[RESEND PATCH v2 0/9] ALSA: compress: Add wma, alac and ape support

[RESEND PATCH v2 0/9] ALSA: compress: Add wma, alac and ape support

[Vinod Koul]

This series adds more WMA profiles and WMA decoder parameters to UAPI and
then support for these in qcom driver. It also adds FLAC and APE IDs and
decoder parameters to UAPI and then support in qcom driver

This was tested on Dragon board RB3.

Last, bump up the compressed version so that userspace can check for the
support.

Since the series touches compress uapi and asoc, it would make sense to go
thru asoc tree with acks.

Changes in v2:
 - use bitflags for wma profiles

Vinod Koul (9):
  ALSA: compress: add wma codec profiles
  ALSA: compress: Add wma decoder params
  ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
  ASoC: qcom: q6asm: add support to wma config
  ASoC: qcom: q6asm-dai: add support to wma decoder
  ALSA: compress: add alac & ape decoder params
  ASoC: qcom: q6asm: add support for alac and ape configs
  ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders
  ALSA: compress: bump the version

 include/uapi/sound/compress_offload.h |   2 +-
 include/uapi/sound/compress_params.h  |  37 +++-
 sound/soc/qcom/qdsp6/q6asm-dai.c      | 136 +++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.c          | 243 +++++++++++++++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.h          |  51 +++++-
 5 files changed, 462 insertions(+), 7 deletions(-)

-- 
2.24.1

[RESEND PATCH v2 0/9] ALSA: compress: Add wma, alac and ape support

[Vinod Koul]

This series adds more WMA profiles and WMA decoder parameters to UAPI and
then support for these in qcom driver. It also adds FLAC and APE IDs and
decoder parameters to UAPI and then support in qcom driver

This was tested on Dragon board RB3.

Last, bump up the compressed version so that userspace can check for the
support.

Since the series touches compress uapi and asoc, it would make sense to go
thru asoc tree with acks.

Changes in v2:
 - use bitflags for wma profiles

Vinod Koul (9):
  ALSA: compress: add wma codec profiles
  ALSA: compress: Add wma decoder params
  ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
  ASoC: qcom: q6asm: add support to wma config
  ASoC: qcom: q6asm-dai: add support to wma decoder
  ALSA: compress: add alac & ape decoder params
  ASoC: qcom: q6asm: add support for alac and ape configs
  ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders
  ALSA: compress: bump the version

 include/uapi/sound/compress_offload.h |   2 +-
 include/uapi/sound/compress_params.h  |  37 +++-
 sound/soc/qcom/qdsp6/q6asm-dai.c      | 136 +++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.c          | 243 +++++++++++++++++++++++++-
 sound/soc/qcom/qdsp6/q6asm.h          |  51 +++++-
 5 files changed, 462 insertions(+), 7 deletions(-)

-- 
2.24.1

[RESEND PATCH v2 1/9] ALSA: compress: add wma codec profiles

[Vinod Koul]

Some codec profiles were missing for WMA, like WMA9/10 lossless and
wma10 pro, so add these profiles

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 9c96fb0e4d90..a47d9df0fd7b 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -142,6 +142,9 @@
 #define SND_AUDIOPROFILE_WMA8                ((__u32) 0x00000002)
 #define SND_AUDIOPROFILE_WMA9                ((__u32) 0x00000004)
 #define SND_AUDIOPROFILE_WMA10               ((__u32) 0x00000008)
+#define SND_AUDIOPROFILE_WMA9_PRO            ((__u32) 0x00000010)
+#define SND_AUDIOPROFILE_WMA9_LOSSLESS       ((__u32) 0x00000020)
+#define SND_AUDIOPROFILE_WMA10_LOSSLESS      ((__u32) 0x00000040)
 
 #define SND_AUDIOMODE_WMA_LEVEL1             ((__u32) 0x00000001)
 #define SND_AUDIOMODE_WMA_LEVEL2             ((__u32) 0x00000002)
-- 
2.24.1

