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Commit 4c73d9cc authored by Linux Build Service Account's avatar Linux Build Service Account Committed by Gerrit - the friendly Code Review server
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Merge changes I914c68a9,I2003e40c into msm-4.4 

* changes:
  ASoC: msm: qdsp6v2: latency mode support for transcode loopback
  ASoC: msm: volume control support for DSP transcode loopback
parents a5a9ec77 97390bee

include/uapi/sound/compress_offload.h

+7 −0
Original line number Diff line number Diff line
@@ -138,6 +138,11 @@ struct snd_compr_audio_info { 
#define SNDRV_COMPRESS_CLK_REC_MODE_NONE 0

#define SNDRV_COMPRESS_CLK_REC_MODE_AUTO 1



enum sndrv_compress_latency_mode {

	SNDRV_COMPRESS_LEGACY_LATENCY_MODE = 0,

	SNDRV_COMPRESS_LOW_LATENCY_MODE = 1,

};



/**

 * enum sndrv_compress_encoder

 * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the
@@ -164,6 +169,7 @@ enum sndrv_compress_encoder { 
	SNDRV_COMPRESS_START_DELAY = 9,

	SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK = 10,

	SNDRV_COMPRESS_ADJUST_SESSION_CLOCK = 11,

	SNDRV_COMPRESS_LATENCY_MODE = 12,

};



#define SNDRV_COMPRESS_PATH_DELAY SNDRV_COMPRESS_PATH_DELAY
@@ -174,6 +180,7 @@ enum sndrv_compress_encoder { 
#define SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK \

			SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK

#define SNDRV_COMPRESS_ADJUST_SESSION_CLOCK SNDRV_COMPRESS_ADJUST_SESSION_CLOCK

#define SNDRV_COMPRESS_LATENCY_MODE SNDRV_COMPRESS_LATENCY_MODE



/**

 * struct snd_compr_metadata - compressed stream metadata

sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c

+173 −3
Original line number Diff line number Diff line
@@ -28,6 +28,7 @@ 
#include <sound/control.h>

#include <sound/q6asm-v2.h>

#include <sound/q6core.h>

#include <sound/q6audio-v2.h>

#include <sound/pcm_params.h>

#include <sound/timer.h>

#include <sound/tlv.h>
@@ -41,6 +42,8 @@ 
#include "msm-qti-pp-config.h"



#define LOOPBACK_SESSION_MAX_NUM_STREAMS 2

/* Max volume corresponding to 24dB */

#define TRANSCODE_LR_VOL_MAX_STEPS 0xFFFF



#define APP_TYPE_CONFIG_IDX_APP_TYPE 0

#define APP_TYPE_CONFIG_IDX_ACDB_ID 1
@@ -52,6 +55,8 @@ static DEFINE_MUTEX(transcode_loopback_session_lock); 
struct trans_loopback_pdata {

	struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX];

	int32_t ion_fd[MSM_FRONTEND_DAI_MAX];

	uint32_t master_gain;

	int perf_mode;

};



struct loopback_stream {
@@ -403,6 +408,8 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 
	struct msm_transcode_loopback *trans = runtime->private_data;

	struct snd_soc_pcm_runtime *soc_pcm_rx;

	struct snd_soc_pcm_runtime *soc_pcm_tx;

	struct snd_soc_pcm_runtime *rtd;

	struct trans_loopback_pdata *pdata;

	uint32_t bit_width = 16;

	int ret = 0;
@@ -413,6 +420,9 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 


	mutex_lock(&trans->lock);



	rtd = snd_pcm_substream_chip(cstream);

	pdata = snd_soc_platform_get_drvdata(rtd->platform);



	if (cstream->direction == SND_COMPRESS_PLAYBACK) {

		if (codec_param->codec.id == SND_AUDIOCODEC_PCM) {

			trans->sink.codec_format =
@@ -494,7 +504,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 
		pr_debug("%s: ASM client allocated, callback %pK\n", __func__,

