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-rw-r--r--sound/soc/qcom/Kconfig2
-rw-r--r--sound/soc/qcom/apq8016_sbc.c9
-rw-r--r--sound/soc/qcom/apq8096.c6
-rw-r--r--sound/soc/qcom/lpass-platform.c4
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c173
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c243
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.h51
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c21
-rw-r--r--sound/soc/qcom/sdm845.c105
-rw-r--r--sound/soc/qcom/storm.c2
10 files changed, 550 insertions, 66 deletions
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 6530d2462a9e..f51b28d1b94d 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -99,7 +99,7 @@ config SND_SOC_MSM8996
config SND_SOC_SDM845
tristate "SoC Machine driver for SDM845 boards"
- depends on QCOM_APR && CROS_EC && I2C
+ depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE
select SND_SOC_QDSP6
select SND_SOC_QCOM_COMMON
select SND_SOC_RT5663
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index ac75838bbfab..2ef090f4af9e 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -33,9 +33,9 @@ struct apq8016_sbc_data {
static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
struct snd_soc_component *component;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_card *card = rtd->card;
struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card);
int i, rval;
@@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
pdata->jack_setup = true;
}
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
- component = dai->component;
+ component = codec_dai->component;
/* Set default mclk for internal codec */
rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE,
SND_SOC_CLOCK_IN);
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd6846a..d55e3ad96716 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -31,8 +31,8 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
int ret = 0;
@@ -66,7 +66,7 @@ static struct snd_soc_ops apq8096_ops = {
static int apq8096_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/*
* Codec SLIMBUS configuration
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index b05091c283b7..34f7fd1bab1c 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -55,7 +55,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
struct lpass_variant *v = drvdata->variant;
int ret, dma_ch, dir = substream->stream;
@@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
struct snd_pcm_substream *substream;
int i;
- for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (substream) {
snd_dma_free_pages(&substream->dma_buffer);
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index c0d422d0ab94..f6c7cddf08e8 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -41,6 +41,9 @@
#define Q6ASM_DAI_TX 1
#define Q6ASM_DAI_RX 2
+#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,
@@ -69,6 +72,8 @@ struct q6asm_dai_rtd {
};
struct q6asm_dai_data {
+ struct snd_soc_dai_driver *dais;
+ int num_dais;
long long int sid;
};
@@ -250,7 +255,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ 0, prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
@@ -328,7 +333,7 @@ static int q6asm_dai_open(struct snd_soc_component *component,
{
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_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
struct q6asm_dai_rtd *prtd;
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
@@ -540,7 +545,7 @@ 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 snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct q6asm_dai_data *pdata;
struct device *dev = c->dev;
struct q6asm_dai_rtd *prtd;
@@ -627,10 +632,17 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
int dir = stream->direction;
struct q6asm_dai_data *pdata;
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 = c->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_param.codec.options);
@@ -652,7 +664,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
prtd->bits_per_sample = 16;
if (dir == SND_COMPRESS_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, params->codec.id,
- prtd->bits_per_sample);
+ params->codec.profile, prtd->bits_per_sample);
if (ret < 0) {
dev_err(dev, "q6asm_open_write failed\n");
@@ -692,6 +704,126 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
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 = params->codec.sample_rate;
+ wma_cfg.num_channels = params->codec.ch_in;
+ wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+ wma_cfg.block_align = params->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 (params->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",
+ params->codec.profile);
+ return -EIO;
+ }
+
+ if (wma_v9)
+ ret = q6asm_stream_media_format_block_wma_v9(
+ prtd->audio_client, &wma_cfg);
+ else
+ ret = q6asm_stream_media_format_block_wma_v10(
+ prtd->audio_client, &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 = params->codec.sample_rate;
+ alac_cfg.avg_bit_rate = params->codec.bit_rate;
+ alac_cfg.bit_depth = prtd->bits_per_sample;
+ alac_cfg.num_channels = params->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 (params->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,
+ &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 = params->codec.sample_rate;
+ ape_cfg.num_channels = params->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,
+ &ape_cfg);
+ if (ret < 0) {
+ dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
