[alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[Vinod Koul]

The current design of sending codec parameters assumes that decoders
will have parsers so they can parse the encoded stream for parameters
and configure the decoder.

But this assumption may not be universally true and we know some DSPs
which do not contain the parsers so additional parameters are required
to be passed.

So add these parameters starting with FLAC decoder. The size of
snd_codec_options is still 120 bytes after this change (due to this
being a union)

I think we should also bump the (minor) version if this proposal is
acceptable so the userspace can check and populate flac specific structure.

Along, with the core header change, patches are added to support FLAC
in Qualcomm drivers. This was tested on 96boards db845c

Srinivas Kandagatla (1):
  ASoC: qcom: q6asm: add support to flac config

Vinod Koul (2):
  ALSA: compress: add flac decoder params
  ASoC: qcom: q6asm-dai: add support to flac decoder

 include/uapi/sound/compress_params.h | 10 +++++
 sound/soc/qcom/qdsp6/q6asm-dai.c     | 35 +++++++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.c         | 55 ++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h         | 15 ++++++++
 4 files changed, 114 insertions(+), 1 deletion(-)

-- 
2.23.0

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[alsa-devel] [RFC PATCH 1/3] ALSA: compress: add flac decoder params

[Vinod Koul]

The current design of sending codec parameters assumes that decoders
will have parsers so they can parse the encoded stream for parameters
and configure the decoder.

But this assumption may not be universally true and we know some DSPs
which do not contain the parsers so additional parameters are required
to be passed.

So add these parameters starting with FLAC decoder. The size of
snd_codec_options is still 120 bytes after this change (due to this
being a union)

Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 10 ++++++++++
 1 file changed, 10 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 3d4d6de66a17..9c96fb0e4d90 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -317,12 +317,22 @@ struct snd_enc_generic {
 	__s32 reserved[15];	/* Can be used for SND_AUDIOCODEC_BESPOKE */
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_flac {
+	__u16 sample_size;
+	__u16 min_blk_size;
+	__u16 max_blk_size;
+	__u16 min_frame_size;
+	__u16 max_frame_size;
+	__u16 reserved;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
 	struct snd_enc_real real;
 	struct snd_enc_flac flac;
 	struct snd_enc_generic generic;
+	struct snd_dec_flac flac_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.23.0

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[alsa-devel] Applied "ALSA: compress: add flac decoder params" to the asoc tree

[Mark Brown]

The patch

   ALSA: compress: add flac decoder params

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From f6ff3e958e0ec5949434b7db3c42c3604c85c8a0 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Fri, 15 Nov 2019 15:57:03 +0530
Subject: [PATCH] ALSA: compress: add flac decoder params

The current design of sending codec parameters assumes that decoders
will have parsers so they can parse the encoded stream for parameters
and configure the decoder.

But this assumption may not be universally true and we know some DSPs
which do not contain the parsers so additional parameters are required
to be passed.

So add these parameters starting with FLAC decoder. The size of
snd_codec_options is still 120 bytes after this change (due to this
being a union)

Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-2-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/uapi/sound/compress_params.h | 10 ++++++++++
 1 file changed, 10 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 3d4d6de66a17..9c96fb0e4d90 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -317,12 +317,22 @@ struct snd_enc_generic {
 	__s32 reserved[15];	/* Can be used for SND_AUDIOCODEC_BESPOKE */
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_flac {
+	__u16 sample_size;
+	__u16 min_blk_size;
+	__u16 max_blk_size;
+	__u16 min_frame_size;
+	__u16 max_frame_size;
+	__u16 reserved;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
 	struct snd_enc_real real;
 	struct snd_enc_flac flac;
 	struct snd_enc_generic generic;
+	struct snd_dec_flac flac_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.20.1

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[alsa-devel] [RFC PATCH 2/3] ASoC: qcom: q6asm: add support to flac config

[Vinod Koul]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Qualcomm DSPs expect flac config to be set for flac decoders, so add the
API to program the flac config to the DSP

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 55 ++++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h | 15 ++++++++++
 2 files changed, 70 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a33a55e..36e0eab13a98 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -38,6 +38,7 @@
 #define ASM_SESSION_CMD_RUN_V2			0x00010DAA
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
+#define ASM_MEDIA_FMT_FLAC			0x00010C16
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
@@ -89,6 +90,20 @@ struct asm_multi_channel_pcm_fmt_blk_v2 {
 	u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
 } __packed;
 