						loopback_event_handler);

		trans->session_id = trans->audio_client->session;

		trans->audio_client->perf_mode = false;

		trans->audio_client->perf_mode = pdata->perf_mode;

		ret = q6asm_open_transcode_loopback(trans->audio_client,

					bit_width,

					trans->source.codec_format,
@@ -513,7 +523,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 
		if (trans->source.codec_format != FORMAT_LINEAR_PCM)

			msm_pcm_routing_reg_phy_compr_stream(

					soc_pcm_tx->dai_link->be_id,

					trans->audio_client->perf_mode,

					false,

					trans->session_id,

					SNDRV_PCM_STREAM_CAPTURE,

					COMPRESSED_PASSTHROUGH_GEN);
@@ -526,7 +536,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 
		/* Opening Rx ADM in LOW_LATENCY mode by default */

		msm_pcm_routing_reg_phy_stream(

					soc_pcm_rx->dai_link->be_id,

					true,

					trans->audio_client->perf_mode,

					trans->session_id,

					SNDRV_PCM_STREAM_PLAYBACK);

		pr_debug("%s: Successfully opened ADM sessions\n", __func__);
@@ -559,6 +569,46 @@ static int msm_transcode_loopback_get_caps(struct snd_compr_stream *cstream, 
	return 0;

}



static int msm_transcode_loopback_set_metadata(struct snd_compr_stream *cstream,

				struct snd_compr_metadata *metadata)

{

	struct snd_soc_pcm_runtime *rtd;

	struct trans_loopback_pdata *pdata;



	if (!metadata || !cstream) {

		pr_err("%s: Invalid arguments\n", __func__);

		return -EINVAL;

	}



	rtd = snd_pcm_substream_chip(cstream);

	pdata = snd_soc_platform_get_drvdata(rtd->platform);



	switch (metadata->key) {

	case SNDRV_COMPRESS_LATENCY_MODE:

	{

		switch (metadata->value[0]) {

		case SNDRV_COMPRESS_LEGACY_LATENCY_MODE:

			pdata->perf_mode = LEGACY_PCM_MODE;

			break;

		case SNDRV_COMPRESS_LOW_LATENCY_MODE:

			pdata->perf_mode = LOW_LATENCY_PCM_MODE;

			break;

		default:

			pr_debug("%s: Unsupported latency mode %d, default to Legacy\n",

					__func__, metadata->value[0]);

			pdata->perf_mode = LEGACY_PCM_MODE;

			break;

		}

	}

		break;

	default:

		pr_debug("%s: Unsupported metadata %d\n",

				__func__, metadata->key);

		break;

	}

	return 0;

}



static int msm_transcode_stream_cmd_put(struct snd_kcontrol *kcontrol,

				struct snd_ctl_elem_value *ucontrol)

{
@@ -813,6 +863,80 @@ done: 
	return ret;

}



static int msm_transcode_set_volume(struct snd_compr_stream *cstream,

				uint32_t master_gain)

{

	int rc = 0;

	struct msm_transcode_loopback *prtd;

	struct snd_soc_pcm_runtime *rtd;



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

	if (!cstream || !cstream->runtime) {

		pr_err("%s: session not active\n", __func__);

		return -EPERM;

	}

	rtd = cstream->private_data;

	prtd = cstream->runtime->private_data;



	if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) {

		pr_err("%s: invalid rtd, prtd or audio client", __func__);

		return -EINVAL;

	}



	rc = q6asm_set_volume(prtd->audio_client, master_gain);

	if (rc < 0)

		pr_err("%s: Send vol gain command failed rc=%d\n",

		       __func__, rc);



	return rc;

}



static int msm_transcode_volume_put(struct snd_kcontrol *kcontrol,

				struct snd_ctl_elem_value *ucontrol)

