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-rw-r--r--sound/soc/qcom/Kconfig2
-rw-r--r--sound/soc/qcom/lpass-platform.c2
-rw-r--r--sound/soc/qcom/qdsp6/q6afe-dai.c30
-rw-r--r--sound/soc/qcom/qdsp6/q6afe.c4
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c372
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c5
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c9
-rw-r--r--sound/soc/qcom/sdm845.c186
8 files changed, 596 insertions, 14 deletions
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 2a4c912d1e48..804ae0d93058 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
@@ -100,6 +101,7 @@ config SND_SOC_SDM845
depends on QCOM_APR
select SND_SOC_QDSP6
select SND_SOC_QCOM_COMMON
+ select SND_SOC_RT5663
help
To add support for audio on Qualcomm Technologies Inc.
SDM845 SoC-based systems.
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index d07271ea4c45..028bce671cbc 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -91,7 +91,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream)
if (ret) {
dev_err(soc_runtime->dev,
"error writing to rdmactl reg: %d\n", ret);
- return ret;
+ return ret;
}
data->dma_ch = dma_ch;
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 8f6c8fc073a9..dc645ba4d8d0 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -341,6 +341,7 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
switch (dai->id) {
case HDMI_RX:
+ case DISPLAY_PORT_RX:
q6afe_hdmi_port_prepare(dai_data->port[dai->id],
&dai_data->port_config[dai->id].hdmi);
break;
@@ -445,6 +446,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai,
static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
{"HDMI Playback", NULL, "HDMI_RX"},
+ {"Display Port Playback", NULL, "DISPLAY_PORT_RX"},
{"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"},
{"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"},
{"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"},
@@ -561,13 +563,13 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
{"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"},
};
-static struct snd_soc_dai_ops q6hdmi_ops = {
+static const struct snd_soc_dai_ops q6hdmi_ops = {
.prepare = q6afe_dai_prepare,
.hw_params = q6hdmi_hw_params,
.shutdown = q6afe_dai_shutdown,
};
-static struct snd_soc_dai_ops q6i2s_ops = {
+static const struct snd_soc_dai_ops q6i2s_ops = {
.prepare = q6afe_dai_prepare,
.hw_params = q6i2s_hw_params,
.set_fmt = q6i2s_set_fmt,
@@ -575,14 +577,14 @@ static struct snd_soc_dai_ops q6i2s_ops = {
.set_sysclk = q6afe_mi2s_set_sysclk,
};
-static struct snd_soc_dai_ops q6slim_ops = {
+static const struct snd_soc_dai_ops q6slim_ops = {
.prepare = q6afe_dai_prepare,
.hw_params = q6slim_hw_params,
.shutdown = q6afe_dai_shutdown,
.set_channel_map = q6slim_set_channel_map,
};
-static struct snd_soc_dai_ops q6tdm_ops = {
+static const struct snd_soc_dai_ops q6tdm_ops = {
.prepare = q6afe_dai_prepare,
.shutdown = q6afe_dai_shutdown,
.set_sysclk = q6afe_mi2s_set_sysclk,
@@ -1090,6 +1092,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
Q6AFE_TDM_CAP_DAI("Quinary", 5, QUINARY_TDM_TX_5),
Q6AFE_TDM_CAP_DAI("Quinary", 6, QUINARY_TDM_TX_6),
Q6AFE_TDM_CAP_DAI("Quinary", 7, QUINARY_TDM_TX_7),
+ {
+ .playback = {
+ .stream_name = "Display Port Playback",
+ .rates = SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 2,
+ .channels_max = 8,
+ .rate_max = 192000,
+ .rate_min = 48000,
+ },
+ .ops = &q6hdmi_ops,
+ .id = DISPLAY_PORT_RX,
+ .name = "DISPLAY_PORT",
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ },
};
static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
@@ -1311,6 +1332,7 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0),
};
static const struct snd_soc_component_driver q6afe_dai_component = {
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 829b5e987b2a..e0945f7a58c8 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -71,6 +71,7 @@
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
/* Clock set API version */
@@ -704,6 +705,8 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
QUINARY_TDM_RX_7, 1, 1},
[QUINARY_TDM_TX_7] = { AFE_PORT_ID_QUINARY_TDM_TX_7,
QUINARY_TDM_TX_7, 0, 1},
+ [DISPLAY_PORT_RX] = { AFE_PORT_ID_HDMI_OVER_DP_RX,
+ DISPLAY_PORT_RX, 1, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -1384,6 +1387,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
switch (port_id) {
case AFE_PORT_ID_MULTICHAN_HDMI_RX:
