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Commit 0be40a35 authored by Damir Didjusto's avatar Damir Didjusto Committed by Gerrit - the friendly Code Review server
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ASoC: msm: Handle data loss at device switch for FM 



- When device switch happens there are underflow errors
in LPASS. And there is some break in audio observed for
around 2 sec.
- To handle this, ASM run command needs to be sent again
to DSP at device switch.

Change-Id: I7d5e279796db190c454b3d1b39ad9add8f933319
Signed-off-by: default avatarDamir Didjusto <damird@codeaurora.org>
Signed-off-by: default avatarVimal Puthanveed <vimalp@codeaurora.org>
parent 444ef0c3

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

+25 −2
Original line number Diff line number Diff line
@@ -68,6 +68,25 @@ static const struct snd_pcm_hardware dummy_pcm_hardware = { 
	.periods_max            = 128,

};



static void msm_pcm_route_event_handler(enum msm_pcm_routing_event event,

					void *priv_data)

{

	struct msm_pcm_loopback *pcm = priv_data;



	BUG_ON(!pcm);



	pr_debug("%s: event %x\n", __func__, event);



	switch (event) {

	case MSM_PCM_RT_EVT_DEVSWITCH:

		q6asm_cmd(pcm->audio_client, CMD_PAUSE);

		q6asm_cmd(pcm->audio_client, CMD_FLUSH);

		q6asm_run(pcm->audio_client, 0, 0, 0);

	default:

		break;

	}

}



static void msm_pcm_loopback_event_handler(uint32_t opcode, uint32_t token,

					   uint32_t *payload, void *priv)

{
@@ -112,6 +131,7 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) 
	struct msm_pcm_loopback *pcm;

	int ret = 0;

	uint16_t bits_per_sample = 16;

	struct msm_pcm_routing_evt event;



	pcm = dev_get_drvdata(rtd->platform->dev);

	mutex_lock(&pcm->lock);
@@ -154,12 +174,15 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) 
			mutex_unlock(&pcm->lock);

			return -ENOMEM;

		}

		event.event_func = msm_pcm_route_event_handler;

		event.priv_data = (void *) pcm;

		msm_pcm_routing_reg_phy_stream(soc_pcm_tx->dai_link->be_id,

			pcm->audio_client->perf_mode,

			pcm->session_id, pcm->capture_substream->stream);

		msm_pcm_routing_reg_phy_stream(soc_pcm_rx->dai_link->be_id,

		msm_pcm_routing_reg_phy_stream_v2(soc_pcm_rx->dai_link->be_id,

			pcm->audio_client->perf_mode,

			pcm->session_id, pcm->playback_substream->stream);

			pcm->session_id, pcm->playback_substream->stream,

			event);

		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

			pcm->playback_substream = substream;

			ret = pcm_loopback_set_volume(pcm, pcm->volume);

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

+6 −0
Original line number Diff line number Diff line
@@ -682,6 +682,12 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set) 
				topology, fe_dai_perf_mode[val][session_type],

				bits_per_sample);



			if (session_type == SESSION_TYPE_RX &&

			    fdai->event_info.event_func)

				fdai->event_info.event_func(

					MSM_PCM_RT_EVT_DEVSWITCH,

					fdai->event_info.priv_data);



			msm_pcm_routing_build_matrix(val,

				fdai->strm_id, path_type,

				fe_dai_perf_mode[val][session_type]);

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

+1 −0
Original line number Diff line number Diff line
@@ -139,6 +139,7 @@ enum { 


enum msm_pcm_routing_event {

	MSM_PCM_RT_EVT_BUF_RECFG,

	MSM_PCM_RT_EVT_DEVSWITCH,

	MSM_PCM_RT_EVT_MAX,

};