+struct asm_flac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 is_stream_info_present;
+	u16 num_channels;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 md5_sum[8];
+	u32 sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+	u16 sample_size;
+	u16 reserved;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -876,6 +891,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case FORMAT_LINEAR_PCM:
 		open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
 		break;
+	case SND_AUDIOCODEC_FLAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1021,6 +1039,42 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
 
+
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 struct q6asm_flac_cfg *cfg)
+{
+	struct asm_flac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->is_stream_info_present = cfg->stream_info_present;
+	fmt->num_channels = cfg->ch_cfg;
+	fmt->min_blk_size = cfg->min_blk_size;
+	fmt->max_blk_size = cfg->max_blk_size;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->min_frame_size = cfg->min_frame_size;
+	fmt->max_frame_size = cfg->max_frame_size;
+	fmt->sample_size = cfg->sample_size;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
@@ -1075,6 +1129,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
 
+
 /**
  * q6asm_read() - read data of period size from audio client
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 9f5fb573e4a0..6764f55f7078 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -32,6 +32,19 @@ enum {
 #define NO_TIMESTAMP    0xFF00
 #define FORMAT_LINEAR_PCM   0x0000
 
+struct q6asm_flac_cfg {
+        u32 sample_rate;
+        u32 ext_sample_rate;
+        u32 min_frame_size;
+        u32 max_frame_size;
+        u16 stream_info_present;
+        u16 min_blk_size;
+        u16 max_blk_size;
+        u16 ch_cfg;
+        u16 sample_size;
+        u16 md5_sum;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -54,6 +67,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 					  uint32_t rate, uint32_t channels,
 					  u8 channel_map[PCM_MAX_NUM_CHANNEL],
 					  uint16_t bits_per_sample);
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 struct q6asm_flac_cfg *cfg);
 int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
 	      uint32_t lsw_ts);
 int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
-- 
2.23.0

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[alsa-devel] Applied "ASoC: qcom: q6asm: add support to flac config" to the asoc tree

[Mark Brown]

The patch

   ASoC: qcom: q6asm: add support to flac config

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From badfe2c0666b1b65ad443aca74540bf6d976ec83 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 15 Nov 2019 15:57:04 +0530
Subject: [PATCH] ASoC: qcom: q6asm: add support to flac config

Qualcomm DSPs expect flac config to be set for flac decoders, so add the
API to program the flac config to the DSP

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-3-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 55 ++++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h | 15 ++++++++++
 2 files changed, 70 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a33a55e..36e0eab13a98 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -38,6 +38,7 @@
 #define ASM_SESSION_CMD_RUN_V2			0x00010DAA
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
+#define ASM_MEDIA_FMT_FLAC			0x00010C16
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
@@ -89,6 +90,20 @@ struct asm_multi_channel_pcm_fmt_blk_v2 {
 	u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
 } __packed;
 
+struct asm_flac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 is_stream_info_present;
+	u16 num_channels;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 md5_sum[8];
+	u32 sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+	u16 sample_size;
+	u16 reserved;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -876,6 +891,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case FORMAT_LINEAR_PCM:
 		open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
 		break;
+	case SND_AUDIOCODEC_FLAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1021,6 +1039,42 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
 
+
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 struct q6asm_flac_cfg *cfg)
+{
+	struct asm_flac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->is_stream_info_present = cfg->stream_info_present;
+	fmt->num_channels = cfg->ch_cfg;
+	fmt->min_blk_size = cfg->min_blk_size;
+	fmt->max_blk_size = cfg->max_blk_size;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->min_frame_size = cfg->min_frame_size;
+	fmt->max_frame_size = cfg->max_frame_size;
+	fmt->sample_size = cfg->sample_size;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
@@ -1075,6 +1129,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
 
+
 /**
  * q6asm_read() - read data of period size from audio client
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 9f5fb573e4a0..6764f55f7078 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -32,6 +32,19 @@ enum {
 #define NO_TIMESTAMP    0xFF00
 #define FORMAT_LINEAR_PCM   0x0000
 
