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Commit 18ff677c authored by Ashish Jain's avatar Ashish Jain Committed by Surendar karka
Browse files

ASoc: msm: Enable support for 32 bit format and 384Khz 



Add support for 32 bit data format (SNDRV_PCM_FORMAT_S32_LE)
and sampling rate of 384Khz. Also update platform drivers
to use latest version of pcm media format command
ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4.

CRs-Fixed: 2129947
Change-Id: I6b7ea860a398a7e4dd5f7e23ac3906ff0c6f2b3e
Signed-off-by: default avatarAshish Jain <ashishj@codeaurora.org>
parent b2dba212

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

+10 −4
Original line number Diff line number Diff line
@@ -822,6 +822,10 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream, 
		}



		switch (prtd->codec_param.codec.format) {

		case SNDRV_PCM_FORMAT_S32_LE:

			bit_width = 32;

			sample_word_size = 32;

			break;

		case SNDRV_PCM_FORMAT_S24_LE:

			bit_width = 24;

			sample_word_size = 32;
@@ -836,14 +840,16 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream, 
			sample_word_size = 16;

			break;

		}

		ret = q6asm_media_format_block_pcm_format_support_v3(

		ret = q6asm_media_format_block_pcm_format_support_v4(

							prtd->audio_client,

							prtd->sample_rate,

							prtd->num_channels,

							bit_width, stream_id,

							use_default_chmap,

							chmap,

							sample_word_size);

							sample_word_size,

							ASM_LITTLE_ENDIAN,

							DEFAULT_QF);

		if (ret < 0)

			pr_err("%s: CMD Format block failed\n", __func__);
@@ -1117,7 +1123,7 @@ static int msm_compr_configure_dsp_for_playback 
	} else {

		pr_debug("%s: stream_id %d bits_per_sample %d\n",

				__func__, ac->stream_id, bits_per_sample);

		ret = q6asm_stream_open_write_v3(ac,

		ret = q6asm_stream_open_write_v4(ac,

				prtd->codec, bits_per_sample,

				ac->stream_id,

				prtd->gapless_state.use_dsp_gapless_mode);
@@ -2332,7 +2338,7 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) 


		pr_debug("%s: open_write stream_id %d bits_per_sample %d",

				__func__, stream_id, bits_per_sample);

		rc = q6asm_stream_open_write_v3(prtd->audio_client,

		rc = q6asm_stream_open_write_v4(prtd->audio_client,

				prtd->codec, bits_per_sample,

				stream_id,

				prtd->gapless_state.use_dsp_gapless_mode);

sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c

+28 −13
Original line number Diff line number Diff line
@@ -68,10 +68,11 @@ static struct snd_pcm_hardware msm_pcm_hardware_capture = { 
				SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),

	.formats =              (SNDRV_PCM_FMTBIT_S16_LE |

				SNDRV_PCM_FMTBIT_S24_LE |

				SNDRV_PCM_FMTBIT_S24_3LE),

	.rates =                SNDRV_PCM_RATE_8000_48000,

				SNDRV_PCM_FMTBIT_S24_3LE |

				SNDRV_PCM_FMTBIT_S32_LE),

	.rates =                SNDRV_PCM_RATE_8000_384000,

	.rate_min =             8000,

	.rate_max =             48000,

	.rate_max =             384000,

	.channels_min =         1,

	.channels_max =         4,

	.buffer_bytes_max =     CAPTURE_MAX_NUM_PERIODS *
@@ -91,10 +92,11 @@ static struct snd_pcm_hardware msm_pcm_hardware_playback = { 
				SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),

	.formats =              (SNDRV_PCM_FMTBIT_S16_LE |

				SNDRV_PCM_FMTBIT_S24_LE |

				SNDRV_PCM_FMTBIT_S24_3LE),

	.rates =                SNDRV_PCM_RATE_8000_192000,

				SNDRV_PCM_FMTBIT_S24_3LE |

				SNDRV_PCM_FMTBIT_S32_LE),

	.rates =                SNDRV_PCM_RATE_8000_384000,

	.rate_min =             8000,

	.rate_max =             192000,

	.rate_max =             384000,

	.channels_min =         1,

	.channels_max =         8,

	.buffer_bytes_max =     PLAYBACK_MAX_NUM_PERIODS *
@@ -109,7 +111,7 @@ static struct snd_pcm_hardware msm_pcm_hardware_playback = { 
/* Conventional and unconventional sample rate supported */

