[PATCH 00/14] Add support for compress offload and gapless playback.

[PATCH 00/14] Add support for compress offload and gapless playback.

[Mohammad Rafi Shaik]

Add support for compress offload and gapless playback in audioreach
platform drivers.

Mohammad Rafi Shaik (14):
  ALSA: compress: Update compress set params for gapless playback
  ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup
    callback
  ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback
  ASoC: q6dsp: audioreach: Add support for compress offload commands
  ASoC: q6dsp: audioreach: Add support to set compress params
  ASoC: q6dsp: audioreach: Add support for sending real module ID to
    ADSP
  ASoC: q6dsp: q6apm-dai: Add async compress write support
  ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks
  ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks
  ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks
  ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI
    callbacks
  ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks
  ASoC: qdsp6: audioreach: Add MP3, AAC and FLAC compress format support
  ASoC: q6dsp: audioreach: Add gapless feature support

 sound/core/compress_offload.c     |  12 +-
 sound/soc/qcom/qdsp6/audioreach.c | 299 +++++++++++++++++--
 sound/soc/qcom/qdsp6/audioreach.h |  56 ++++
 sound/soc/qcom/qdsp6/q6apm-dai.c  | 464 ++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6apm.c      | 117 ++++++++
 sound/soc/qcom/qdsp6/q6apm.h      |   8 +
 sound/soc/qcom/sc7280.c           |  21 +-
 7 files changed, 950 insertions(+), 27 deletions(-)

-- 
2.25.1

[PATCH 01/14] ALSA: compress: Update compress set params for gapless playback

[Mohammad Rafi Shaik]

Update compress set params for supporting next track settings
during gapless playback.
Update the runtime state to setup state only if it's in open state.
Allow parameter change only when stream has been opened and running
state.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/core/compress_offload.c | 12 ++++++++----
 1 file changed, 8 insertions(+), 4 deletions(-)

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 243acad89fd3..9b951d76c120 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -589,7 +589,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 	struct snd_compr_params *params;
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
+	if (stream->runtime->state == SNDRV_PCM_STATE_OPEN ||
+			stream->runtime->state == SNDRV_PCM_STATE_RUNNING) {
 		/*
 		 * we should allow parameter change only when stream has been
 		 * opened not in other cases
@@ -612,10 +613,13 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		if (retval)
 			goto out;
 
-		stream->metadata_set = false;
-		stream->next_track = false;
+		if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
+			stream->metadata_set = false;
+			stream->next_track = false;
+
+			stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+		}
 
-		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
 	} else {
 		return -EPERM;
 	}
-- 
2.25.1

[PATCH 02/14] ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback

[Mohammad Rafi Shaik]

Add support to set backend params such as sampling rate and
number of channels using backend params fixup callback.
Also remove hardware params constraints setting.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/sc7280.c | 21 ++++++++++++++++++---
 1 file changed, 18 insertions(+), 3 deletions(-)

diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c
index da7469a6a267..aaa95fe63d83 100644
--- a/sound/soc/qcom/sc7280.c
+++ b/sound/soc/qcom/sc7280.c
@@ -14,6 +14,7 @@
 #include <sound/soc.h>
 #include <sound/rt5682s.h>
 #include <linux/soundwire/sdw.h>
+#include <sound/pcm_params.h>
 
 #include "../codecs/rt5682.h"
 #include "../codecs/rt5682s.h"
@@ -24,6 +25,7 @@
 #define DEFAULT_MCLK_RATE              19200000
 #define RT5682_PLL_FREQ (48000 * 512)
 #define MI2S_BCLK_RATE		1536000
+#define DEFAULT_SAMPLE_RATE_48K	48000
 
 struct sc7280_snd_data {
 	struct snd_soc_card card;
@@ -188,7 +190,6 @@ static int sc7280_init(struct snd_soc_pcm_runtime *rtd)
 static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai;
 	const struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
@@ -196,8 +197,6 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
 	struct sdw_stream_runtime *sruntime;
 	int i;
 
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, 48000, 48000);
 
 	switch (cpu_dai->id) {
 	case LPASS_CDC_DMA_TX3:
@@ -358,6 +357,20 @@ static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = {
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 };
 
+static int sc7280_snd_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+					 struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K;
+	channels->min = channels->max = 2;
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
 static int sc7280_snd_platform_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card;
@@ -387,6 +400,8 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev)
 	for_each_card_prelinks(card, i, link) {
 		link->init = sc7280_init;
 		link->ops = &sc7280_ops;
+		if (link->no_pcm == 1)
+			link->be_hw_params_fixup = sc7280_snd_be_hw_params_fixup;
 	}
 
 	return devm_snd_soc_register_card(dev, card);
-- 
2.25.1

[PATCH 03/14] ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback

[Mohammad Rafi Shaik]

Add placeholder decoder graph module for compressed playback feature.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c |  2 ++
 sound/soc/qcom/qdsp6/audioreach.h | 39 +++++++++++++++++++++++++++++++
 2 files changed, 41 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 1e0c918eb576..6d3d2a04ffe8 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -1120,6 +1120,8 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
 	case MODULE_ID_PCM_DEC:
 	case MODULE_ID_PCM_ENC:
 	case MODULE_ID_PCM_CNV:
+	case MODULE_ID_PLACEHOLDER_DECODER:
+	case MODULE_ID_PLACEHOLDER_ENCODER:
 		rc = audioreach_pcm_set_media_format(graph, module, cfg);
 		break;
 	case MODULE_ID_I2S_SOURCE:
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 1d1d47d47d40..b78fd9bc8eb3 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -15,6 +15,8 @@ struct q6apm_graph;
 #define MODULE_ID_PCM_CNV		0x07001003
 #define MODULE_ID_PCM_ENC		0x07001004
 #define MODULE_ID_PCM_DEC		0x07001005
+#define MODULE_ID_PLACEHOLDER_ENCODER	0x07001008
+#define MODULE_ID_PLACEHOLDER_DECODER	0x07001009
 #define MODULE_ID_SAL			0x07001010
 #define MODULE_ID_MFC			0x07001015
 #define MODULE_ID_CODEC_DMA_SINK	0x07001023
@@ -22,6 +24,10 @@ struct q6apm_graph;
 #define MODULE_ID_I2S_SINK		0x0700100A
 #define MODULE_ID_I2S_SOURCE		0x0700100B
 #define MODULE_ID_DATA_LOGGING		0x0700101A
+#define MODULE_ID_AAC_DEC		0x0700101F
+#define MODULE_ID_FLAC_DEC		0x0700102F
+#define MODULE_ID_MP3_DECODE		0x0700103B
+#define MODULE_ID_GAPLESS		0x0700104D
 
 #define APM_CMD_GET_SPF_STATE		0x01001021
 #define APM_CMD_RSP_GET_SPF_STATE	0x02001007
@@ -142,12 +148,15 @@ struct param_id_enc_bitrate_param {
 } __packed;
 
 #define DATA_FORMAT_FIXED_POINT		1
+#define DATA_FORMAT_GENERIC_COMPRESSED		5
+#define DATA_FORMAT_RAW_COMPRESSED		6
 #define PCM_LSB_ALIGNED			1
 #define PCM_MSB_ALIGNED			2
 #define PCM_LITTLE_ENDIAN		1
 #define PCM_BIT_ENDIAN			2
 
 #define MEDIA_FMT_ID_PCM	0x09001000
+#define MEDIA_FMT_ID_MP3	0x09001009
 #define PCM_CHANNEL_L		1
 #define PCM_CHANNEL_R		2
 #define SAMPLE_RATE_48K		48000
@@ -225,6 +234,28 @@ struct apm_media_format {
 	uint32_t payload_size;
 } __packed;
 
+#define MEDIA_FMT_ID_FLAC	0x09001004
+
+struct payload_media_fmt_flac_t {
+	uint16_t num_channels;
+	uint16_t sample_size;
+	uint16_t min_blk_size;
+	uint16_t max_blk_size;
+	uint32_t sample_rate;
+	uint32_t min_frame_size;
+	uint32_t max_frame_size;
+} __packed;
+
+#define MEDIA_FMT_ID_AAC	0x09001001
+
+struct payload_media_fmt_aac_t {
+	uint16_t aac_fmt_flag;
+	uint16_t audio_obj_type;
+	uint16_t num_channels;
+	uint16_t total_size_of_PCE_bits;
+	uint32_t sample_rate;
+} __packed;
+
 #define DATA_CMD_WR_SH_MEM_EP_EOS			0x04001002
 #define WR_SH_MEM_EP_EOS_POLICY_LAST	1
 #define WR_SH_MEM_EP_EOS_POLICY_EACH	2
@@ -598,6 +629,13 @@ struct param_id_vol_ctrl_master_gain {
 } __packed;
 
 
+
+#define PARAM_ID_REAL_MODULE_ID	0x0800100B
+
+struct param_id_placeholder_real_module_id {
+	uint32_t real_module_id;
+} __packed;
+
 /* Graph */
 struct audioreach_connection {
 	/* Connections */
@@ -704,6 +742,7 @@ struct audioreach_module_config {
 	u16	active_channels_mask;
 	u32	sd_line_mask;
 	int	fmt;
+	struct snd_codec codec;
 	u8 channel_map[AR_PCM_MAX_NUM_CHANNEL];
 };
 