{

	struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);

	unsigned long fe_id = kcontrol->private_value;

	struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *)

			snd_soc_component_get_drvdata(comp);

	struct snd_compr_stream *cstream = NULL;

	uint32_t ret = 0;



	if (fe_id >= MSM_FRONTEND_DAI_MAX) {

		pr_err("%s Received out of bounds fe_id %lu\n",

			__func__, fe_id);

		return -EINVAL;

	}



	cstream = pdata->cstream[fe_id];

	pdata->master_gain = ucontrol->value.integer.value[0];



	pr_debug("%s: fe_id %lu master_gain %d\n",

		 __func__, fe_id, pdata->master_gain);

	if (cstream)

		ret = msm_transcode_set_volume(cstream, pdata->master_gain);

	return ret;

}



static int msm_transcode_volume_get(struct snd_kcontrol *kcontrol,

				struct snd_ctl_elem_value *ucontrol)

{

	struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);

	unsigned long fe_id = kcontrol->private_value;



	struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *)

			snd_soc_component_get_drvdata(comp);



	if (fe_id >= MSM_FRONTEND_DAI_MAX) {

		pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id);

		return -EINVAL;

	}



	pr_debug("%s: fe_id %lu\n", __func__, fe_id);

	ucontrol->value.integer.value[0] = pdata->master_gain;



	return 0;

}



static int msm_transcode_stream_cmd_control(

			struct snd_soc_pcm_runtime *rtd)

{
@@ -1089,6 +1213,7 @@ static int msm_transcode_add_app_type_cfg_control( 


	if (!rtd) {

		pr_err("%s NULL rtd\n", __func__);



		return -EINVAL;

	}
@@ -1114,6 +1239,44 @@ static int msm_transcode_add_app_type_cfg_control( 


	return rc;

}

static int msm_transcode_volume_info(struct snd_kcontrol *kcontrol,

				 struct snd_ctl_elem_info *uinfo)

{

	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

	uinfo->count = 1;

	uinfo->value.integer.min = 0;

	uinfo->value.integer.max = TRANSCODE_LR_VOL_MAX_STEPS;

	return 0;

}



static int msm_transcode_add_volume_control(struct snd_soc_pcm_runtime *rtd)

{

	struct snd_kcontrol_new fe_volume_control[1] = {

		{

		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,

		.name = "Transcode Loopback Rx Volume",

		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |

			  SNDRV_CTL_ELEM_ACCESS_READWRITE,

		.info = msm_transcode_volume_info,

		.get = msm_transcode_volume_get,

		.put = msm_transcode_volume_put,

		.private_value = 0,

		}

	};



	if (!rtd) {

		pr_err("%s NULL rtd\n", __func__);

		return -EINVAL;

	}

	if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) {

		fe_volume_control[0].private_value = rtd->dai_link->be_id;

		pr_debug("Registering new mixer ctl %s",

			     fe_volume_control[0].name);

		snd_soc_add_platform_controls(rtd->platform, fe_volume_control,

						ARRAY_SIZE(fe_volume_control));

	}

	return 0;

}



static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd)

{
@@ -1148,6 +1311,11 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) 
		pr_err("%s: Could not add Compr App Type Cfg Control\n",

			__func__);



	rc = msm_transcode_add_volume_control(rtd);

	if (rc)

		pr_err("%s: Could not add transcode volume Control\n",

			__func__);



	return 0;

}
@@ -1157,6 +1325,7 @@ static struct snd_compr_ops msm_transcode_loopback_ops = { 
	.trigger		= msm_transcode_loopback_trigger,

	.set_params		= msm_transcode_loopback_set_params,

	.get_caps		= msm_transcode_loopback_get_caps,

	.set_metadata		= msm_transcode_loopback_set_metadata,

};
@@ -1171,6 +1340,7 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) 
	if (!pdata)

		return -ENOMEM;



	pdata->perf_mode = LOW_LATENCY_PCM_MODE;

	snd_soc_platform_set_drvdata(platform, pdata);

	return 0;

}