[RESEND PATCH v2 1/9] ALSA: compress: add wma codec profiles

[Vinod Koul]

Some codec profiles were missing for WMA, like WMA9/10 lossless and
wma10 pro, so add these profiles

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 9c96fb0e4d90..a47d9df0fd7b 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -142,6 +142,9 @@
 #define SND_AUDIOPROFILE_WMA8                ((__u32) 0x00000002)
 #define SND_AUDIOPROFILE_WMA9                ((__u32) 0x00000004)
 #define SND_AUDIOPROFILE_WMA10               ((__u32) 0x00000008)
+#define SND_AUDIOPROFILE_WMA9_PRO            ((__u32) 0x00000010)
+#define SND_AUDIOPROFILE_WMA9_LOSSLESS       ((__u32) 0x00000020)
+#define SND_AUDIOPROFILE_WMA10_LOSSLESS      ((__u32) 0x00000040)
 
 #define SND_AUDIOMODE_WMA_LEVEL1             ((__u32) 0x00000001)
 #define SND_AUDIOMODE_WMA_LEVEL2             ((__u32) 0x00000002)
-- 
2.24.1

[RESEND PATCH v2 2/9] ALSA: compress: Add wma decoder params

[Vinod Koul]

Some WMA decoders like WMAv10 etc need some additional encoder option
parameters, so add these as WMA decoder params.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index a47d9df0fd7b..bf6f7155e775 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -329,6 +329,13 @@ struct snd_dec_flac {
 	__u16 reserved;
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_wma {
+	__u32 encoder_option;
+	__u32 adv_encoder_option;
+	__u32 adv_encoder_option2;
+	__u32 reserved;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -336,6 +343,7 @@ union snd_codec_options {
 	struct snd_enc_flac flac;
 	struct snd_enc_generic generic;
 	struct snd_dec_flac flac_d;
+	struct snd_dec_wma wma_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.24.1

[RESEND PATCH v2 2/9] ALSA: compress: Add wma decoder params

[Vinod Koul]

Some WMA decoders like WMAv10 etc need some additional encoder option
parameters, so add these as WMA decoder params.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index a47d9df0fd7b..bf6f7155e775 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -329,6 +329,13 @@ struct snd_dec_flac {
 	__u16 reserved;
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_wma {
+	__u32 encoder_option;
+	__u32 adv_encoder_option;
+	__u32 adv_encoder_option2;
+	__u32 reserved;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -336,6 +343,7 @@ union snd_codec_options {
 	struct snd_enc_flac flac;
 	struct snd_enc_generic generic;
 	struct snd_dec_flac flac_d;
+	struct snd_dec_wma wma_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.24.1

[RESEND PATCH v2 3/9] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write

[Vinod Koul]

Codec profile is required to be passed for WMA codecs so that we know
the codec profile present and tell DSP accordingly, so update this API
to pass the codec profile as argument

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
 sound/soc/qcom/qdsp6/q6asm.c     | 2 +-
 sound/soc/qcom/qdsp6/q6asm.h     | 2 +-
 3 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index c0d422d0ab94..8f245d03b6f5 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -250,7 +250,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
-				       prtd->bits_per_sample);
+				       0, prtd->bits_per_sample);
 	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
 				       prtd->bits_per_sample);
@@ -652,7 +652,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	prtd->bits_per_sample = 16;
 	if (dir == SND_COMPRESS_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, params->codec.id,
-					prtd->bits_per_sample);
+				params->codec.profile, prtd->bits_per_sample);
 
 		if (ret < 0) {
 			dev_err(dev, "q6asm_open_write failed\n");
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..64eb7b6ba305 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
  * Return: Will be an negative value on error or zero on success
  */
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample)
+		     u32 codec_profile, uint16_t bits_per_sample)
 {
 	struct asm_stream_cmd_open_write_v3 *open;
 	struct apr_pkt *pkt;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6764f55f7078..1cff7f68b95d 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
 int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 		       uint32_t lsw_ts, uint32_t flags);
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample);
+		     u32 codec_profile, uint16_t bits_per_sample);
 
 int q6asm_open_read(struct audio_client *ac, uint32_t format,
 		     uint16_t bits_per_sample);
-- 
2.24.1