-- 
2.25.1

[PATCH 04/14] ASoC: q6dsp: audioreach: Add support for compress offload commands

[Mohammad Rafi Shaik]

Add functions to send commands to the ADSP for supporting compressed
offload playback.
This includes functions to enable module ID, to remove trailing and
initial silence.
Also add functionality to send 32 bit parameter to ADSP.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c | 56 +++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/audioreach.h |  7 ++++
 sound/soc/qcom/qdsp6/q6apm.c      | 39 +++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6apm.h      |  3 ++
 4 files changed, 105 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 6d3d2a04ffe8..a11bab69a676 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -1258,3 +1258,59 @@ int audioreach_shared_memory_send_eos(struct q6apm_graph *graph)
 	return rc;
 }
 EXPORT_SYMBOL_GPL(audioreach_shared_memory_send_eos);
+
+int audioreach_send_u32_param(struct q6apm *apm, struct audioreach_module *module,
+			      uint32_t param_id, uint32_t param_val)
+{
+	struct apm_module_param_data *param_data;
+	struct gpr_pkt *pkt;
+	uint32_t *param;
+	int rc, payload_size;
+	void *p;
+
+	payload_size = sizeof(uint32_t) + APM_MODULE_PARAM_DATA_SIZE;
+	p = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
+	if (IS_ERR(p))
+		return -ENOMEM;
+
+	pkt = p;
+	p = p + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
+
+	param_data = p;
+	param_data->module_instance_id = module->instance_id;
+	param_data->error_code = 0;
+	param_data->param_id = param_id;
+	param_data->param_size = payload_size - APM_MODULE_PARAM_DATA_SIZE;
+
+	p = p + APM_MODULE_PARAM_DATA_SIZE;
+	param = p;
+	*param = param_val;
+	rc = q6apm_send_cmd_sync(apm, pkt, 0);
+
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(audioreach_send_u32_param);
+
+int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_module *module,
+				       uint32_t trailing_samples)
+{
+	return audioreach_send_u32_param(apm, module, PARAM_ID_REMOVE_TRAILING_SILENCE,
+					 trailing_samples);
+}
+EXPORT_SYMBOL_GPL(audioreach_remove_trailing_silence);
+
+int audioreach_remove_initial_silence(struct q6apm *apm, struct audioreach_module *module,
+				      uint32_t initial_samples)
+{
+	return audioreach_send_u32_param(apm, module, PARAM_ID_REMOVE_INITIAL_SILENCE,
+					 initial_samples);
+}
+EXPORT_SYMBOL_GPL(audioreach_remove_initial_silence);
+
+int audioreach_enable_module(struct q6apm *apm, struct audioreach_module *module, bool en)
+{
+	return audioreach_send_u32_param(apm, module, PARAM_ID_MODULE_ENABLE, en);
+}
+EXPORT_SYMBOL_GPL(audioreach_enable_module);
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index b78fd9bc8eb3..76dea97773cc 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -629,6 +629,8 @@ struct param_id_vol_ctrl_master_gain {
 } __packed;
 
 
+#define PARAM_ID_REMOVE_INITIAL_SILENCE		0x0800114B
+#define PARAM_ID_REMOVE_TRAILING_SILENCE	0x0800115D
 
 #define PARAM_ID_REAL_MODULE_ID	0x0800100B
 
@@ -779,4 +781,9 @@ int audioreach_set_media_format(struct q6apm_graph *graph,
 int audioreach_shared_memory_send_eos(struct q6apm_graph *graph);
 int audioreach_gain_set_vol_ctrl(struct q6apm *apm,
 				 struct audioreach_module *module, int vol);
+int audioreach_enable_module(struct q6apm *apm, struct audioreach_module *module, bool en);
+int audioreach_remove_initial_silence(struct q6apm *apm, struct audioreach_module *module,
+				uint32_t initial_samples);
+int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_module *module,
+				uint32_t trailing_samples);
 #endif /* __AUDIOREACH_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 8a7dfd27d3c5..78c1a7c13348 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -297,6 +297,45 @@ int q6apm_unmap_memory_regions(struct q6apm_graph *graph, unsigned int dir)
 }
 EXPORT_SYMBOL_GPL(q6apm_unmap_memory_regions);
 
+int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
+{
+	struct q6apm *apm = dev_get_drvdata(dev->parent);
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_remove_initial_silence(apm, module, samples);
+}
+EXPORT_SYMBOL_GPL(q6apm_remove_initial_silence);
+
+int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
+{
+	struct q6apm *apm = dev_get_drvdata(dev->parent);
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_remove_trailing_silence(apm, module, samples);
+}
+EXPORT_SYMBOL_GPL(q6apm_remove_trailing_silence);
+
+int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en)
+{
+	struct q6apm *apm = dev_get_drvdata(dev->parent);
+	struct audioreach_module *module;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	return audioreach_enable_module(apm, module, en);
+}
+EXPORT_SYMBOL_GPL(q6apm_enable_compress_module);
+
 int q6apm_graph_media_format_pcm(struct q6apm_graph *graph, struct audioreach_module_config *cfg)
 {
 	struct audioreach_graph_info *info = graph->info;
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index 7005be9b63e3..08b64f78c750 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -147,4 +147,7 @@ int q6apm_graph_get_rx_shmem_module_iid(struct q6apm_graph *graph);
 
 bool q6apm_is_adsp_ready(void);
 
+int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en);
+int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
+int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 #endif /* __APM_GRAPH_ */
-- 
2.25.1

[PATCH 05/14] ASoC: q6dsp: audioreach: Add support to set compress params

[Mohammad Rafi Shaik]

Add function for setting compress params to set the next track
parameters during gapless playback.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c | 51 +++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/audioreach.h |  1 +
 sound/soc/qcom/qdsp6/q6apm-dai.c  |  1 +
 3 files changed, 53 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index a11bab69a676..a87df09d187f 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -1314,3 +1314,54 @@ int audioreach_enable_module(struct q6apm *apm, struct audioreach_module *module
 	return audioreach_send_u32_param(apm, module, PARAM_ID_MODULE_ENABLE, en);
 }
 EXPORT_SYMBOL_GPL(audioreach_enable_module);
+
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
+{
+	struct media_format *header;
+	struct gpr_pkt *pkt;
+	struct payload_media_fmt_pcm *cfg;
+	uint32_t num_channels = mcfg->num_channels;
+	int iid, payload_size, rc;
+	void *p;
+
+	payload_size = sizeof(struct apm_sh_module_media_fmt_cmd);
+
+	iid = q6apm_graph_get_rx_shmem_module_iid(graph);
+	pkt = audioreach_alloc_cmd_pkt(payload_size, DATA_CMD_WR_SH_MEM_EP_MEDIA_FORMAT,
+			0, graph->port->id, iid);
+
+	if (IS_ERR(pkt))
+		return -ENOMEM;
+
+	p = (void *)pkt + GPR_HDR_SIZE;
+	header = p;
+
+	if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
+		header->data_format = DATA_FORMAT_FIXED_POINT;
+		header->fmt_id =  MEDIA_FMT_ID_PCM;
+		header->payload_size = sizeof(*cfg);
+
+		p = p + sizeof(*header);
+		cfg = p;
+		cfg->sample_rate = mcfg->sample_rate;
+		cfg->bit_width = mcfg->bit_width;
+		cfg->alignment = PCM_LSB_ALIGNED;
+		cfg->bits_per_sample = mcfg->bit_width;
+		cfg->q_factor = mcfg->bit_width - 1;
+		cfg->endianness = PCM_LITTLE_ENDIAN;
+		cfg->num_channels = mcfg->num_channels;
+
+		if (mcfg->num_channels == 1)
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+		else if (num_channels == 2) {
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+			cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+		}
+	}
+
+	rc = gpr_send_port_pkt(graph->port, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(audioreach_compr_set_param);
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 76dea97773cc..4c4bdff45cf1 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -786,4 +786,5 @@ int audioreach_remove_initial_silence(struct q6apm *apm, struct audioreach_modul
 				uint32_t initial_samples);
 int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_module *module,
 				uint32_t trailing_samples);
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
 #endif /* __AUDIOREACH_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index ee59ef36b85a..8f5d744b3365 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -146,6 +146,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	cfg.sample_rate = runtime->rate;
 	cfg.num_channels = runtime->channels;
 	cfg.bit_width = prtd->bits_per_sample;
+	cfg.fmt = SND_AUDIOCODEC_PCM;
 
 	if (prtd->state) {
 		/* clear the previous setup if any  */
-- 
2.25.1

[PATCH 06/14] ASoC: q6dsp: audioreach: Add support for sending real module ID to ADSP

[Mohammad Rafi Shaik]

Add support for sending the placeholder real module ID to ADSP
for enabling compressed playback. 

