Unverified Commit 81e7d4ce authored by Mark Brown's avatar Mark Brown
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Merge series "ASoC: qdsp6: add gapless compressed audio support" from Srinivas... 

Merge series "ASoC: qdsp6: add gapless compressed audio support" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:

This patchset adds gapless compressed audio support on q6asm.
Gapless on q6asm is implemented using 2 streams in a single asm session.

First few patches are enhacements done to q6asm interface to allow
stream id per each command, gapless flags and silence meta data.
Along with this there are few trivial changes which I thought are necessary!
Last patch implements copy callback to allow finer control over buffer offsets,
specially in partial drain cases.

This patchset is tested on RB3 aka DB845c platform.

Thanks,
srini

Srinivas Kandagatla (11):
  ASoC: q6asm: add command opcode to timeout error report
  ASoC: q6asm: rename misleading session id variable
  ASoC: q6asm: make commands specific to streams
  ASoC: q6asm: use flags directly from asm-dai
  ASoC: q6asm: add length to write command token
  ASoC: q6asm: add support to remove intial and trailing silence
  ASoC: q6asm: add support to gapless flag in asm open
  ASoC: q6asm-dai: add next track metadata support
  ASoC: qdsp6: use dev_err instead of pr_err
  ASoC: qdsp6-dai: add gapless support
  ASoC: q6asm-dai: add support to copy callback

 sound/soc/qcom/qdsp6/q6asm-dai.c | 397 +++++++++++++++++++++++--------
 sound/soc/qcom/qdsp6/q6asm.c     | 173 +++++++++-----
 sound/soc/qcom/qdsp6/q6asm.h     |  48 ++--
 3 files changed, 458 insertions(+), 160 deletions(-)

--
2.21.0
parents ee3d1339 0579ece8

sound/soc/qcom/qdsp6/q6asm-dai.c

+14 −13
Original line number Diff line number Diff line
@@ -218,6 +218,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 
	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;

	struct q6asm_dai_rtd *prtd = runtime->private_data;

	struct q6asm_dai_data *pdata;

	struct device *dev = component->dev;

	int ret, i;



	pdata = snd_soc_component_get_drvdata(component);
@@ -225,7 +226,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 
		return -EINVAL;



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

		pr_err("%s: private data null or audio client freed\n",

		dev_err(dev, "%s: private data null or audio client freed\n",

			__func__);

		return -EINVAL;

	}
@@ -248,7 +249,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 
				       prtd->periods);



	if (ret < 0) {

		pr_err("Audio Start: Buffer Allocation failed rc = %d\n",

		dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",

							ret);

		return -ENOMEM;

	}
@@ -262,7 +263,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 
	}



	if (ret < 0) {

		pr_err("%s: q6asm_open_write failed\n", __func__);

		dev_err(dev, "%s: q6asm_open_write failed\n", __func__);

		q6asm_audio_client_free(prtd->audio_client);

		prtd->audio_client = NULL;

		return -ENOMEM;
@@ -272,7 +273,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 
	ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,

			      prtd->session_id, substream->stream);

	if (ret) {

		pr_err("%s: stream reg failed ret:%d\n", __func__, ret);

		dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);

		return ret;

	}
@@ -292,7 +293,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, 


	}

	if (ret < 0)

		pr_info("%s: CMD Format block failed\n", __func__);

		dev_info(dev, "%s: CMD Format block failed\n", __func__);



	prtd->state = Q6ASM_STREAM_RUNNING;
@@ -344,7 +345,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, 


	pdata = snd_soc_component_get_drvdata(component);

	if (!pdata) {

		pr_err("Drv data not found ..\n");

		dev_err(dev, "Drv data not found ..\n");

		return -EINVAL;

	}
@@ -357,7 +358,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, 
				(q6asm_cb)event_handler, prtd, stream_id,

				LEGACY_PCM_MODE);

	if (IS_ERR(prtd->audio_client)) {

		pr_info("%s: Could not allocate memory\n", __func__);

		dev_info(dev, "%s: Could not allocate memory\n", __func__);

		ret = PTR_ERR(prtd->audio_client);

		kfree(prtd);

		return ret;
@@ -372,12 +373,12 @@ static int q6asm_dai_open(struct snd_soc_component *component, 
				SNDRV_PCM_HW_PARAM_RATE,

				&constraints_sample_rates);

	if (ret < 0)

		pr_info("snd_pcm_hw_constraint_list failed\n");

		dev_info(dev, "snd_pcm_hw_constraint_list failed\n");

	/* Ensure that buffer size is a multiple of period size */

	ret = snd_pcm_hw_constraint_integer(runtime,

					    SNDRV_PCM_HW_PARAM_PERIODS);

	if (ret < 0)

		pr_info("snd_pcm_hw_constraint_integer failed\n");

		dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");



	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

		ret = snd_pcm_hw_constraint_minmax(runtime,
@@ -385,7 +386,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, 
			PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,

			PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);

		if (ret < 0) {

			pr_err("constraint for buffer bytes min max ret = %d\n",

			dev_err(dev, "constraint for buffer bytes min max ret = %d\n",

				ret);

		}

	}
@@ -393,13 +394,13 @@ static int q6asm_dai_open(struct snd_soc_component *component, 
	ret = snd_pcm_hw_constraint_step(runtime, 0,

		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);

	if (ret < 0) {

		pr_err("constraint for period bytes step ret = %d\n",

		dev_err(dev, "constraint for period bytes step ret = %d\n",

								ret);

	}

	ret = snd_pcm_hw_constraint_step(runtime, 0,

		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);

	if (ret < 0) {

		pr_err("constraint for buffer bytes step ret = %d\n",

		dev_err(dev, "constraint for buffer bytes step ret = %d\n",

								ret);

	}

sound/soc/qcom/qdsp6/q6asm.c

+2 −2
Original line number Diff line number Diff line
@@ -311,7 +311,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac, 
					5 * HZ);



	if (!rc) {

		dev_err(a->dev, "CMD timeout\n");

		dev_err(a->dev, "CMD %x timeout\n", hdr->opcode);

		rc = -ETIMEDOUT;

	} else if (ac->result.status > 0) {

		dev_err(a->dev, "DSP returned error[%x]\n",
@@ -891,7 +891,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) 
	rc = wait_event_timeout(ac->cmd_wait,

				(ac->result.opcode == hdr->opcode), 5 * HZ);

	if (!rc) {

		dev_err(ac->dev, "CMD timeout\n");

		dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode);

		rc =  -ETIMEDOUT;

		goto err;

	}

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