[RESEND PATCH v2 3/9] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write

[Vinod Koul]

Codec profile is required to be passed for WMA codecs so that we know
the codec profile present and tell DSP accordingly, so update this API
to pass the codec profile as argument

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
 sound/soc/qcom/qdsp6/q6asm.c     | 2 +-
 sound/soc/qcom/qdsp6/q6asm.h     | 2 +-
 3 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index c0d422d0ab94..8f245d03b6f5 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -250,7 +250,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
-				       prtd->bits_per_sample);
+				       0, prtd->bits_per_sample);
 	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
 				       prtd->bits_per_sample);
@@ -652,7 +652,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	prtd->bits_per_sample = 16;
 	if (dir == SND_COMPRESS_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, params->codec.id,
-					prtd->bits_per_sample);
+				params->codec.profile, prtd->bits_per_sample);
 
 		if (ret < 0) {
 			dev_err(dev, "q6asm_open_write failed\n");
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..64eb7b6ba305 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
  * Return: Will be an negative value on error or zero on success
  */
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample)
+		     u32 codec_profile, uint16_t bits_per_sample)
 {
 	struct asm_stream_cmd_open_write_v3 *open;
 	struct apr_pkt *pkt;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6764f55f7078..1cff7f68b95d 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
 int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 		       uint32_t lsw_ts, uint32_t flags);
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample);
+		     u32 codec_profile, uint16_t bits_per_sample);
 
 int q6asm_open_read(struct audio_client *ac, uint32_t format,
 		     uint16_t bits_per_sample);
-- 
2.24.1

Re: [RESEND PATCH v2 3/9] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Codec profile is required to be passed for WMA codecs so that we know
> the codec profile present and tell DSP accordingly, so update this API
> to pass the codec profile as argument
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

LGTM,

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
>   sound/soc/qcom/qdsp6/q6asm.c     | 2 +-
>   sound/soc/qcom/qdsp6/q6asm.h     | 2 +-
>   3 files changed, 4 insertions(+), 4 deletions(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index c0d422d0ab94..8f245d03b6f5 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -250,7 +250,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
>   
>   	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
>   		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
> -				       prtd->bits_per_sample);
> +				       0, prtd->bits_per_sample);
>   	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
>   		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
>   				       prtd->bits_per_sample);
> @@ -652,7 +652,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	prtd->bits_per_sample = 16;
>   	if (dir == SND_COMPRESS_PLAYBACK) {
>   		ret = q6asm_open_write(prtd->audio_client, params->codec.id,
> -					prtd->bits_per_sample);
> +				params->codec.profile, prtd->bits_per_sample);
>   
>   		if (ret < 0) {
>   			dev_err(dev, "q6asm_open_write failed\n");
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 36e0eab13a98..64eb7b6ba305 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
>    * Return: Will be an negative value on error or zero on success
>    */
>   int q6asm_open_write(struct audio_client *ac, uint32_t format,
> -		     uint16_t bits_per_sample)
> +		     u32 codec_profile, uint16_t bits_per_sample)
>   {
>   	struct asm_stream_cmd_open_write_v3 *open;
>   	struct apr_pkt *pkt;
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 6764f55f7078..1cff7f68b95d 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
>   int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
>   		       uint32_t lsw_ts, uint32_t flags);
>   int q6asm_open_write(struct audio_client *ac, uint32_t format,
> -		     uint16_t bits_per_sample);
> +		     u32 codec_profile, uint16_t bits_per_sample);
>   
>   int q6asm_open_read(struct audio_client *ac, uint32_t format,
>   		     uint16_t bits_per_sample);
> 

Re: [RESEND PATCH v2 3/9] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Codec profile is required to be passed for WMA codecs so that we know
> the codec profile present and tell DSP accordingly, so update this API
> to pass the codec profile as argument
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