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c |  6 ++++++
 sound/soc/qcom/qdsp6/audioreach.h |  1 +
 sound/soc/qcom/qdsp6/q6apm.c      | 29 +++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6apm.h      |  1 +
 4 files changed, 37 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index a87df09d187f..e84ccbacc0f7 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -1315,6 +1315,12 @@ int audioreach_enable_module(struct q6apm *apm, struct audioreach_module *module
 }
 EXPORT_SYMBOL_GPL(audioreach_enable_module);
 
+int audioreach_set_real_module_id(struct q6apm *apm, struct audioreach_module *module, uint32_t id)
+{
+	return audioreach_send_u32_param(apm, module, PARAM_ID_REAL_MODULE_ID, id);
+}
+EXPORT_SYMBOL_GPL(audioreach_set_real_module_id);
+
 int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
 {
 	struct media_format *header;
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 4c4bdff45cf1..0faaf75115fd 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -787,4 +787,5 @@ int audioreach_remove_initial_silence(struct q6apm *apm, struct audioreach_modul
 int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_module *module,
 				uint32_t trailing_samples);
 int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
+int audioreach_set_real_module_id(struct q6apm *apm, struct audioreach_module *module, uint32_t id);
 #endif /* __AUDIOREACH_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 78c1a7c13348..811d86bdc092 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -336,6 +336,35 @@ int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph,
 }
 EXPORT_SYMBOL_GPL(q6apm_enable_compress_module);
 
+int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph,
+			     uint32_t codec_id)
+{
+	struct q6apm *apm = dev_get_drvdata(dev->parent);
+	struct audioreach_module *module;
+	uint32_t module_id;
+
+	module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
+	if (!module)
+		return -ENODEV;
+
+	switch (codec_id) {
+	case SND_AUDIOCODEC_MP3:
+		module_id = MODULE_ID_MP3_DECODE;
+		break;
+	case SND_AUDIOCODEC_AAC:
+		module_id = MODULE_ID_AAC_DEC;
+		break;
+	case SND_AUDIOCODEC_FLAC:
+		module_id = MODULE_ID_FLAC_DEC;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return audioreach_set_real_module_id(apm, module, module_id);
+}
+EXPORT_SYMBOL_GPL(q6apm_set_real_module_id);
+
 int q6apm_graph_media_format_pcm(struct q6apm_graph *graph, struct audioreach_module_config *cfg)
 {
 	struct audioreach_graph_info *info = graph->info;
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index 08b64f78c750..87d67faf5f1a 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -150,4 +150,5 @@ bool q6apm_is_adsp_ready(void);
 int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en);
 int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
+int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id);
 #endif /* __APM_GRAPH_ */
-- 
2.25.1

[PATCH 07/14] ASoC: q6dsp: q6apm-dai: Add async compress write support

[Mohammad Rafi Shaik]

Add async compress write API to send the compressed audio data packet
to ADSP. 

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c | 38 +++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/audioreach.h |  2 ++
 sound/soc/qcom/qdsp6/q6apm-dai.c  |  7 +++++
 sound/soc/qcom/qdsp6/q6apm.c      | 46 +++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6apm.h      |  3 ++
 5 files changed, 96 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index e84ccbacc0f7..7c45c36e9156 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -240,6 +240,44 @@ void *audioreach_alloc_pkt(int payload_size, uint32_t opcode, uint32_t token,
 }
 EXPORT_SYMBOL_GPL(audioreach_alloc_pkt);
 
+static void __audioreach_update_pkt(struct gpr_pkt *pkt, int payload_size, uint32_t opcode,
+				    uint32_t token, uint32_t src_port, uint32_t dest_port,
+				    bool has_cmd_hdr)
+{
+	int pkt_size = GPR_HDR_SIZE + payload_size;
+	void *p;
+
+	if (has_cmd_hdr)
+		pkt_size += APM_CMD_HDR_SIZE;
+
+	p = pkt;
+	pkt->hdr.version = GPR_PKT_VER;
+	pkt->hdr.hdr_size = GPR_PKT_HEADER_WORD_SIZE;
+	pkt->hdr.pkt_size = pkt_size;
+	pkt->hdr.dest_port = dest_port;
+	pkt->hdr.src_port = src_port;
+
+	pkt->hdr.dest_domain = GPR_DOMAIN_ID_ADSP;
+	pkt->hdr.src_domain = GPR_DOMAIN_ID_APPS;
+	pkt->hdr.token = token;
+	pkt->hdr.opcode = opcode;
+
+	if (has_cmd_hdr) {
+		struct apm_cmd_header *cmd_header;
+
+		p = p + GPR_HDR_SIZE;
+		cmd_header = p;
+		cmd_header->payload_size = payload_size;
+	}
+}
+
+void audioreach_update_pkt(struct gpr_pkt *pkt, int payload_size, uint32_t opcode, uint32_t token,
+			   uint32_t src_port, uint32_t dest_port)
+{
+	__audioreach_update_pkt(pkt, payload_size, opcode, token, src_port, dest_port, false);
+}
+EXPORT_SYMBOL_GPL(audioreach_update_pkt);
+
 void *audioreach_alloc_apm_pkt(int pkt_size, uint32_t opcode, uint32_t token, uint32_t src_port)
 {
 	return __audioreach_alloc_pkt(pkt_size, opcode, token, src_port, APM_MODULE_INSTANCE_ID,
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 0faaf75115fd..044994ca4811 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -788,4 +788,6 @@ int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_modu
 				uint32_t trailing_samples);
 int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
 int audioreach_set_real_module_id(struct q6apm *apm, struct audioreach_module *module, uint32_t id);
+void audioreach_update_pkt(struct gpr_pkt *pkt, int payload_size, uint32_t opcode,
+			   uint32_t token, uint32_t src_port, uint32_t dest_port);
 #endif /* __AUDIOREACH_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 8f5d744b3365..e621e31294a1 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -38,8 +38,10 @@ enum stream_state {
 struct q6apm_dai_rtd {
 	struct snd_pcm_substream *substream;
 	struct snd_compr_stream *cstream;
+	struct snd_codec codec;
 	struct snd_compr_params codec_param;
 	struct snd_dma_buffer dma_buffer;
+	spinlock_t lock;
 	phys_addr_t phys;
 	unsigned int pcm_size;
 	unsigned int pcm_count;
@@ -51,8 +53,13 @@ struct q6apm_dai_rtd {
 	uint16_t bits_per_sample;
 	uint16_t source; /* Encoding source bit mask */
 	uint16_t session_id;
+	bool next_track;
 	enum stream_state state;
 	struct q6apm_graph *graph;
+	uint32_t initial_samples_drop;
+	uint32_t trailing_samples_drop;
+	uint32_t next_track_stream_id;
+	bool notify_on_drain;
 };
 
 struct q6apm_dai_data {
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 811d86bdc092..1a6c7108bae0 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -25,6 +25,8 @@ struct apm_graph_mgmt_cmd {
 	uint32_t sub_graph_id_list[];
 } __packed;
 
+struct gpr_pkt *pkt;
+
 #define APM_GRAPH_MGMT_PSIZE(p, n) ALIGN(struct_size(p, sub_graph_id_list, n), 8)
 
 struct q6apm *g_apm;
@@ -457,6 +459,45 @@ int q6apm_write_async(struct q6apm_graph *graph, uint32_t len, uint32_t msw_ts,
 }
 EXPORT_SYMBOL_GPL(q6apm_write_async);
 
+int q6apm_write_async_compr(struct q6apm_graph *graph, uint32_t len, uint32_t msw_ts,
+		      uint32_t lsw_ts, uint32_t wflags)
+{
+	struct apm_data_cmd_wr_sh_mem_ep_data_buffer_v2 *write_buffer;
+	struct audio_buffer *ab;
+
+	int rc, iid;
+
+	iid = q6apm_graph_get_rx_shmem_module_iid(graph);
+
+	audioreach_update_pkt(pkt, sizeof(*write_buffer), DATA_CMD_WR_SH_MEM_EP_DATA_BUFFER_V2,
+			      graph->rx_data.dsp_buf | (len << APM_WRITE_TOKEN_LEN_SHIFT),
+			      graph->port->id, iid);
+
+	write_buffer = (void *)pkt + GPR_HDR_SIZE;
+
+	ab = &graph->rx_data.buf[graph->rx_data.dsp_buf];
+
+	write_buffer->buf_addr_lsw = lower_32_bits(ab->phys);
+	write_buffer->buf_addr_msw = upper_32_bits(ab->phys);
+	write_buffer->buf_size = len;
+	write_buffer->timestamp_lsw = lsw_ts;
+	write_buffer->timestamp_msw = msw_ts;
+	write_buffer->mem_map_handle = graph->rx_data.mem_map_handle;
+	write_buffer->flags = wflags;
+
+	graph->rx_data.dsp_buf++;
+
+	if (graph->rx_data.dsp_buf >= graph->rx_data.num_periods)
+		graph->rx_data.dsp_buf = 0;
+
+	rc = gpr_send_port_pkt(graph->port, pkt);
+
+	memset(pkt, 0, sizeof(write_buffer) + GPR_HDR_SIZE);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6apm_write_async_compr);
+
 int q6apm_read(struct q6apm_graph *graph)
 {
 	struct data_cmd_rd_sh_mem_ep_data_buffer_v2 *read_buffer;
@@ -724,6 +765,11 @@ static int apm_probe(gpr_device_t *gdev)
 
 	dev_set_drvdata(dev, apm);
 
