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Commit b3ca15c1 authored by Mingming Yin's avatar Mingming Yin Committed by Krishnankutty Kolathappilly
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ASoc: msm: Add check for compressed capture format 



- Only AMRWB format is supported in compress capture
  path, use default codec.id(SND_AUDIOCODEC_AMRWB) if
  request codec id is invalid.
- Use fixed band mode and dtx mode since these are the
  only supported parameters for AMRWB format

Change-Id: I45a48952a5833c3e85588b515d38d311064cd7fb
Signed-off-by: default avatarMingming Yin <mingming@codeaurora.org>
Signed-off-by: default avatarKrishnankutty Kolathappilly <kkolat@codeaurora.org>
parent f28186f6

sound/soc/msm/msm-dai-fe.c

+11 −0
Original line number Diff line number Diff line
@@ -324,6 +324,17 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { 
			.rate_min =	8000,

			.rate_max = 192000,

		},

		.capture = {

			.stream_name = "MultiMedia8 Capture",

			.aif_name = "MM_UL8",

			.rates = (SNDRV_PCM_RATE_8000_48000|

					SNDRV_PCM_RATE_KNOT),

			.formats = SNDRV_PCM_FMTBIT_S16_LE,

			.channels_min = 1,

			.channels_max = 8,

			.rate_min =     8000,

			.rate_max =	48000,

		},

		.ops = &msm_fe_Multimedia_dai_ops,

		.name = "MultiMedia8",

		.probe = fe_dai_probe,

sound/soc/msm/msm8974.c

+3 −3
Original line number Diff line number Diff line
@@ -2154,10 +2154,10 @@ static struct snd_soc_dai_link msm8974_common_dai_links[] = { 
		.be_id = MSM_FRONTEND_DAI_MULTIMEDIA7,

	},

	{

		.name = "MSM8974 Compr4",

		.stream_name = "COMPR4",

		.name = "MSM8974 Compr8",

		.stream_name = "COMPR8",

		.cpu_dai_name	= "MultiMedia8",

		.platform_name  = "msm-compress-dsp",

		.platform_name  = "msm-compr-dsp",

		.dynamic = 1,

		.trigger = {SND_SOC_DPCM_TRIGGER_POST,

			 SND_SOC_DPCM_TRIGGER_POST},

sound/soc/msm/qdsp6v2/Makefile

+3 −2
Original line number Diff line number Diff line 
snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \

			msm-compress-q6-v2.o msm-multi-ch-pcm-q6-v2.o \

			msm-pcm-lpa-v2.o msm-pcm-afe-v2.o msm-pcm-voip-v2.o \

			msm-compress-q6-v2.o msm-compr-q6-v2.o \

			msm-multi-ch-pcm-q6-v2.o msm-pcm-lpa-v2.o \

			msm-pcm-afe-v2.o msm-pcm-voip-v2.o \

			msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \

			msm-lsm-client.o msm-pcm-host-voice-v2.o \

			msm-audio-effects-q6-v2.o

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

+57 −4
Original line number Diff line number Diff line
@@ -52,6 +52,9 @@ 
#define COMPRESSED_LR_VOL_MAX_STEPS	0x20002000



#define MAX_AC3_PARAM_SIZE		(18*2*sizeof(int))

#define AMR_WB_BAND_MODE 8

#define AMR_WB_DTX_MODE 0





const DECLARE_TLV_DB_LINEAR(compr_rx_vol_gain, 0,

			    COMPRESSED_LR_VOL_MAX_STEPS);
@@ -108,12 +111,30 @@ static unsigned int supported_sample_rates[] = { 
	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000

};



/* Add supported codecs for compress capture path */

static uint32_t supported_compr_capture_codecs[] = {

	SND_AUDIOCODEC_AMRWB

};



static struct snd_pcm_hw_constraint_list constraints_sample_rates = {

	.count = ARRAY_SIZE(supported_sample_rates),

	.list = supported_sample_rates,

	.mask = 0,

};



static bool msm_compr_capture_codecs(uint32_t req_codec)

{

	int i;

	pr_debug("%s req_codec:%d\n", __func__, req_codec);

	if (req_codec == 0)

		return false;

	for (i = 0; i < ARRAY_SIZE(supported_compr_capture_codecs); i++) {

		if (req_codec == supported_compr_capture_codecs[i])

			return true;

	}

	return false;

}



static void compr_event_handler(uint32_t opcode,

		uint32_t token, uint32_t *payload, void *priv)

{
@@ -428,6 +449,13 @@ static int msm_compr_capture_prepare(struct snd_pcm_substream *substream) 
	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);

	prtd->pcm_irq_pos = 0;



	if (!msm_compr_capture_codecs(codec->id)) {

		/*

		 * request codec invalid or not supported,

		 * use default compress format

		 */

		codec->id = SND_AUDIOCODEC_AMRWB;