+ case AFE_PORT_ID_HDMI_OVER_DP_RX:
cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
break;
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
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));
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e1cfa846a1dc..4f85cb19a309 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -12,6 +12,7 @@
#include <linux/kref.h>
#include <linux/of.h>
#include <uapi/sound/asound.h>
+#include <uapi/sound/compress_params.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@@ -36,6 +37,7 @@
#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@@ -868,6 +870,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
switch (format) {
+ case SND_AUDIOCODEC_MP3:
+ open->dec_fmt_id = ASM_MEDIA_FMT_MP3;
+ break;
case FORMAT_LINEAR_PCM:
open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index d61b8404f7da..ddcd9978cf57 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -453,6 +453,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
Q6ROUTING_RX_MIXERS(HDMI_RX) };
+static const struct snd_kcontrol_new display_port_mixer_controls[] = {
+ Q6ROUTING_RX_MIXERS(DISPLAY_PORT_RX) };
+
static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
Q6ROUTING_RX_MIXERS(PRIMARY_MI2S_RX) };
@@ -655,6 +658,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
hdmi_mixer_controls,
ARRAY_SIZE(hdmi_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DISPLAY_PORT_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ display_port_mixer_controls,
+ ARRAY_SIZE(display_port_mixer_controls)),
+
SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
slimbus_rx_mixer_controls,
ARRAY_SIZE(slimbus_rx_mixer_controls)),
@@ -833,6 +840,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
static const struct snd_soc_dapm_route intercon[] = {
Q6ROUTING_RX_DAPM_ROUTE("HDMI Mixer", "HDMI_RX"),
+ Q6ROUTING_RX_DAPM_ROUTE("DISPLAY_PORT_RX Audio Mixer",
+ "DISPLAY_PORT_RX"),
Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_0_RX Audio Mixer", "SLIMBUS_0_RX"),
Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_1_RX Audio Mixer", "SLIMBUS_1_RX"),
Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_2_RX Audio Mixer", "SLIMBUS_2_RX"),
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 9effbecc571f..1db8ef668223 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -6,18 +6,31 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/of_device.h>
+#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <uapi/linux/input-event-codes.h>
#include "common.h"
#include "qdsp6/q6afe.h"
+#include "../codecs/rt5663.h"
#define DEFAULT_SAMPLE_RATE_48K 48000
#define DEFAULT_MCLK_RATE 24576000
-#define DEFAULT_BCLK_RATE 12288000
+#define TDM_BCLK_RATE 6144000
+#define MI2S_BCLK_RATE 1536000
+#define LEFT_SPK_TDM_TX_MASK 0x30
+#define RIGHT_SPK_TDM_TX_MASK 0xC0
+#define SPK_TDM_RX_MASK 0x03
+#define NUM_TDM_SLOTS 8
struct sdm845_snd_data {
+ struct snd_soc_jack jack;
+ bool jack_setup;
struct snd_soc_card *card;
uint32_t pri_mi2s_clk_count;
+ uint32_t sec_mi2s_clk_count;
uint32_t quat_tdm_clk_count;
};
@@ -28,12 +41,12 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret = 0;
+ int ret = 0, j;
int channels, slot_width;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- slot_width = 32;
+ slot_width = 16;
break;
default:
dev_err(rtd->dev, "%s: invalid param format 0x%x\n",
@@ -75,6 +88,35 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
goto end;
}
}
+
+ for (j = 0; j < rtd->num_codecs; j++) {
+ struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+
+ if (!strcmp(codec_dai->component->name_prefix, "Left")) {
+ ret = snd_soc_dai_set_tdm_slot(
+ codec_dai, LEFT_SPK_TDM_TX_MASK,
+ SPK_TDM_RX_MASK, NUM_TDM_SLOTS,
+ slot_width);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "DEV0 TDM slot err:%d\n", ret);
+ return ret;
+ }
+ }
+
+ if (!strcmp(codec_dai->component->name_prefix, "Right")) {
+ ret = snd_soc_dai_set_tdm_slot(
+ codec_dai, RIGHT_SPK_TDM_TX_MASK,
+ SPK_TDM_RX_MASK, NUM_TDM_SLOTS,
+ slot_width);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "DEV1 TDM slot err:%d\n", ret);
+ return ret;
+ }
+ }
+ }
+
end:
return ret;
}
@@ -84,9 +126,27 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret = 0;
switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ case PRIMARY_MI2S_TX:
+ /*
+ * Use ASRC for internal clocks, as PLL rate isn't multiple
+ * of BCLK.