+	pkt = devm_kzalloc(dev, sizeof(struct apm_data_cmd_wr_sh_mem_ep_data_buffer_v2) +
+			   GPR_HDR_SIZE, GFP_KERNEL);
+	if (!pkt)
+		return -ENOMEM;
+
 	mutex_init(&apm->lock);
 	apm->dev = dev;
 	apm->gdev = gdev;
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index 87d67faf5f1a..630c2bca0f06 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -45,6 +45,7 @@
 #define APM_WRITE_TOKEN_LEN_SHIFT              16
 
 #define APM_MAX_SESSIONS			8
+#define APM_LAST_BUFFER_FLAG			BIT(30)
 
 struct q6apm {
 	struct device *dev;
@@ -128,6 +129,8 @@ int q6apm_send_eos_nowait(struct q6apm_graph *graph);
 int q6apm_read(struct q6apm_graph *graph);
 int q6apm_write_async(struct q6apm_graph *graph, uint32_t len, uint32_t msw_ts,
 		      uint32_t lsw_ts, uint32_t wflags);
+int q6apm_write_async_compr(struct q6apm_graph *graph, uint32_t len, uint32_t msw_ts,
+			    uint32_t lsw_ts, uint32_t wflags);
 
 /* Memory Map related */
 int q6apm_map_memory_regions(struct q6apm_graph *graph,
-- 
2.25.1

[PATCH 08/14] ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks

[Mohammad Rafi Shaik]

Add q6apm open and free compress DAI callbacks to support compress
offload playback.
Include compress event handler callback also.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 135 +++++++++++++++++++++++++++++++
 1 file changed, 135 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index e621e31294a1..fd134c268189 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -27,6 +27,8 @@
 #define CAPTURE_MIN_PERIOD_SIZE		320
 #define BUFFER_BYTES_MAX (PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE)
 #define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
+#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
+#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
 #define SID_MASK_DEFAULT	0xF
 
 enum stream_state {
@@ -130,6 +132,69 @@ static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, vo
 	}
 }
 
+void event_handler_compr(uint32_t opcode, uint32_t token,
+				uint32_t *payload, void *priv)
+{
+	struct q6apm_dai_rtd *prtd = priv;
+	struct snd_compr_stream *substream = prtd->cstream;
+	unsigned long flags;
+	uint32_t wflags = 0;
+	uint64_t avail;
+	uint32_t bytes_written, bytes_to_write;
+	bool is_last_buffer = false;
+
+	switch (opcode) {
+	case APM_CLIENT_EVENT_CMD_EOS_DONE:
+		spin_lock_irqsave(&prtd->lock, flags);
+		if (prtd->notify_on_drain) {
+			snd_compr_drain_notify(prtd->cstream);
+			prtd->notify_on_drain = false;
+		} else {
+			prtd->state = Q6APM_STREAM_STOPPED;
+		}
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+	case APM_CLIENT_EVENT_DATA_WRITE_DONE:
+		spin_lock_irqsave(&prtd->lock, flags);
+		bytes_written = token >> APM_WRITE_TOKEN_LEN_SHIFT;
+		prtd->copied_total += bytes_written;
+		snd_compr_fragment_elapsed(substream);
+
+		if (prtd->state != Q6APM_STREAM_RUNNING) {
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			break;
+		}
+
+		avail = prtd->bytes_received - prtd->bytes_sent;
+
+		if (avail > prtd->pcm_count) {
+			bytes_to_write = prtd->pcm_count;
+		} else {
+			if (substream->partial_drain || prtd->notify_on_drain)
+				is_last_buffer = true;
+			bytes_to_write = avail;
+		}
+
+		if (bytes_to_write) {
+			if (substream->partial_drain && is_last_buffer)
+				wflags |= APM_LAST_BUFFER_FLAG;
+
+			q6apm_write_async_compr(prtd->graph,
+						bytes_to_write, 0, 0, wflags);
+
+			prtd->bytes_sent += bytes_to_write;
+
+			if (prtd->notify_on_drain && is_last_buffer)
+				audioreach_shared_memory_send_eos(prtd->graph);
+		}
+
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+	default:
+		break;
+	}
+}
+
 static int q6apm_dai_prepare(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream)
 {
@@ -378,6 +443,75 @@ static int q6apm_dai_pcm_new(struct snd_soc_component *component, struct snd_soc
 	return snd_pcm_set_fixed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, component->dev, size);
 }
 
+static int q6apm_dai_compr_open(struct snd_soc_component *component,
+				struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd;
+	struct q6apm_dai_data *pdata;
+	struct device *dev = component->dev;
+	int ret, size;
+	int graph_id;
+
+	graph_id = cpu_dai->driver->id;
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata)
+		return -EINVAL;
+
+	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	prtd->cstream = stream;
+	prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id);
+	if (IS_ERR(prtd->graph)) {
+		ret = PTR_ERR(prtd->graph);
+		kfree(prtd);
+		return ret;
+	}
+
+	runtime->private_data = prtd;
+	runtime->dma_bytes = BUFFER_BYTES_MAX;
+	size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, &prtd->dma_buffer);
+	if (ret)
+		return ret;
+
+	if (pdata->sid < 0)
+		prtd->phys = prtd->dma_buffer.addr;
+	else
+		prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
+
+	snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
+	spin_lock_init(&prtd->lock);
+
+	q6apm_enable_compress_module(dev, prtd->graph, true);
+	return 0;
+}
+
+static int q6apm_dai_compr_free(struct snd_soc_component *component,
+				struct snd_compr_stream *stream)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+
+	q6apm_graph_stop(prtd->graph);
+	q6apm_unmap_memory_regions(prtd->graph, SNDRV_PCM_STREAM_PLAYBACK);
+	q6apm_graph_close(prtd->graph);
+	snd_dma_free_pages(&prtd->dma_buffer);
+	prtd->graph = NULL;
+	kfree(prtd);
+	runtime->private_data = NULL;
+
+	return 0;
+}
+static const struct snd_compress_ops q6apm_dai_compress_ops = {
+	.open		= q6apm_dai_compr_open,
+	.free		= q6apm_dai_compr_free,
+};
+
 static const struct snd_soc_component_driver q6apm_fe_dai_component = {
 	.name		= DRV_NAME,
 	.open		= q6apm_dai_open,
@@ -387,6 +521,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = {
 	.hw_params	= q6apm_dai_hw_params,
 	.pointer	= q6apm_dai_pointer,
 	.trigger	= q6apm_dai_trigger,
+	.compress_ops	= &q6apm_dai_compress_ops,
 };
 
 static int q6apm_dai_probe(struct platform_device *pdev)
-- 
2.25.1

[PATCH 09/14] ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks

[Mohammad Rafi Shaik]

Add q6apm get compress DAI capabilities and codec capabilities callbacks
to support compress offload playback.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 51 ++++++++++++++++++++++++++++++++
 1 file changed, 51 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index fd134c268189..54e1aca61e4c 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -29,8 +29,25 @@
 #define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
 #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
 #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
 #define SID_MASK_DEFAULT	0xF
 
+static const struct snd_compr_codec_caps q6apm_compr_caps = {
+	.num_descriptors = 1,
+	.descriptor[0].max_ch = 2,
+	.descriptor[0].sample_rates = {	8000, 11025, 12000, 16000, 22050,
+					24000, 32000, 44100, 48000, 88200,
+					96000, 176400, 192000 },
+	.descriptor[0].num_sample_rates = 13,
+	.descriptor[0].bit_rate[0] = 320,
+	.descriptor[0].bit_rate[1] = 128,
+	.descriptor[0].num_bitrates = 2,
+	.descriptor[0].profiles = 0,
+	.descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
+	.descriptor[0].formats = 0,
+};
+
 enum stream_state {
 	Q6APM_STREAM_IDLE = 0,
 	Q6APM_STREAM_STOPPED,
@@ -507,9 +524,43 @@ static int q6apm_dai_compr_free(struct snd_soc_component *component,
 
 	return 0;
 }
+
+static int q6apm_dai_compr_get_caps(struct snd_soc_component *component,
+				    struct snd_compr_stream *stream,
+				    struct snd_compr_caps *caps)
+{
+	caps->direction = SND_COMPRESS_PLAYBACK;
+	caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+	caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+	caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+	caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+	caps->num_codecs = 3;
+	caps->codecs[0] = SND_AUDIOCODEC_MP3;
+	caps->codecs[1] = SND_AUDIOCODEC_AAC;
+	caps->codecs[2] = SND_AUDIOCODEC_FLAC;
+
+	return 0;
+}
+
+static int q6apm_dai_compr_get_codec_caps(struct snd_soc_component *component,
+					  struct snd_compr_stream *stream,
+					  struct snd_compr_codec_caps *codec)
+{
+	switch (codec->codec) {
+	case SND_AUDIOCODEC_MP3:
+		*codec = q6apm_compr_caps;
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
 static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.open		= q6apm_dai_compr_open,
 	.free		= q6apm_dai_compr_free,
+	.get_caps	= q6apm_dai_compr_get_caps,
+	.get_codec_caps	= q6apm_dai_compr_get_codec_caps,
 };
 
 static const struct snd_soc_component_driver q6apm_fe_dai_component = {
-- 
2.25.1

[PATCH 10/14] ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks

[Mohammad Rafi Shaik]