+struct q6asm_flac_cfg {
+        u32 sample_rate;
+        u32 ext_sample_rate;
+        u32 min_frame_size;
+        u32 max_frame_size;
+        u16 stream_info_present;
+        u16 min_blk_size;
+        u16 max_blk_size;
+        u16 ch_cfg;
+        u16 sample_size;
+        u16 md5_sum;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -54,6 +67,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 					  uint32_t rate, uint32_t channels,
 					  u8 channel_map[PCM_MAX_NUM_CHANNEL],
 					  uint16_t bits_per_sample);
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 struct q6asm_flac_cfg *cfg);
 int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
 	      uint32_t lsw_ts);
 int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
-- 
2.20.1

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[alsa-devel] [RFC PATCH 3/3] ASoC: qcom: q6asm-dai: add support to flac decoder

[Vinod Koul]

Qualcomm DSPs also support the flac decoder, so add support for FLAC
decoder and convert the snd_dec_flac params to qdsp format.

Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 35 +++++++++++++++++++++++++++++++-
 1 file changed, 34 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 548eb4fa2da6..56e306bdbbe1 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -635,8 +635,14 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	int dir = stream->direction;
 	struct q6asm_dai_data *pdata;
+	struct q6asm_flac_cfg flac_cfg;
 	struct device *dev = c->dev;
 	int ret;
+	union snd_codec_options *codec_options;
+	struct snd_dec_flac *flac;
+
+	codec_options = &(prtd->codec_param.codec.options);
+
 
 	memcpy(&prtd->codec_param, params, sizeof(*params));
 
@@ -673,6 +679,32 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 		return ret;
 	}
 
+	switch (params->codec.id) {
+	case SND_AUDIOCODEC_FLAC:
+
+		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
+		flac = &codec_options->flac_d;
+
+		flac_cfg.ch_cfg = params->codec.ch_in;
+		flac_cfg.sample_rate =  params->codec.sample_rate;
+		flac_cfg.stream_info_present = 1;
+		flac_cfg.sample_size = flac->sample_size;
+		flac_cfg.min_blk_size = flac->min_blk_size;
+		flac_cfg.max_blk_size = flac->max_blk_size;
+		flac_cfg.max_frame_size = flac->max_frame_size;
+		flac_cfg.min_frame_size = flac->min_frame_size;
+
+		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+							   &flac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+	default:
+		break;
+	}
+
 	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
 				       (prtd->pcm_size / prtd->periods),
 				       prtd->periods);
@@ -768,8 +800,9 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream,
 	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 = 1;
+	caps->num_codecs = 2;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
+	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
 
 	return 0;
 }
-- 
2.23.0

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[alsa-devel] Applied "ASoC: qcom: q6asm-dai: add support to flac decoder" to the asoc tree

[Mark Brown]

The patch

   ASoC: qcom: q6asm-dai: add support to flac decoder

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

From 69efab0fc638704e100c3dfb29eac971a4c1cc29 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Fri, 15 Nov 2019 15:57:05 +0530
Subject: [PATCH] ASoC: qcom: q6asm-dai: add support to flac decoder

Qualcomm DSPs also support the flac decoder, so add support for FLAC
decoder and convert the snd_dec_flac params to qdsp format.

Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-4-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 35 +++++++++++++++++++++++++++++++-
 1 file changed, 34 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index f59353f510b8..8150c10f081e 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -626,8 +626,14 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	int dir = stream->direction;
 	struct q6asm_dai_data *pdata;
+	struct q6asm_flac_cfg flac_cfg;
 	struct device *dev = c->dev;
 	int ret;
+	union snd_codec_options *codec_options;
+	struct snd_dec_flac *flac;
+
+	codec_options = &(prtd->codec_param.codec.options);
+
 
 	memcpy(&prtd->codec_param, params, sizeof(*params));
 
@@ -664,6 +670,32 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 		return ret;
 	}
 
+	switch (params->codec.id) {
+	case SND_AUDIOCODEC_FLAC:
+
+		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
+		flac = &codec_options->flac_d;
+
+		flac_cfg.ch_cfg = params->codec.ch_in;
+		flac_cfg.sample_rate =  params->codec.sample_rate;
+		flac_cfg.stream_info_present = 1;
+		flac_cfg.sample_size = flac->sample_size;
+		flac_cfg.min_blk_size = flac->min_blk_size;
+		flac_cfg.max_blk_size = flac->max_blk_size;
+		flac_cfg.max_frame_size = flac->max_frame_size;
+		flac_cfg.min_frame_size = flac->min_frame_size;
+
+		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+							   &flac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+	default:
+		break;
+	}
+
 	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
 				       (prtd->pcm_size / prtd->periods),
 				       prtd->periods);
@@ -759,8 +791,9 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream,
 	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 = 1;
+	caps->num_codecs = 2;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
+	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
 
 	return 0;
 }
-- 
2.20.1

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Re: [alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[Takashi Iwai]