LGTM,

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
>   sound/soc/qcom/qdsp6/q6asm.c     | 2 +-
>   sound/soc/qcom/qdsp6/q6asm.h     | 2 +-
>   3 files changed, 4 insertions(+), 4 deletions(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index c0d422d0ab94..8f245d03b6f5 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -250,7 +250,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
>   
>   	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
>   		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
> -				       prtd->bits_per_sample);
> +				       0, prtd->bits_per_sample);
>   	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
>   		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
>   				       prtd->bits_per_sample);
> @@ -652,7 +652,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	prtd->bits_per_sample = 16;
>   	if (dir == SND_COMPRESS_PLAYBACK) {
>   		ret = q6asm_open_write(prtd->audio_client, params->codec.id,
> -					prtd->bits_per_sample);
> +				params->codec.profile, prtd->bits_per_sample);
>   
>   		if (ret < 0) {
>   			dev_err(dev, "q6asm_open_write failed\n");
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 36e0eab13a98..64eb7b6ba305 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
>    * Return: Will be an negative value on error or zero on success
>    */
>   int q6asm_open_write(struct audio_client *ac, uint32_t format,
> -		     uint16_t bits_per_sample)
> +		     u32 codec_profile, uint16_t bits_per_sample)
>   {
>   	struct asm_stream_cmd_open_write_v3 *open;
>   	struct apr_pkt *pkt;
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 6764f55f7078..1cff7f68b95d 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
>   int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
>   		       uint32_t lsw_ts, uint32_t flags);
>   int q6asm_open_write(struct audio_client *ac, uint32_t format,
> -		     uint16_t bits_per_sample);
> +		     u32 codec_profile, uint16_t bits_per_sample);
>   
>   int q6asm_open_read(struct audio_client *ac, uint32_t format,
>   		     uint16_t bits_per_sample);
> 

[RESEND PATCH v2 4/9] ASoC: qcom: q6asm: add support to wma config

[Vinod Koul]

Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma
decoders, so add the API to program the respective wma config to the DSP

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h |  17 +++++
 2 files changed, 140 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 64eb7b6ba305..4cec95c657ba 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -39,6 +39,8 @@
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
 #define ASM_MEDIA_FMT_FLAC			0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
@@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 {
 	u16 reserved;
 } __packed;
 
+struct asm_wmastdv9_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          advanced_enc_options1;
+	u32          advanced_enc_options2;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case SND_AUDIOCODEC_FLAC:
 		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
 		break;
+	case SND_AUDIOCODEC_WMA:
+		switch (codec_profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+			break;
+		case SND_AUDIOPROFILE_WMA10:
+		case SND_AUDIOPROFILE_WMA9_PRO:
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+			break;
+		default:
+			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+				codec_profile);
+			rc = -EINVAL;
+			goto err;
+		}
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 	return rc;
 }
 EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmastdv9_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->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->reserved = 0;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmaprov10_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->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->advanced_enc_options1 = cfg->adv_enc_options;
+	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 1cff7f68b95d..5d9fbc75688c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -45,6 +45,19 @@ struct q6asm_flac_cfg {
         u16 md5_sum;
 };
 
+struct q6asm_wma_cfg {
+	u32 fmtag;
+	u32 num_channels;
+	u32 sample_rate;
+	u32 bytes_per_sec;
+	u32 block_align;
+	u32 bits_per_sample;
+	u32 channel_mask;
+	u32 enc_options;
+	u32 adv_enc_options;
+	u32 adv_enc_options2;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 					  uint16_t bits_per_sample);
 int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 					 struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_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,
-- 
2.24.1

[RESEND PATCH v2 4/9] ASoC: qcom: q6asm: add support to wma config

[Vinod Koul]

Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma
decoders, so add the API to program the respective wma config to the DSP

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h |  17 +++++
 2 files changed, 140 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 64eb7b6ba305..4cec95c657ba 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -39,6 +39,8 @@
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
 #define ASM_MEDIA_FMT_FLAC			0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
@@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 {
 	u16 reserved;
 } __packed;
 
