+ .set_params = q6asm_dai_compr_set_params,
+ .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,
+ .ack = q6asm_dai_compr_ack,
+};
+
static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_pcm_substream *psubstream, *csubstream;
@@ -515,7 +848,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.ops = &q6asm_dai_ops,
.pcm_new = q6asm_dai_pcm_new,
.pcm_free = q6asm_dai_pcm_free,
-
+ .compr_ops = &q6asm_dai_compr_ops,
};
static struct snd_soc_dai_driver q6asm_fe_dais[] = {
@@ -529,6 +862,41 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
Q6ASM_FEDAI_DRIVER(8),
};
+static int of_q6asm_parse_dai_data(struct device *dev,
+ struct q6asm_dai_data *pdata)
+{
+ static struct snd_soc_dai_driver *dai_drv;
+ struct snd_soc_pcm_stream empty_stream;
+ struct device_node *node;
+ int ret, id, dir;
+
+ 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 = &q6asm_fe_dais[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)
{
struct device *dev = &pdev->dev;
@@ -549,6 +917,8 @@ static int q6asm_dai_probe(struct platform_device *pdev)
dev_set_drvdata(dev, pdata);
+ of_q6asm_parse_dai_data(dev, pdata);
+
return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
q6asm_fe_dais,
ARRAY_SIZE(q6asm_fe_dais));
--
2.19.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@xxxxxxxxxxxxxxxx
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