Add q6apm trigger and pointer compress DAI callbacks to support
compress offload playback.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 68 ++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6apm.h     |  1 +
 2 files changed, 69 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 54e1aca61e4c..f43b60742e2f 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -556,11 +556,79 @@ static int q6apm_dai_compr_get_codec_caps(struct snd_soc_component *component,
 
 	return 0;
 }
+
+static int q6apm_dai_compr_pointer(struct snd_soc_component *component,
+				   struct snd_compr_stream *stream,
+				   struct snd_compr_tstamp *tstamp)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	tstamp->copied_total = prtd->copied_total;
+	tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return 0;
+}
+
+int q6apm_dai_compr_trigger(struct snd_soc_component *component,
+			    struct snd_compr_stream *stream, int cmd)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ret = q6apm_write_async_compr(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	case SND_COMPR_TRIGGER_NEXT_TRACK:
+		prtd->next_track = true;
+		prtd->next_track_stream_id = (prtd->graph->id == 1 ? 2 : 1);
+		break;
+	case SND_COMPR_TRIGGER_DRAIN:
+	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+		prtd->notify_on_drain = true;
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_stream *stream,
+			size_t count)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	prtd->bytes_received += count;
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return count;
+}
+
 static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.open		= q6apm_dai_compr_open,
 	.free		= q6apm_dai_compr_free,
 	.get_caps	= q6apm_dai_compr_get_caps,
 	.get_codec_caps	= q6apm_dai_compr_get_codec_caps,
+	.pointer	= q6apm_dai_compr_pointer,
+	.trigger	= q6apm_dai_compr_trigger,
+	.ack		= q6apm_dai_compr_ack,
 };
 
 static const struct snd_soc_component_driver q6apm_fe_dai_component = {
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index 630c2bca0f06..e0247e6e4fd2 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -46,6 +46,7 @@
 
 #define APM_MAX_SESSIONS			8
 #define APM_LAST_BUFFER_FLAG			BIT(30)
+#define NO_TIMESTAMP				0xFF00
 
 struct q6apm {
 	struct device *dev;
-- 
2.25.1

[PATCH 11/14] ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks

[Mohammad Rafi Shaik]

Add q6apm compress DAI callbacks for setting params and metadata to support
compress offload playback.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 121 +++++++++++++++++++++++++++++++
 1 file changed, 121 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index f43b60742e2f..8ee14822362b 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -621,6 +621,125 @@ int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_st
 	return count;
 }
 
+static int __q6apm_dai_compr_set_codec_params(struct snd_soc_component *component,
+					      struct snd_compr_stream *stream,
+					      struct snd_codec *codec,
+					      int stream_id)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	struct device *dev = component->dev;
+	union snd_codec_options *codec_options;
+
+	codec_options = &(prtd->codec.options);
+
+	memcpy(&prtd->codec, codec, sizeof(*codec));
+	q6apm_set_real_module_id(dev, prtd->graph, codec->id);
+
+	return 0;
+}
+
+static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
+				      struct snd_compr_stream *stream,
+				      struct snd_compr_params *params)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	struct q6apm_dai_data *pdata;
+	struct audioreach_module_config cfg;
+	struct snd_codec *codec = &params->codec;
+	int dir = stream->direction;
+	int ret;
+
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata)
+		return -EINVAL;
+
+	prtd->periods = runtime->fragments;
+	prtd->pcm_count = runtime->fragment_size;
+	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+	prtd->bits_per_sample = 16;
+
+	prtd->pos = 0;
+
+	if (prtd->next_track != true) {
+		ret = __q6apm_dai_compr_set_codec_params(component, stream,
+				&params->codec, 0);
+		if (ret)
+			return ret;
+
+		cfg.direction = dir;
+		cfg.sample_rate = codec->sample_rate;
+		cfg.num_channels = 2;
+		cfg.bit_width = prtd->bits_per_sample;
+		cfg.fmt = codec->id;
+		memcpy(&cfg.codec, codec, sizeof(*codec));
+
+		ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
+		if (ret < 0)
+			return ret;
+
+		ret = q6apm_graph_media_format_pcm(prtd->graph, &cfg);
+		if (ret)
+			return ret;
+
+		ret = q6apm_map_memory_regions(prtd->graph, SNDRV_PCM_STREAM_PLAYBACK,
+					       prtd->phys, (prtd->pcm_size / prtd->periods),
+					       prtd->periods);
+		if (ret < 0)
+			return -ENOMEM;
+
+		ret = q6apm_graph_prepare(prtd->graph);
+		if (ret)
+			return ret;
+		ret = q6apm_graph_start(prtd->graph);
+		if (ret)
+			return ret;
+
+	} else {
+		cfg.direction = dir;
+		cfg.sample_rate = codec->sample_rate;
+		cfg.num_channels = 2;
+		cfg.bit_width = prtd->bits_per_sample;
+		cfg.fmt = codec->id;
+		memcpy(&cfg.codec, codec, sizeof(*codec));
+
+		ret = audioreach_compr_set_param(prtd->graph,  &cfg);
+		if (ret < 0)
+			return ret;
+	}
+	prtd->state = Q6APM_STREAM_RUNNING;
+
+	return 0;
+}
+
+static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
+					struct snd_compr_stream *stream,
+					struct snd_compr_metadata *metadata)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	int ret = 0;
+
+	switch (metadata->key) {
+	case SNDRV_COMPRESS_ENCODER_PADDING:
+		prtd->trailing_samples_drop = metadata->value[0];
+		q6apm_remove_trailing_silence(component->dev, prtd->graph,
+					      prtd->trailing_samples_drop);
+		break;
+	case SNDRV_COMPRESS_ENCODER_DELAY:
+		prtd->initial_samples_drop = metadata->value[0];
+		q6apm_remove_initial_silence(component->dev, prtd->graph,
+					     prtd->initial_samples_drop);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
 static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.open		= q6apm_dai_compr_open,
 	.free		= q6apm_dai_compr_free,
@@ -629,6 +748,8 @@ static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.pointer	= q6apm_dai_compr_pointer,
 	.trigger	= q6apm_dai_compr_trigger,
 	.ack		= q6apm_dai_compr_ack,
+	.set_params	= q6apm_dai_compr_set_params,
+	.set_metadata	= q6apm_dai_compr_set_metadata,
 };
 
 static const struct snd_soc_component_driver q6apm_fe_dai_component = {
-- 
2.25.1

[PATCH 12/14] ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks

[Mohammad Rafi Shaik]

Add q6apm mmap and copy compress DAI callbacks to support compress
offload playback.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 81 ++++++++++++++++++++++++++++++++
 1 file changed, 81 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 8ee14822362b..2c0aa2f4caaf 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -740,6 +740,85 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
 	return ret;
 }
 
+static int q6apm_dai_compr_mmap(struct snd_soc_component *component,
+				struct snd_compr_stream *stream,
+				struct vm_area_struct *vma)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	struct device *dev = component->dev;
+
+	return dma_mmap_coherent(dev, vma, prtd->dma_buffer.area, prtd->dma_buffer.addr,
+				 prtd->dma_buffer.bytes);
+}
+
+static int q6apm_compr_copy(struct snd_soc_component *component,
+			    struct snd_compr_stream *stream, char __user *buf,
+			    size_t count)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	void *dstn;
+	unsigned long flags;
+	size_t copy;
+	u32 wflags = 0;
+	u32 app_pointer;
+	u32 bytes_received;
+	uint32_t bytes_to_write;
+	int avail, bytes_in_flight = 0;
+
+	bytes_received = prtd->bytes_received;
+
+	/**
+	 * Make sure that next track data pointer is aligned at 32 bit boundary
+	 * This is a Mandatory requirement from DSP data buffers alignment
+	 */
+	if (prtd->next_track)
+		bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+	app_pointer = bytes_received/prtd->pcm_size;
+	app_pointer = bytes_received -  (app_pointer * prtd->pcm_size);
+	dstn = prtd->dma_buffer.area + app_pointer;
+
+	if (count < prtd->pcm_size - app_pointer) {
+		if (copy_from_user(dstn, buf, count))
+			return -EFAULT;
+	} else {
+		copy = prtd->pcm_size - app_pointer;
+		if (copy_from_user(dstn, buf, copy))
+			return -EFAULT;
+		if (copy_from_user(prtd->dma_buffer.area, buf + copy, count - copy))
+			return -EFAULT;
+	}
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+	if (prtd->next_track) {
+		prtd->next_track = false;
+		prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
+		prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+	}
+
+	prtd->bytes_received = bytes_received + count;
+
+	/* Kick off the data to dsp if its starving!! */
+	if (prtd->state == Q6APM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+		bytes_to_write = prtd->pcm_count;
+		avail = prtd->bytes_received - prtd->bytes_sent;
+
+		if (avail < prtd->pcm_count)
+			bytes_to_write = avail;
+
+		q6apm_write_async_compr(prtd->graph, bytes_to_write, 0, 0, wflags);
+		prtd->bytes_sent += bytes_to_write;
+	}
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return count;
+}
+
 static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.open		= q6apm_dai_compr_open,
 	.free		= q6apm_dai_compr_free,
@@ -750,6 +829,8 @@ static const struct snd_compress_ops q6apm_dai_compress_ops = {
 	.ack		= q6apm_dai_compr_ack,
 	.set_params	= q6apm_dai_compr_set_params,
 	.set_metadata	= q6apm_dai_compr_set_metadata,
+	.mmap		= q6apm_dai_compr_mmap,
+	.copy		= q6apm_compr_copy,
 };
 
 static const struct snd_soc_component_driver q6apm_fe_dai_component = {
-- 
2.25.1