+struct asm_wmastdv9_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          advanced_enc_options1;
+	u32          advanced_enc_options2;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case SND_AUDIOCODEC_FLAC:
 		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
 		break;
+	case SND_AUDIOCODEC_WMA:
+		switch (codec_profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+			break;
+		case SND_AUDIOPROFILE_WMA10:
+		case SND_AUDIOPROFILE_WMA9_PRO:
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+			break;
+		default:
+			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+				codec_profile);
+			rc = -EINVAL;
+			goto err;
+		}
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 	return rc;
 }
 EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmastdv9_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->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->reserved = 0;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmaprov10_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->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->advanced_enc_options1 = cfg->adv_enc_options;
+	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 1cff7f68b95d..5d9fbc75688c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -45,6 +45,19 @@ struct q6asm_flac_cfg {
         u16 md5_sum;
 };
 
+struct q6asm_wma_cfg {
+	u32 fmtag;
+	u32 num_channels;
+	u32 sample_rate;
+	u32 bytes_per_sec;
+	u32 block_align;
+	u32 bits_per_sample;
+	u32 channel_mask;
+	u32 enc_options;
+	u32 adv_enc_options;
+	u32 adv_enc_options2;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 					  uint16_t bits_per_sample);
 int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 					 struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_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,
-- 
2.24.1

Re: [RESEND PATCH v2 4/9] ASoC: qcom: q6asm: add support to wma config

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma
> decoders, so add the API to program the respective wma config to the DSP
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> ---
>   sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++
>   sound/soc/qcom/qdsp6/q6asm.h |  17 +++++
>   2 files changed, 140 insertions(+)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 64eb7b6ba305..4cec95c657ba 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -39,6 +39,8 @@
>   #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
>   #define ASM_MEDIA_FMT_MP3			0x00010BE9
>   #define ASM_MEDIA_FMT_FLAC			0x00010C16
> +#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
> +#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
>   #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
>   #define ASM_DATA_CMD_READ_V2			0x00010DAC
>   #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
> @@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 {
>   	u16 reserved;
>   } __packed;
>   
> +struct asm_wmastdv9_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16          fmtag;
> +	u16          num_channels;
> +	u32          sample_rate;
> +	u32          bytes_per_sec;
> +	u16          blk_align;
> +	u16          bits_per_sample;
> +	u32          channel_mask;
> +	u16          enc_options;
> +	u16          reserved;
> +} __packed;
> +
> +struct asm_wmaprov10_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16          fmtag;
> +	u16          num_channels;
> +	u32          sample_rate;
> +	u32          bytes_per_sec;
> +	u16          blk_align;
> +	u16          bits_per_sample;
> +	u32          channel_mask;
> +	u16          enc_options;
> +	u16          advanced_enc_options1;
> +	u32          advanced_enc_options2;
> +} __packed;
> +
>   struct asm_stream_cmd_set_encdec_param {
>   	u32                  param_id;
>   	u32                  param_size;
> @@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
>   	case SND_AUDIOCODEC_FLAC:
>   		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
>   		break;
> +	case SND_AUDIOCODEC_WMA:
> +		switch (codec_profile) {
> +		case SND_AUDIOPROFILE_WMA9:
> +			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
> +			break;
> +		case SND_AUDIOPROFILE_WMA10:
> +		case SND_AUDIOPROFILE_WMA9_PRO:
> +		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
> +		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
> +			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
> +			break;
> +		default:
> +			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
> +				codec_profile);
> +			rc = -EINVAL;
> +			goto err;
> +		}
> +		break;
>   	default:
>   		dev_err(ac->dev, "Invalid format 0x%x\n", format);
>   		rc = -EINVAL;
> @@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
>   	return rc;
>   }
>   EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
> +
> +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
> +					   struct q6asm_wma_cfg *cfg)
> +{
> +	struct asm_wmastdv9_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->fmtag = cfg->fmtag;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->bytes_per_sec = cfg->bytes_per_sec;
> +	fmt->blk_align = cfg->block_align;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->channel_mask = cfg->channel_mask;
> +	fmt->enc_options = cfg->enc_options;
> +	fmt->reserved = 0;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
> +
> +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
> +					    struct q6asm_wma_cfg *cfg)
> +{
> +	struct asm_wmaprov10_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->fmtag = cfg->fmtag;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->bytes_per_sec = cfg->bytes_per_sec;
> +	fmt->blk_align = cfg->block_align;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->channel_mask = cfg->channel_mask;
> +	fmt->enc_options = cfg->enc_options;
> +	fmt->advanced_enc_options1 = cfg->adv_enc_options;
> +	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
> +
>   /**
>    * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
>    *
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 1cff7f68b95d..5d9fbc75688c 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -45,6 +45,19 @@ struct q6asm_flac_cfg {
>           u16 md5_sum;
>   };
>   
> +struct q6asm_wma_cfg {
> +	u32 fmtag;
> +	u32 num_channels;
> +	u32 sample_rate;
> +	u32 bytes_per_sec;
> +	u32 block_align;
> +	u32 bits_per_sample;
> +	u32 channel_mask;
> +	u32 enc_options;
> +	u32 adv_enc_options;
> +	u32 adv_enc_options2;
> +};
> +
>   typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
>   			  void *payload, void *priv);
>   struct audio_client;
> @@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
>   					  uint16_t bits_per_sample);
>   int q6asm_stream_media_format_block_flac(struct audio_client *ac,
>   					 struct q6asm_flac_cfg *cfg);
> +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
> +					   struct q6asm_wma_cfg *cfg);
> +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
> +					    struct q6asm_wma_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,
> 