default:
break;
}
@@ -791,9 +923,12 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream,
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 = 2;
+ 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;
}
@@ -889,7 +1024,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.compr_ops = &q6asm_dai_compr_ops,
};
-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),
@@ -903,10 +1038,22 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
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_dai_driver *dai_drv;
struct snd_soc_pcm_stream empty_stream;
struct device_node *node;
- int ret, id, dir;
+ 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));
@@ -917,7 +1064,8 @@ static int of_q6asm_parse_dai_data(struct device *dev,
continue;
}
- dai_drv = &q6asm_fe_dais[id];
+ dai_drv = &pdata->dais[idx++];
+ *dai_drv = q6asm_fe_dais_template[id];
ret = of_property_read_u32(node, "direction", &dir);
if (ret)
@@ -955,11 +1103,12 @@ static int q6asm_dai_probe(struct platform_device *pdev)
dev_set_drvdata(dev, pdata);
- of_q6asm_parse_dai_data(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);
}
static const struct of_device_id q6asm_dai_device_id[] = {
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..0e0e8f7a460a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -39,6 +39,8 @@
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@@ -46,6 +48,8 @@
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_MEDIA_FMT_ALAC 0x00012f31
+#define ASM_MEDIA_FMT_APE 0x00012f32
#define ASM_LEGACY_STREAM_SESSION 0
@@ -104,6 +108,63 @@ struct asm_flac_fmt_blk_v2 {
u16 reserved;
} __packed;
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 advanced_enc_options1;
+ u32 advanced_enc_options2;
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+} __packed;
+
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@@ -858,7 +919,7 @@ err:
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+ u32 codec_profile, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
@@ -894,6 +955,30 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
+ case SND_AUDIOCODEC_WMA:
+ switch (codec_profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+ break;
+ case SND_AUDIOPROFILE_WMA10:
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+ codec_profile);
+ rc = -EINVAL;
+ goto err;
+ }
+ break;
+ case SND_AUDIOCODEC_ALAC:
+ open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+ break;
+ case SND_AUDIOCODEC_APE:
+ open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+ break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@@ -1075,6 +1160,162 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmastdv9_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->reserved = 0;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmaprov10_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->advanced_enc_options1 = cfg->adv_enc_options;
+ fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg)
+{
+ struct asm_alac_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->frame_length = cfg->frame_length;
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->bit_depth = cfg->bit_depth;
+ fmt->num_channels = cfg->num_channels;
+ fmt->max_run = cfg->max_run;
+ fmt->max_frame_bytes = cfg->max_frame_bytes;
+ fmt->avg_bit_rate = cfg->avg_bit_rate;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->channel_layout_tag = cfg->channel_layout_tag;
+ fmt->pb = cfg->pb;
+ fmt->mb = cfg->mb;
+ fmt->kb = cfg->kb;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg)
+{
+ struct asm_ape_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->compression_level = cfg->compression_level;
+ fmt->format_flags = cfg->format_flags;
+ fmt->blocks_per_frame = cfg->blocks_per_frame;
+ fmt->final_frame_blocks = cfg->final_frame_blocks;
+ fmt->total_frames = cfg->total_frames;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->seek_table_present = cfg->seek_table_present;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6764f55f7078..38a207d6cd95 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -45,6 +45,47 @@ struct q6asm_flac_cfg {
u16 md5_sum;
};
+struct q6asm_wma_cfg {
+ u32 fmtag;
+ u32 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u32 block_align;
+ u32 bits_per_sample;
+ u32 channel_mask;
+ u32 enc_options;
+ u32 adv_enc_options;
+ u32 adv_enc_options2;
+};
+
+struct q6asm_alac_cfg {
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+};
+
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
struct audio_client;
@@ -55,7 +96,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags);
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample);
+ u32 codec_profile, uint16_t bits_per_sample);
int q6asm_open_read(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
@@ -69,6 +110,14 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint16_t bits_per_sample);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg);
int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 20724102e85a..46e50612b92c 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -918,25 +918,6 @@ static const struct snd_soc_dapm_route intercon[] = {
{"MM_UL6", NULL, "MultiMedia6 Mixer"},
{"MM_UL7", NULL, "MultiMedia7 Mixer"},
{"MM_UL8", NULL, "MultiMedia8 Mixer"},
-
- {"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_DL8", 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 int routing_hw_params(struct snd_soc_component *component,