[PATCH 13/14] ASoC: qdsp6: audioreach: Add MP3, AAC and FLAC compress format support

[Mohammad Rafi Shaik]

Add support for handling compressed formats such as MP3, AAC and FLAC.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c | 106 ++++++++++++++++++++++++------
 1 file changed, 86 insertions(+), 20 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 7c45c36e9156..250ed828c7d3 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -852,6 +852,68 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
 	return rc;
 }
 
+static int  audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
+					      void *p, struct audioreach_module_config *mcfg)
+{
+	struct payload_media_fmt_aac_t *aac_cfg;
+	struct payload_media_fmt_pcm *mp3_cfg;
+	struct payload_media_fmt_flac_t *flac_cfg;
+	int ret = 0;
+
+	switch (mcfg->fmt) {
+	case SND_AUDIOCODEC_MP3:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_MP3;
+		media_fmt_hdr->payload_size = 0;
+		p = p + sizeof(*media_fmt_hdr);
+		mp3_cfg = p;
+		mp3_cfg->sample_rate = mcfg->sample_rate;
+		mp3_cfg->bit_width = mcfg->bit_width;
+		mp3_cfg->alignment = PCM_LSB_ALIGNED;
+		mp3_cfg->bits_per_sample = mcfg->bit_width;
+		mp3_cfg->q_factor = mcfg->bit_width - 1;
+		mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
+		mp3_cfg->num_channels = mcfg->num_channels;
+
+		if (mcfg->num_channels == 1) {
+			mp3_cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+		} else if (mcfg->num_channels == 2) {
+			mp3_cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+			mp3_cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+		}
+		break;
+	case SND_AUDIOCODEC_AAC:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_AAC;
+		media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_aac_t);
+		p = p + sizeof(*media_fmt_hdr);
+		aac_cfg = p;
+		aac_cfg->aac_fmt_flag = 0;
+		aac_cfg->audio_obj_type = 5;
+		aac_cfg->num_channels = mcfg->num_channels;
+		aac_cfg->total_size_of_PCE_bits = 0;
+		aac_cfg->sample_rate = mcfg->sample_rate;
+		break;
+	case SND_AUDIOCODEC_FLAC:
+		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_FLAC;
+		media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_flac_t);
+		p = p + sizeof(*media_fmt_hdr);
+		flac_cfg = p;
+		flac_cfg->sample_size = mcfg->codec.options.flac_d.sample_size;
+		flac_cfg->num_channels = mcfg->num_channels;
+		flac_cfg->min_blk_size = mcfg->codec.options.flac_d.min_blk_size;
+		flac_cfg->max_blk_size = mcfg->codec.options.flac_d.max_blk_size;
+		flac_cfg->sample_rate = mcfg->sample_rate;
+		flac_cfg->min_frame_size = mcfg->codec.options.flac_d.min_frame_size;
+		flac_cfg->max_frame_size = mcfg->codec.options.flac_d.max_frame_size;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return ret;
+}
+
 static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
 					   struct audioreach_module *module,
 					   struct audioreach_module_config *cfg)
@@ -1055,26 +1117,29 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
 	p = p + APM_MODULE_PARAM_DATA_SIZE;
 
 	header = p;
-	header->data_format = DATA_FORMAT_FIXED_POINT;
-	header->fmt_id = MEDIA_FMT_ID_PCM;
-	header->payload_size = payload_size - sizeof(*header);
+	if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
+		header->data_format = DATA_FORMAT_FIXED_POINT;
+		header->fmt_id =  MEDIA_FMT_ID_PCM;
+		header->payload_size = payload_size - sizeof(*header);
 
-	p = p + sizeof(*header);
-	cfg = p;
-	cfg->sample_rate = mcfg->sample_rate;
-	cfg->bit_width = mcfg->bit_width;
-	cfg->alignment = PCM_LSB_ALIGNED;
-	cfg->bits_per_sample = mcfg->bit_width;
-	cfg->q_factor = mcfg->bit_width - 1;
-	cfg->endianness = PCM_LITTLE_ENDIAN;
-	cfg->num_channels = mcfg->num_channels;
-
-	if (mcfg->num_channels == 1) {
-		cfg->channel_mapping[0] =  PCM_CHANNEL_L;
-	} else if (num_channels == 2) {
-		cfg->channel_mapping[0] =  PCM_CHANNEL_L;
-		cfg->channel_mapping[1] =  PCM_CHANNEL_R;
-	}
+		p = p + sizeof(*header);
+		cfg = p;
+		cfg->sample_rate = mcfg->sample_rate;
+		cfg->bit_width = mcfg->bit_width;
+		cfg->alignment = PCM_LSB_ALIGNED;
+		cfg->bits_per_sample = mcfg->bit_width;
+		cfg->q_factor = mcfg->bit_width - 1;
+		cfg->endianness = PCM_LITTLE_ENDIAN;
+		cfg->num_channels = mcfg->num_channels;
+
+		if (mcfg->num_channels == 1)
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+		else if (num_channels == 2) {
+			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
+			cfg->channel_mapping[1] =  PCM_CHANNEL_R;
+		}
+	} else
+		audioreach_set_compr_media_format(header, p, mcfg);
 
 	rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);
 
@@ -1401,7 +1466,8 @@ int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_modu
 			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
 			cfg->channel_mapping[1] =  PCM_CHANNEL_R;
 		}
-	}
+	} else
+		audioreach_set_compr_media_format(header, p, mcfg);
 
 	rc = gpr_send_port_pkt(graph->port, pkt);
 	kfree(pkt);
-- 
2.25.1

[PATCH 14/14] ASoC: q6dsp: audioreach: Add gapless feature support

[Mohammad Rafi Shaik]

Add support for setting EOS delay command and receive the
EOS response from ADSP, for seamless compress offload
playback feature.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/audioreach.c | 40 +++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/audioreach.h |  6 +++++
 sound/soc/qcom/qdsp6/q6apm.c      |  3 +++
 3 files changed, 49 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 250ed828c7d3..71ff705e27cb 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -852,6 +852,43 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
 	return rc;
 }
 
+static int audioreach_gapless_set_media_format(struct q6apm_graph *graph,
+					   struct audioreach_module *module,
+					   struct audioreach_module_config *cfg)
+{
+	struct apm_module_param_data *param_data;
+	struct param_id_gapless_early_eos_delay_t *media_format;
+	int payload_size;
+	struct gpr_pkt *pkt;
+	int rc;
+	void *p;
+
+	payload_size = sizeof(struct param_id_gapless_early_eos_delay_t) +
+		APM_MODULE_PARAM_DATA_SIZE;
+
+	pkt = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
+	if (IS_ERR(pkt))
+		return PTR_ERR(pkt);
+
+	p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
+
+	param_data = p;
+	param_data->module_instance_id = module->instance_id;
+	param_data->error_code = 0;
+	param_data->param_id = PARAM_ID_EARLY_EOS_DELAY;
+	param_data->param_size = sizeof(struct param_id_gapless_early_eos_delay_t);
+	p = p + APM_MODULE_PARAM_DATA_SIZE;
+	media_format = p;
+
+	media_format->early_eos_delay_ms = EARLY_EOS_DELAY_MS;
+
+	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
+
+	kfree(pkt);
+
+	return rc;
+}
+
 static int  audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
 					      void *p, struct audioreach_module_config *mcfg)
 {
@@ -1249,6 +1286,9 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
 	case MODULE_ID_MFC:
 		rc = audioreach_mfc_set_media_format(graph, module, cfg);
 		break;
+	case MODULE_ID_GAPLESS:
+		rc = audioreach_gapless_set_media_format(graph, module, cfg);
+		break;
 	default:
 		rc = 0;
 	}
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 044994ca4811..a9d4f70ad3e0 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -543,6 +543,8 @@ struct param_id_sal_limiter_enable {
 } __packed;
 
 #define PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT	0x08001024
+#define PARAM_ID_EARLY_EOS_DELAY		0x0800114C
+#define EARLY_EOS_DELAY_MS			150
 
 struct param_id_mfc_media_format {
 	uint32_t sample_rate;
@@ -551,6 +553,10 @@ struct param_id_mfc_media_format {
 	uint16_t channel_mapping[];
 } __packed;
 
+struct param_id_gapless_early_eos_delay_t {
+	uint32_t early_eos_delay_ms;
+} __packed;
+
 struct media_format {
 	uint32_t data_format;
 	uint32_t fmt_id;
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 1a6c7108bae0..37ded6e0f4ea 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -605,6 +605,9 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		}
 		break;
 	case DATA_CMD_WR_SH_MEM_EP_EOS_RENDERED:
+		client_event = APM_CLIENT_EVENT_CMD_EOS_DONE;
+		if (graph->cb)
+			graph->cb(client_event, hdr->token, data->payload, graph->priv);
 		break;
 	case GPR_BASIC_RSP_RESULT:
 		switch (result->opcode) {
-- 
2.25.1

Re: [PATCH 02/14] ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 569 bytes --]

On Wed, Feb 01, 2023 at 07:19:35PM +0530, Mohammad Rafi Shaik wrote:

> +#define DEFAULT_SAMPLE_RATE_48K	48000

Why are we bothering with a define here given that the define also
encodes the value and it's only used in once place?