Re: [RESEND PATCH v2 4/9] ASoC: qcom: q6asm: add support to wma config

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma
> decoders, so add the API to program the respective wma config to the DSP
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> ---
>   sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++
>   sound/soc/qcom/qdsp6/q6asm.h |  17 +++++
>   2 files changed, 140 insertions(+)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 64eb7b6ba305..4cec95c657ba 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -39,6 +39,8 @@
>   #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
>   #define ASM_MEDIA_FMT_MP3			0x00010BE9
>   #define ASM_MEDIA_FMT_FLAC			0x00010C16
> +#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
> +#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
>   #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
>   #define ASM_DATA_CMD_READ_V2			0x00010DAC
>   #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
> @@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 {
>   	u16 reserved;
>   } __packed;
>   
> +struct asm_wmastdv9_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16          fmtag;
> +	u16          num_channels;
> +	u32          sample_rate;
> +	u32          bytes_per_sec;
> +	u16          blk_align;
> +	u16          bits_per_sample;
> +	u32          channel_mask;
> +	u16          enc_options;
> +	u16          reserved;
> +} __packed;
> +
> +struct asm_wmaprov10_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16          fmtag;
> +	u16          num_channels;
> +	u32          sample_rate;
> +	u32          bytes_per_sec;
> +	u16          blk_align;
> +	u16          bits_per_sample;
> +	u32          channel_mask;
> +	u16          enc_options;
> +	u16          advanced_enc_options1;
> +	u32          advanced_enc_options2;
> +} __packed;
> +
>   struct asm_stream_cmd_set_encdec_param {
>   	u32                  param_id;
>   	u32                  param_size;
> @@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
>   	case SND_AUDIOCODEC_FLAC:
>   		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
>   		break;
> +	case SND_AUDIOCODEC_WMA:
> +		switch (codec_profile) {
> +		case SND_AUDIOPROFILE_WMA9:
> +			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
> +			break;
> +		case SND_AUDIOPROFILE_WMA10:
> +		case SND_AUDIOPROFILE_WMA9_PRO:
> +		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
> +		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
> +			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
> +			break;
> +		default:
> +			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
> +				codec_profile);
> +			rc = -EINVAL;
> +			goto err;
> +		}
> +		break;
>   	default:
>   		dev_err(ac->dev, "Invalid format 0x%x\n", format);
>   		rc = -EINVAL;
> @@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
>   	return rc;
>   }
>   EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
> +
> +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
> +					   struct q6asm_wma_cfg *cfg)
> +{
> +	struct asm_wmastdv9_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->fmtag = cfg->fmtag;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->bytes_per_sec = cfg->bytes_per_sec;
> +	fmt->blk_align = cfg->block_align;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->channel_mask = cfg->channel_mask;
> +	fmt->enc_options = cfg->enc_options;
> +	fmt->reserved = 0;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
> +
> +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
> +					    struct q6asm_wma_cfg *cfg)
> +{