@@ -945,7 +926,7 @@ static int routing_hw_params(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct msm_routing_data *data = dev_get_drvdata(component->dev);
- unsigned int be_id = rtd->cpu_dai->id;
+ unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id;
struct session_data *session;
int path_type;
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 3b5547a27aad..b2de65c7f95c 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -11,6 +11,7 @@
#include <sound/pcm_params.h>
#include <sound/jack.h>
#include <sound/soc.h>
+#include <linux/soundwire/sdw.h>
#include <uapi/linux/input-event-codes.h>
#include "common.h"
#include "qdsp6/q6afe.h"
@@ -31,10 +32,12 @@
struct sdm845_snd_data {
struct snd_soc_jack jack;
bool jack_setup;
+ bool stream_prepared[SLIM_MAX_RX_PORTS];
struct snd_soc_card *card;
uint32_t pri_mi2s_clk_count;
uint32_t sec_mi2s_clk_count;
uint32_t quat_tdm_clk_count;
+ struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
};
static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
@@ -43,14 +46,21 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ struct sdw_stream_runtime *sruntime;
u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
int ret = 0, i;
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ sruntime = snd_soc_dai_get_sdw_stream(codec_dai,
+ substream->stream);
+ if (sruntime != ERR_PTR(-ENOTSUPP))
+ pdata->sruntime[cpu_dai->id] = sruntime;
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
if (ret != 0 && ret != -ENOTSUPP) {
@@ -76,7 +86,8 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
int ret = 0, j;
int channels, slot_width;
@@ -125,8 +136,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
}
}
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix, "Left")) {
ret = snd_soc_dai_set_tdm_slot(
@@ -161,8 +171,8 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
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;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret = 0;
switch (cpu_dai->id) {
@@ -210,11 +220,10 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component;
struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
struct snd_jack *jack;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
/*
* Codec SLIMBUS configuration
* RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13
@@ -266,8 +275,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
}
break;
case SLIMBUS_0_RX...SLIMBUS_6_TX:
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ rval = snd_soc_dai_set_channel_map(codec_dai,
ARRAY_SIZE(tx_ch),
tx_ch,
ARRAY_SIZE(rx_ch),
@@ -275,7 +284,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
if (rval != 0 && rval != -ENOTSUPP)
return rval;
- snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0,
+ snd_soc_dai_set_sysclk(codec_dai, 0,
WCD934X_DEFAULT_MCLK_RATE,
SNDRV_PCM_STREAM_PLAYBACK);
}
@@ -295,8 +304,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
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;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int j;
int ret;
@@ -345,8 +354,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
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];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix,
"Left")) {
@@ -386,7 +394,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
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 *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
switch (cpu_dai->id) {
case PRIMARY_MI2S_RX:
@@ -427,8 +435,65 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
}
}
+static int sdm845_snd_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+ int ret;
+
+ if (!sruntime)
+ return 0;
+
+ if (data->stream_prepared[cpu_dai->id]) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ data->stream_prepared[cpu_dai->id] = false;
+ }
+
+ ret = sdw_prepare_stream(sruntime);
+ if (ret)
+ return ret;
+
+ /**
+ * NOTE: there is a strict hw requirement about the ordering of port
+ * enables and actual WSA881x PA enable. PA enable should only happen
+ * after soundwire ports are enabled if not DC on the line is
+ * accumulated resulting in Click/Pop Noise
+ * PA enable/mute are handled as part of codec DAPM and digital mute.
+ */
+
+ ret = sdw_enable_stream(sruntime);
+ if (ret) {
+ sdw_deprepare_stream(sruntime);
+ return ret;
+ }
+ data->stream_prepared[cpu_dai->id] = true;
+
+ return ret;
+}
+
+static int sdm845_snd_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+
+ if (sruntime && data->stream_prepared[cpu_dai->id]) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ data->stream_prepared[cpu_dai->id] = false;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_ops sdm845_be_ops = {
.hw_params = sdm845_snd_hw_params,
+ .hw_free = sdm845_snd_hw_free,
+ .prepare = sdm845_snd_prepare,
.startup = sdm845_snd_startup,
.shutdown = sdm845_snd_shutdown,
};
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index e6666e597265..3a6e18709b9e 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -39,7 +39,7 @@ static int storm_ops_hw_params(struct snd_pcm_substream *substream,
*/
sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT;
- ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0);
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(soc_runtime, 0), 0, sysclk_freq, 0);
if (ret) {
dev_err(card->dev, "error setting sysclk to %u: %d\n",
sysclk_freq, ret);
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-26 17:04:42 +0300