>  	for_each_card_prelinks(card, i, link) {
>  		link->init = sc7280_init;
>  		link->ops = &sc7280_ops;
> +		if (link->no_pcm == 1)
> +			link->be_hw_params_fixup = sc7280_snd_be_hw_params_fixup;

We only set the fixup in the case where there's no PCM but we removed
the constraint in all cases - isn't the constraint needed otherwise?

[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]

Re: [PATCH 11/14] ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks

[kernel test robot]

Hi Mohammad,

Thank you for the patch! Perhaps something to improve:

[auto build test WARNING on broonie-sound/for-next]
[also build test WARNING on next-20230201]
[cannot apply to tiwai-sound/for-next tiwai-sound/for-linus linus/master v6.2-rc6]
[If your patch is applied to the wrong git tree, kindly drop us a note.
And when submitting patch, we suggest to use '--base' as documented in
https://git-scm.com/docs/git-format-patch#_base_tree_information]

url:    https://github.com/intel-lab-lkp/linux/commits/Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
base:   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
patch link:    https://lore.kernel.org/r/20230201134947.1638197-12-quic_mohs%40quicinc.com
patch subject: [PATCH 11/14] ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks
config: m68k-allyesconfig (https://download.01.org/0day-ci/archive/20230201/202302012348.LL8vhyj4-lkp@intel.com/config)
compiler: m68k-linux-gcc (GCC) 12.1.0
reproduce (this is a W=1 build):
        wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
        chmod +x ~/bin/make.cross
        # https://github.com/intel-lab-lkp/linux/commit/f6204693a956c267d6cdfd17f5c27da0f4594ca3
        git remote add linux-review https://github.com/intel-lab-lkp/linux
        git fetch --no-tags linux-review Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
        git checkout f6204693a956c267d6cdfd17f5c27da0f4594ca3
        # save the config file
        mkdir build_dir && cp config build_dir/.config
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k olddefconfig
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k SHELL=/bin/bash sound/soc/qcom/qdsp6/

If you fix the issue, kindly add following tag where applicable
| Reported-by: kernel test robot <lkp@intel.com>

All warnings (new ones prefixed by >>):

   sound/soc/qcom/qdsp6/q6apm-dai.c:152:6: warning: no previous prototype for 'event_handler_compr' [-Wmissing-prototypes]
     152 | void event_handler_compr(uint32_t opcode, uint32_t token,
         |      ^~~~~~~~~~~~~~~~~~~
   sound/soc/qcom/qdsp6/q6apm-dai.c:576:5: warning: no previous prototype for 'q6apm_dai_compr_trigger' [-Wmissing-prototypes]
     576 | int q6apm_dai_compr_trigger(struct snd_soc_component *component,
         |     ^~~~~~~~~~~~~~~~~~~~~~~
   sound/soc/qcom/qdsp6/q6apm-dai.c:610:5: warning: no previous prototype for 'q6apm_dai_compr_ack' [-Wmissing-prototypes]
     610 | int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_stream *stream,
         |     ^~~~~~~~~~~~~~~~~~~
   sound/soc/qcom/qdsp6/q6apm-dai.c: In function '__q6apm_dai_compr_set_codec_params':
>> sound/soc/qcom/qdsp6/q6apm-dai.c:632:34: warning: variable 'codec_options' set but not used [-Wunused-but-set-variable]
     632 |         union snd_codec_options *codec_options;
         |                                  ^~~~~~~~~~~~~


vim +/codec_options +632 sound/soc/qcom/qdsp6/q6apm-dai.c

   623	
   624	static int __q6apm_dai_compr_set_codec_params(struct snd_soc_component *component,
   625						      struct snd_compr_stream *stream,
   626						      struct snd_codec *codec,
   627						      int stream_id)
   628	{
   629		struct snd_compr_runtime *runtime = stream->runtime;
   630		struct q6apm_dai_rtd *prtd = runtime->private_data;
   631		struct device *dev = component->dev;
 > 632		union snd_codec_options *codec_options;
   633	
   634		codec_options = &(prtd->codec.options);
   635	
   636		memcpy(&prtd->codec, codec, sizeof(*codec));
   637		q6apm_set_real_module_id(dev, prtd->graph, codec->id);
   638	
   639		return 0;
   640	}
   641	

-- 
0-DAY CI Kernel Test Service
https://github.com/intel/lkp-tests

Re: [PATCH 04/14] ASoC: q6dsp: audioreach: Add support for compress offload commands

[kernel test robot]

Hi Mohammad,

Thank you for the patch! Perhaps something to improve:

[auto build test WARNING on broonie-sound/for-next]
[also build test WARNING on next-20230201]
[cannot apply to tiwai-sound/for-next tiwai-sound/for-linus linus/master v6.2-rc6]
[If your patch is applied to the wrong git tree, kindly drop us a note.
And when submitting patch, we suggest to use '--base' as documented in
https://git-scm.com/docs/git-format-patch#_base_tree_information]

url:    https://github.com/intel-lab-lkp/linux/commits/Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
base:   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
patch link:    https://lore.kernel.org/r/20230201134947.1638197-5-quic_mohs%40quicinc.com
patch subject: [PATCH 04/14] ASoC: q6dsp: audioreach: Add support for compress offload commands
config: m68k-allyesconfig (https://download.01.org/0day-ci/archive/20230201/202302012309.BtyJn8FN-lkp@intel.com/config)
compiler: m68k-linux-gcc (GCC) 12.1.0
reproduce (this is a W=1 build):
        wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
        chmod +x ~/bin/make.cross
        # https://github.com/intel-lab-lkp/linux/commit/6a2982489303bcf32b927da80e4baffae58437e0
        git remote add linux-review https://github.com/intel-lab-lkp/linux
        git fetch --no-tags linux-review Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
        git checkout 6a2982489303bcf32b927da80e4baffae58437e0
        # save the config file
        mkdir build_dir && cp config build_dir/.config
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k olddefconfig
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k SHELL=/bin/bash sound/soc/qcom/qdsp6/

If you fix the issue, kindly add following tag where applicable
| Reported-by: kernel test robot <lkp@intel.com>

All warnings (new ones prefixed by >>):

>> sound/soc/qcom/qdsp6/audioreach.c:1262:5: warning: no previous prototype for 'audioreach_send_u32_param' [-Wmissing-prototypes]
    1262 | int audioreach_send_u32_param(struct q6apm *apm, struct audioreach_module *module,
         |     ^~~~~~~~~~~~~~~~~~~~~~~~~


vim +/audioreach_send_u32_param +1262 sound/soc/qcom/qdsp6/audioreach.c

  1261	
> 1262	int audioreach_send_u32_param(struct q6apm *apm, struct audioreach_module *module,
  1263				      uint32_t param_id, uint32_t param_val)
  1264	{
  1265		struct apm_module_param_data *param_data;
  1266		struct gpr_pkt *pkt;
  1267		uint32_t *param;
  1268		int rc, payload_size;
  1269		void *p;
  1270	
  1271		payload_size = sizeof(uint32_t) + APM_MODULE_PARAM_DATA_SIZE;
  1272		p = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
  1273		if (IS_ERR(p))
  1274			return -ENOMEM;
  1275	
  1276		pkt = p;
  1277		p = p + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
  1278	
  1279		param_data = p;
  1280		param_data->module_instance_id = module->instance_id;
  1281		param_data->error_code = 0;
  1282		param_data->param_id = param_id;
  1283		param_data->param_size = payload_size - APM_MODULE_PARAM_DATA_SIZE;
  1284	
  1285		p = p + APM_MODULE_PARAM_DATA_SIZE;
  1286		param = p;
  1287		*param = param_val;
  1288		rc = q6apm_send_cmd_sync(apm, pkt, 0);
  1289	
  1290		kfree(pkt);
  1291	
  1292		return rc;
  1293	}
  1294	EXPORT_SYMBOL_GPL(audioreach_send_u32_param);
  1295	

-- 
0-DAY CI Kernel Test Service
https://github.com/intel/lkp-tests

Re: [PATCH 08/14] ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks

[kernel test robot]

Hi Mohammad,

Thank you for the patch! Perhaps something to improve:

[auto build test WARNING on broonie-sound/for-next]
[also build test WARNING on next-20230201]
[cannot apply to tiwai-sound/for-next tiwai-sound/for-linus linus/master v6.2-rc6]
[If your patch is applied to the wrong git tree, kindly drop us a note.
And when submitting patch, we suggest to use '--base' as documented in
https://git-scm.com/docs/git-format-patch#_base_tree_information]

url:    https://github.com/intel-lab-lkp/linux/commits/Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
base:   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
patch link:    https://lore.kernel.org/r/20230201134947.1638197-9-quic_mohs%40quicinc.com
patch subject: [PATCH 08/14] ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks
config: m68k-allyesconfig (https://download.01.org/0day-ci/archive/20230201/202302012337.pC5Q3lLy-lkp@intel.com/config)
compiler: m68k-linux-gcc (GCC) 12.1.0
reproduce (this is a W=1 build):
        wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
        chmod +x ~/bin/make.cross
        # https://github.com/intel-lab-lkp/linux/commit/78a6016e006a8e405679fd335940ee710416c43f
        git remote add linux-review https://github.com/intel-lab-lkp/linux
        git fetch --no-tags linux-review Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
        git checkout 78a6016e006a8e405679fd335940ee710416c43f
        # save the config file
        mkdir build_dir && cp config build_dir/.config
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k olddefconfig
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k SHELL=/bin/bash sound/soc/qcom/qdsp6/