	}

	/* rate and channels are sent to audio driver */

	prtd->samp_rate = runtime->rate;

	prtd->channel_mode = runtime->channels;
@@ -441,8 +469,13 @@ static int msm_compr_capture_prepare(struct snd_pcm_substream *substream) 
		pr_debug("SND_AUDIOCODEC_AMRWB\n");

		ret = q6asm_enc_cfg_blk_amrwb(prtd->audio_client,

			MAX_NUM_FRAMES_PER_BUFFER,

			codec->options.generic.reserved[0] /*bitrate 0-8*/,

			codec->options.generic.reserved[1] /*dtx mode 0/1*/);

			/*

			 * use fixed band mode and dtx mode

			 * band mode - 23.85 kbps

			 */

			AMR_WB_BAND_MODE,

			/* dtx mode - disable */

			AMR_WB_DTX_MODE);

		if (ret < 0)

			pr_err("%s: CMD Format block" \

				"failed: %d\n", __func__, ret);
@@ -500,6 +533,15 @@ static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd) 
		prtd->pcm_irq_pos = 0;



		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {

			if (!msm_compr_capture_codecs(

				compr->info.codec_param.codec.id)) {

				/*

				 * request codec invalid or not supported,

				 * use default compress format

				 */

				compr->info.codec_param.codec.id =

				SND_AUDIOCODEC_AMRWB;

			}

			switch (compr->info.codec_param.codec.id) {

			case SND_AUDIOCODEC_AMRWB:

				break;
@@ -815,8 +857,10 @@ static int msm_compr_hw_params(struct snd_pcm_substream *substream, 
			prtd->audio_client->perf_mode,

			prtd->session_id,

			substream->stream);

		/* the number of channels are required to call volume api

		   accoridngly. So, get channels from hw params */

		/*

		 * the number of channels are required to call volume api

		 * accoridngly. So, get channels from hw params

		 */

		if ((params_channels(params) > 0) &&

		    (params_periods(params) <= runtime->hw.channels_max))

			prtd->channel_mode = params_channels(params);
@@ -830,6 +874,15 @@ static int msm_compr_hw_params(struct snd_pcm_substream *substream, 
			pr_err("%s: Send SoftVolume Param failed ret=%d\n",

				__func__, ret);

	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {

		if (!msm_compr_capture_codecs(

			compr->info.codec_param.codec.id)) {

			/*

			 * request codec invalid or not supported,

			 * use default compress format

			 */

			compr->info.codec_param.codec.id =

				SND_AUDIOCODEC_AMRWB;

		}

		switch (compr->info.codec_param.codec.id) {

		case SND_AUDIOCODEC_AMRWB:

			pr_debug("q6asm_open_read(FORMAT_AMRWB)\n");

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

+36 −0
Original line number Diff line number Diff line
@@ -2105,6 +2105,31 @@ static const struct snd_kcontrol_new mmul5_mixer_controls[] = { 
	msm_routing_put_audio_mixer),

};



static const struct snd_kcontrol_new mmul8_mixer_controls[] = {

	SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

	SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,

	MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,

	msm_routing_put_audio_mixer),

};





static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = {

	SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX,

	MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
@@ -3221,6 +3246,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 
	SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_OUT("MM_UL9", "MultiMedia9 Capture", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0),

	SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0),
@@ -3400,6 +3426,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 
	mmul4_mixer_controls, ARRAY_SIZE(mmul4_mixer_controls)),

	SND_SOC_DAPM_MIXER("MultiMedia5 Mixer", SND_SOC_NOPM, 0, 0,

	mmul5_mixer_controls, ARRAY_SIZE(mmul5_mixer_controls)),

	SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,

	mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),

	SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,

	auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)),

	SND_SOC_DAPM_MIXER("SEC_AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
@@ -3594,12 +3622,16 @@ static const struct snd_soc_dapm_route intercon[] = { 


	{"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},

	{"MultiMedia4 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},

	{"MultiMedia8 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},

	{"MultiMedia1 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},

	{"MultiMedia4 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},

	{"MultiMedia8 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},

	{"MultiMedia1 Mixer", "SLIM_4_TX", "SLIMBUS_4_TX"},

	{"MultiMedia1 Mixer", "SLIM_6_TX", "SLIMBUS_6_TX"},

	{"MultiMedia4 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},

	{"MultiMedia8 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},

	{"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},

	{"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},

	{"MultiMedia5 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},

	{"MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},

	{"MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
@@ -3696,18 +3728,22 @@ static const struct snd_soc_dapm_route intercon[] = { 
	{"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},

	{"MultiMedia4 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},

	{"MultiMedia5 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},

	{"MultiMedia8 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},

	{"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},

	{"MultiMedia4 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},

	{"MultiMedia5 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},

	{"MultiMedia8 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},



	{"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},

	{"MultiMedia4 Mixer", "AFE_PCM_TX", "PCM_TX"},

	{"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"},

	{"MultiMedia8 Mixer", "AFE_PCM_TX", "PCM_TX"},

	{"MM_UL1", NULL, "MultiMedia1 Mixer"},

	{"MultiMedia2 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},

	{"MM_UL2", NULL, "MultiMedia2 Mixer"},

	{"MM_UL4", NULL, "MultiMedia4 Mixer"},

	{"MM_UL5", NULL, "MultiMedia5 Mixer"},

	{"MM_UL8", NULL, "MultiMedia8 Mixer"},



	{"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},

	{"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},