On Fri, 15 Nov 2019 11:27:02 +0100,
Vinod Koul wrote:
> 
> The current design of sending codec parameters assumes that decoders
> will have parsers so they can parse the encoded stream for parameters
> and configure the decoder.
> 
> But this assumption may not be universally true and we know some DSPs
> which do not contain the parsers so additional parameters are required
> to be passed.
> 
> So add these parameters starting with FLAC decoder. The size of
> snd_codec_options is still 120 bytes after this change (due to this
> being a union)
> 
> I think we should also bump the (minor) version if this proposal is
> acceptable so the userspace can check and populate flac specific structure.
> 
> Along, with the core header change, patches are added to support FLAC
> in Qualcomm drivers. This was tested on 96boards db845c
> 
> Srinivas Kandagatla (1):
>   ASoC: qcom: q6asm: add support to flac config
> 
> Vinod Koul (2):
>   ALSA: compress: add flac decoder params
>   ASoC: qcom: q6asm-dai: add support to flac decoder

Feel free to take my ACK for ALSA core part:
  Acked-by: Takashi Iwai <tiwai@suse.de>


thanks,

Takashi
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Re: [alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[Vinod Koul]

On 15-11-19, 14:21, Takashi Iwai wrote:
> On Fri, 15 Nov 2019 11:27:02 +0100,
> Vinod Koul wrote:
> > 
> > The current design of sending codec parameters assumes that decoders
> > will have parsers so they can parse the encoded stream for parameters
> > and configure the decoder.
> > 
> > But this assumption may not be universally true and we know some DSPs
> > which do not contain the parsers so additional parameters are required
> > to be passed.
> > 
> > So add these parameters starting with FLAC decoder. The size of
> > snd_codec_options is still 120 bytes after this change (due to this
> > being a union)
> > 
> > I think we should also bump the (minor) version if this proposal is
> > acceptable so the userspace can check and populate flac specific structure.
> > 
> > Along, with the core header change, patches are added to support FLAC
> > in Qualcomm drivers. This was tested on 96boards db845c
> > 
> > Srinivas Kandagatla (1):
> >   ASoC: qcom: q6asm: add support to flac config
> > 
> > Vinod Koul (2):
> >   ALSA: compress: add flac decoder params
> >   ASoC: qcom: q6asm-dai: add support to flac decoder
> 
> Feel free to take my ACK for ALSA core part:
>   Acked-by: Takashi Iwai <tiwai@suse.de>

Thanks Takashi, should we bump the version for the header to check for.
Btw I plan to add other decoders required as well. I have mp3 working
without any additional params but rest need additional info

Thanks
-- 
~Vinod
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Re: [alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[Pierre-Louis Bossart]

On 11/15/19 4:27 AM, Vinod Koul wrote:
> The current design of sending codec parameters assumes that decoders
> will have parsers so they can parse the encoded stream for parameters
> and configure the decoder.

that's not quite correct. It's rather than there was no need so far for 
existing implementations to have parameters on decode, this was never a 
limitation of the design, see e.g. the comments below:

/* AAC modes are required for encoders and decoders */

/*
  * IEC modes are mandatory for decoders. Format autodetection
  * will only happen on the DSP side with mode 0. The PCM mode should
  * not be used, the PCM codec should be used instead.
  */
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Re: [alsa-devel] [RFC PATCH 0/3] ALSA: compress: Add support for FLAC

[Vinod Koul]

On 15-11-19, 08:55, Pierre-Louis Bossart wrote:
> 
> 
> On 11/15/19 4:27 AM, Vinod Koul wrote:
> > The current design of sending codec parameters assumes that decoders
> > will have parsers so they can parse the encoded stream for parameters
> > and configure the decoder.
> 
> that's not quite correct. It's rather than there was no need so far for
> existing implementations to have parameters on decode, this was never a
> limitation of the design, see e.g. the comments below:

You might be correct here as this is implementation based and in this
case looks like decoders also need the additional params

-- 
~Vinod
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