If you fix the issue, kindly add following tag where applicable
| Reported-by: kernel test robot <lkp@intel.com>

All warnings (new ones prefixed by >>):

>> sound/soc/qcom/qdsp6/q6apm-dai.c:135:6: warning: no previous prototype for 'event_handler_compr' [-Wmissing-prototypes]
     135 | void event_handler_compr(uint32_t opcode, uint32_t token,
         |      ^~~~~~~~~~~~~~~~~~~


vim +/event_handler_compr +135 sound/soc/qcom/qdsp6/q6apm-dai.c

   134	
 > 135	void event_handler_compr(uint32_t opcode, uint32_t token,
   136					uint32_t *payload, void *priv)
   137	{
   138		struct q6apm_dai_rtd *prtd = priv;
   139		struct snd_compr_stream *substream = prtd->cstream;
   140		unsigned long flags;
   141		uint32_t wflags = 0;
   142		uint64_t avail;
   143		uint32_t bytes_written, bytes_to_write;
   144		bool is_last_buffer = false;
   145	
   146		switch (opcode) {
   147		case APM_CLIENT_EVENT_CMD_EOS_DONE:
   148			spin_lock_irqsave(&prtd->lock, flags);
   149			if (prtd->notify_on_drain) {
   150				snd_compr_drain_notify(prtd->cstream);
   151				prtd->notify_on_drain = false;
   152			} else {
   153				prtd->state = Q6APM_STREAM_STOPPED;
   154			}
   155			spin_unlock_irqrestore(&prtd->lock, flags);
   156			break;
   157		case APM_CLIENT_EVENT_DATA_WRITE_DONE:
   158			spin_lock_irqsave(&prtd->lock, flags);
   159			bytes_written = token >> APM_WRITE_TOKEN_LEN_SHIFT;
   160			prtd->copied_total += bytes_written;
   161			snd_compr_fragment_elapsed(substream);
   162	
   163			if (prtd->state != Q6APM_STREAM_RUNNING) {
   164				spin_unlock_irqrestore(&prtd->lock, flags);
   165				break;
   166			}
   167	
   168			avail = prtd->bytes_received - prtd->bytes_sent;
   169	
   170			if (avail > prtd->pcm_count) {
   171				bytes_to_write = prtd->pcm_count;
   172			} else {
   173				if (substream->partial_drain || prtd->notify_on_drain)
   174					is_last_buffer = true;
   175				bytes_to_write = avail;
   176			}
   177	
   178			if (bytes_to_write) {
   179				if (substream->partial_drain && is_last_buffer)
   180					wflags |= APM_LAST_BUFFER_FLAG;
   181	
   182				q6apm_write_async_compr(prtd->graph,
   183							bytes_to_write, 0, 0, wflags);
   184	
   185				prtd->bytes_sent += bytes_to_write;
   186	
   187				if (prtd->notify_on_drain && is_last_buffer)
   188					audioreach_shared_memory_send_eos(prtd->graph);
   189			}
   190	
   191			spin_unlock_irqrestore(&prtd->lock, flags);
   192			break;
   193		default:
   194			break;
   195		}
   196	}
   197	

-- 
0-DAY CI Kernel Test Service
https://github.com/intel/lkp-tests

Re: [PATCH 10/14] ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks

[kernel test robot]

Hi Mohammad,

Thank you for the patch! Perhaps something to improve:

[auto build test WARNING on broonie-sound/for-next]
[also build test WARNING on next-20230201]
[cannot apply to tiwai-sound/for-next tiwai-sound/for-linus linus/master v6.2-rc6]
[If your patch is applied to the wrong git tree, kindly drop us a note.
And when submitting patch, we suggest to use '--base' as documented in
https://git-scm.com/docs/git-format-patch#_base_tree_information]

url:    https://github.com/intel-lab-lkp/linux/commits/Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
base:   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
patch link:    https://lore.kernel.org/r/20230201134947.1638197-11-quic_mohs%40quicinc.com
patch subject: [PATCH 10/14] ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks
config: m68k-allyesconfig (https://download.01.org/0day-ci/archive/20230202/202302020014.J221iZHe-lkp@intel.com/config)
compiler: m68k-linux-gcc (GCC) 12.1.0
reproduce (this is a W=1 build):
        wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
        chmod +x ~/bin/make.cross
        # https://github.com/intel-lab-lkp/linux/commit/2b44c079fb2d53ef9e13fc7d7b257fd4c6f4b56a
        git remote add linux-review https://github.com/intel-lab-lkp/linux
        git fetch --no-tags linux-review Mohammad-Rafi-Shaik/ALSA-compress-Update-compress-set-params-for-gapless-playback/20230201-215622
        git checkout 2b44c079fb2d53ef9e13fc7d7b257fd4c6f4b56a
        # save the config file
        mkdir build_dir && cp config build_dir/.config
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k olddefconfig
        COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 make.cross W=1 O=build_dir ARCH=m68k SHELL=/bin/bash sound/soc/

If you fix the issue, kindly add following tag where applicable
| Reported-by: kernel test robot <lkp@intel.com>

All warnings (new ones prefixed by >>):

   sound/soc/qcom/qdsp6/q6apm-dai.c:152:6: warning: no previous prototype for 'event_handler_compr' [-Wmissing-prototypes]
     152 | void event_handler_compr(uint32_t opcode, uint32_t token,
         |      ^~~~~~~~~~~~~~~~~~~
>> sound/soc/qcom/qdsp6/q6apm-dai.c:576:5: warning: no previous prototype for 'q6apm_dai_compr_trigger' [-Wmissing-prototypes]
     576 | int q6apm_dai_compr_trigger(struct snd_soc_component *component,
         |     ^~~~~~~~~~~~~~~~~~~~~~~
>> sound/soc/qcom/qdsp6/q6apm-dai.c:610:5: warning: no previous prototype for 'q6apm_dai_compr_ack' [-Wmissing-prototypes]
     610 | int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_stream *stream,
         |     ^~~~~~~~~~~~~~~~~~~


vim +/q6apm_dai_compr_trigger +576 sound/soc/qcom/qdsp6/q6apm-dai.c

   575	
 > 576	int q6apm_dai_compr_trigger(struct snd_soc_component *component,
   577				    struct snd_compr_stream *stream, int cmd)
   578	{
   579		struct snd_compr_runtime *runtime = stream->runtime;
   580		struct q6apm_dai_rtd *prtd = runtime->private_data;
   581		int ret = 0;
   582	
   583		switch (cmd) {
   584		case SNDRV_PCM_TRIGGER_START:
   585		case SNDRV_PCM_TRIGGER_RESUME:
   586		case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
   587			ret = q6apm_write_async_compr(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP);
   588			break;
   589		case SNDRV_PCM_TRIGGER_STOP:
   590			break;
   591		case SNDRV_PCM_TRIGGER_SUSPEND:
   592		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
   593			break;
   594		case SND_COMPR_TRIGGER_NEXT_TRACK:
   595			prtd->next_track = true;
   596			prtd->next_track_stream_id = (prtd->graph->id == 1 ? 2 : 1);
   597			break;
   598		case SND_COMPR_TRIGGER_DRAIN:
   599		case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
   600			prtd->notify_on_drain = true;
   601			break;
   602		default:
   603			ret = -EINVAL;
   604			break;
   605		}
   606	
   607		return ret;
   608	}
   609	
 > 610	int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_stream *stream,
   611				size_t count)
   612	{
   613		struct snd_compr_runtime *runtime = stream->runtime;
   614		struct q6apm_dai_rtd *prtd = runtime->private_data;
   615		unsigned long flags;
   616	
   617		spin_lock_irqsave(&prtd->lock, flags);
   618		prtd->bytes_received += count;
   619		spin_unlock_irqrestore(&prtd->lock, flags);
   620	
   621		return count;
   622	}
   623	

-- 
0-DAY CI Kernel Test Service
https://github.com/intel/lkp-tests

Re: [PATCH 02/14] ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback

[Mohammad Rafi Shaik]

On 2/1/2023 8:10 PM, Mark Brown wrote:
> On Wed, Feb 01, 2023 at 07:19:35PM +0530, Mohammad Rafi Shaik wrote:
>
>> +#define DEFAULT_SAMPLE_RATE_48K	48000
> Why are we bothering with a define here given that the define also
> encodes the value and it's only used in once place?
okay, will remove it.
>
>>   	for_each_card_prelinks(card, i, link) {
>>   		link->init = sc7280_init;
>>   		link->ops = &sc7280_ops;
>> +		if (link->no_pcm == 1)
>> +			link->be_hw_params_fixup = sc7280_snd_be_hw_params_fixup;
> We only set the fixup in the case where there's no PCM but we removed
> the constraint in all cases - isn't the constraint needed otherwise?
okay, will add conditional check for constraint and will only do if 
no_pcm is zero.