static unsigned int supported_sample_rates[] = {

	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,

	88200, 96000, 176400, 192000

	88200, 96000, 176400, 192000, 384000

};



static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
@@ -321,6 +323,10 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream) 
	pr_debug("%s: perf: %x\n", __func__, prtd->audio_client->perf_mode);



	switch (params_format(params)) {

	case SNDRV_PCM_FORMAT_S32_LE:

		bits_per_sample = 32;

		sample_word_size = 32;

		break;

	case SNDRV_PCM_FORMAT_S24_LE:

		bits_per_sample = 24;

		sample_word_size = 32;
@@ -336,7 +342,7 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream) 
		break;

	}



	ret = q6asm_open_write_v3(prtd->audio_client,

	ret = q6asm_open_write_v4(prtd->audio_client,

				  FORMAT_LINEAR_PCM, bits_per_sample);



	if (ret < 0) {
@@ -361,11 +367,12 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream) 
		return ret;

	}



	ret = q6asm_media_format_block_multi_ch_pcm_v3(

	ret = q6asm_media_format_block_multi_ch_pcm_v4(

				prtd->audio_client, runtime->rate,

				runtime->channels, !prtd->set_channel_map,

				prtd->channel_map, bits_per_sample,

				sample_word_size);

				sample_word_size, ASM_LITTLE_ENDIAN,

				DEFAULT_QF);

	if (ret < 0)

		pr_info("%s: CMD Format block failed\n", __func__);
@@ -410,6 +417,8 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
		if ((params_format(params) == SNDRV_PCM_FORMAT_S24_LE) ||

			(params_format(params) == SNDRV_PCM_FORMAT_S24_3LE))

			bits_per_sample = 24;

		else if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE)

			bits_per_sample = 32;



		/* ULL mode is not supported in capture path */

		if (pdata->perf_mode_set == LEGACY_PCM_MODE)
@@ -421,7 +430,7 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
				__func__, params_channels(params),

				prtd->audio_client->perf_mode);



		ret = q6asm_open_read_v3(prtd->audio_client, FORMAT_LINEAR_PCM,

		ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,

				bits_per_sample);

		if (ret < 0) {

			pr_err("%s: q6asm_open_read failed\n", __func__);
@@ -467,6 +476,10 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
		return 0;



	switch (runtime->format) {

	case SNDRV_PCM_FORMAT_S32_LE:

		bits_per_sample = 32;

		sample_word_size = 32;

		break;

	case SNDRV_PCM_FORMAT_S24_LE:

		bits_per_sample = 24;

		sample_word_size = 32;
@@ -485,11 +498,13 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
	pr_debug("%s: Samp_rate = %d Channel = %d bit width = %d, word size = %d\n",

			__func__, prtd->samp_rate, prtd->channel_mode,

			bits_per_sample, sample_word_size);

	ret = q6asm_enc_cfg_blk_pcm_format_support_v3(prtd->audio_client,

	ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,

						      prtd->samp_rate,

						      prtd->channel_mode,

						      bits_per_sample,

						      sample_word_size);

						      sample_word_size,

						      ASM_LITTLE_ENDIAN,

						      DEFAULT_QF);

	if (ret < 0)

		pr_debug("%s: cmd cfg pcm was block failed", __func__);

sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h

+2 −2
Original line number Diff line number Diff line
@@ -59,11 +59,11 @@ struct msm_audio_in_frame_info { 


#define PLAYBACK_MIN_NUM_PERIODS    2

#define PLAYBACK_MAX_NUM_PERIODS    8

#define PLAYBACK_MAX_PERIOD_SIZE    61440

#define PLAYBACK_MAX_PERIOD_SIZE    122880

#define PLAYBACK_MIN_PERIOD_SIZE    128

#define CAPTURE_MIN_NUM_PERIODS     2

#define CAPTURE_MAX_NUM_PERIODS     8

#define CAPTURE_MAX_PERIOD_SIZE     61440

#define CAPTURE_MAX_PERIOD_SIZE     122880

#define CAPTURE_MIN_PERIOD_SIZE     320



struct msm_audio {