+ */
+ rt5663_sel_asrc_clk_src(
+ codec_dai->component,
+ RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER,
+ RT5663_CLK_SEL_I2S1_ASRC);
+ ret = snd_soc_dai_set_sysclk(
+ codec_dai, RT5663_SCLK_S_MCLK, DEFAULT_MCLK_RATE,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(rtd->dev,
+ "snd_soc_dai_set_sysclk err = %d\n", ret);
+ break;
case QUATERNARY_TDM_RX_0:
case QUATERNARY_TDM_TX_0:
ret = sdm845_tdm_snd_hw_params(substream, params);
@@ -98,24 +158,87 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *component;
+ struct snd_soc_dai_link *dai_link = rtd->dai_link;
+ struct snd_soc_card *card = rtd->card;
+ struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
+ int i, rval;
+
+ if (!pdata->jack_setup) {
+ struct snd_jack *jack;
+
+ rval = snd_soc_card_jack_new(card, "Headset Jack",
+ SND_JACK_HEADSET |
+ SND_JACK_HEADPHONE |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3,
+ &pdata->jack, NULL, 0);
+
+ if (rval < 0) {
+ dev_err(card->dev, "Unable to add Headphone Jack\n");
+ return rval;
+ }
+
+ jack = pdata->jack.jack;
+
+ snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+ snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+ snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+ pdata->jack_setup = true;
+ }
+
+ for (i = 0 ; i < dai_link->num_codecs; i++) {
+ struct snd_soc_dai *dai = rtd->codec_dais[i];
+
+ component = dai->component;
+ rval = snd_soc_component_set_jack(
+ component, &pdata->jack, NULL);
+ if (rval != 0 && rval != -ENOTSUPP) {
+ dev_warn(card->dev, "Failed to set jack: %d\n", rval);
+ return rval;
+ }
+ }
+
+ return 0;
+}
+
+
static int sdm845_snd_startup(struct snd_pcm_substream *substream)
{
unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
+ unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int j;
+ int ret;
switch (cpu_dai->id) {
case PRIMARY_MI2S_RX:
case PRIMARY_MI2S_TX:
+ codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF;
if (++(data->pri_mi2s_clk_count) == 1) {
snd_soc_dai_set_sysclk(cpu_dai,
Q6AFE_LPASS_CLK_ID_MCLK_1,
DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
snd_soc_dai_set_sysclk(cpu_dai,
Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
- DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ snd_soc_dai_set_fmt(cpu_dai, fmt);
+ snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt);
+ break;
+
+ case SECONDARY_MI2S_TX:
+ if (++(data->sec_mi2s_clk_count) == 1) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT,
+ MI2S_BCLK_RATE, SNDRV_PCM_STREAM_CAPTURE);
}
snd_soc_dai_set_fmt(cpu_dai, fmt);
break;
@@ -125,7 +248,35 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
if (++(data->quat_tdm_clk_count) == 1) {
snd_soc_dai_set_sysclk(cpu_dai,
Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
- DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ TDM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+ codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B;
+
+ for (j = 0; j < rtd->num_codecs; j++) {
+ codec_dai = rtd->codec_dais[j];
+
+ if (!strcmp(codec_dai->component->name_prefix,
+ "Left")) {
+ ret = snd_soc_dai_set_fmt(
+ codec_dai, codec_dai_fmt);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "Left TDM fmt err:%d\n", ret);
+ return ret;
+ }
+ }
+
+ if (!strcmp(codec_dai->component->name_prefix,
+ "Right")) {
+ ret = snd_soc_dai_set_fmt(
+ codec_dai, codec_dai_fmt);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "Right TDM slot err:%d\n", ret);
+ return ret;
+ }
+ }
}
break;
@@ -156,6 +307,14 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
};
break;
+ case SECONDARY_MI2S_TX:
+ if (--(data->sec_mi2s_clk_count) == 0) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT,
+ 0, SNDRV_PCM_STREAM_CAPTURE);
+ }
+ break;
+
case QUATERNARY_TDM_RX_0:
case QUATERNARY_TDM_TX_0:
if (--(data->quat_tdm_clk_count) == 0) {
@@ -171,7 +330,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
}
}
-static struct snd_soc_ops sdm845_be_ops = {
+static const struct snd_soc_ops sdm845_be_ops = {
.hw_params = sdm845_snd_hw_params,
.startup = sdm845_snd_startup,
.shutdown = sdm845_snd_shutdown,
@@ -193,7 +352,15 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
-static void sdm845_add_be_ops(struct snd_soc_card *card)
+static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+};
+
+static void sdm845_add_ops(struct snd_soc_card *card)
{
struct snd_soc_dai_link *link;
int i;
@@ -203,6 +370,7 @@ static void sdm845_add_be_ops(struct snd_soc_card *card)
link->ops = &sdm845_be_ops;
link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
}
+ link->init = sdm845_dai_init;
}
}
@@ -224,6 +392,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
goto data_alloc_fail;
}
+ card->dapm_widgets = sdm845_snd_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
card->dev = dev;
dev_set_drvdata(dev, card);
ret = qcom_snd_parse_of(card);
@@ -235,7 +405,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
data->card = card;
snd_soc_card_set_drvdata(card, data);
- sdm845_add_be_ops(card);
+ sdm845_add_ops(card);
ret = snd_soc_register_card(card);
if (ret) {
dev_err(dev, "Sound card registration failed\n");
generated by cgit v1.2.3 (git 2.39.1) at 2024-04-28 13:37:10 +0000