> +	struct asm_wmaprov10_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->fmtag = cfg->fmtag;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->bytes_per_sec = cfg->bytes_per_sec;
> +	fmt->blk_align = cfg->block_align;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->channel_mask = cfg->channel_mask;
> +	fmt->enc_options = cfg->enc_options;
> +	fmt->advanced_enc_options1 = cfg->adv_enc_options;
> +	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
> +
>   /**
>    * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
>    *
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 1cff7f68b95d..5d9fbc75688c 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -45,6 +45,19 @@ struct q6asm_flac_cfg {
>           u16 md5_sum;
>   };
>   
> +struct q6asm_wma_cfg {
> +	u32 fmtag;
> +	u32 num_channels;
> +	u32 sample_rate;
> +	u32 bytes_per_sec;
> +	u32 block_align;
> +	u32 bits_per_sample;
> +	u32 channel_mask;
> +	u32 enc_options;
> +	u32 adv_enc_options;
> +	u32 adv_enc_options2;
> +};
> +
>   typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
>   			  void *payload, void *priv);
>   struct audio_client;
> @@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
>   					  uint16_t bits_per_sample);
>   int q6asm_stream_media_format_block_flac(struct audio_client *ac,
>   					 struct q6asm_flac_cfg *cfg);
> +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
> +					   struct q6asm_wma_cfg *cfg);
> +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
> +					    struct q6asm_wma_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,
> 

[RESEND PATCH v2 5/9] ASoC: qcom: q6asm-dai: add support to wma decoder

[Vinod Koul]

Qualcomm DSPs also supports the wma decoder, so add support for wma
decoder and convert the snd_codec_params to qdsp format.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
 1 file changed, 66 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 8f245d03b6f5..53c250778eea 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -627,10 +627,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	int dir = stream->direction;
 	struct q6asm_dai_data *pdata;
 	struct q6asm_flac_cfg flac_cfg;
+	struct q6asm_wma_cfg wma_cfg;
+	unsigned int wma_v9 = 0;
 	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
+	struct snd_dec_wma *wma;
 
 	codec_options = &(prtd->codec_param.codec.options);
 
@@ -692,6 +695,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 			return -EIO;
 		}
 		break;
+
+	case SND_AUDIOCODEC_WMA:
+		wma = &codec_options->wma_d;
+
+		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+		wma_cfg.sample_rate =  params->codec.sample_rate;
+		wma_cfg.num_channels = params->codec.ch_in;
+		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+		wma_cfg.block_align = params->codec.align;
+		wma_cfg.bits_per_sample = prtd->bits_per_sample;
+		wma_cfg.enc_options = wma->encoder_option;
+		wma_cfg.adv_enc_options = wma->adv_encoder_option;
+		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+		if (wma_cfg.num_channels == 1)
+			wma_cfg.channel_mask = 4; /* Mono Center */
+		else if (wma_cfg.num_channels == 2)
+			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+		else
+			return -EINVAL;
+
+		/* check the codec profile */
+		switch (params->codec.profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			wma_cfg.fmtag = 0x161;
+			wma_v9 = 1;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10:
+			wma_cfg.fmtag = 0x166;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_PRO:
+			wma_cfg.fmtag = 0x162;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+			wma_cfg.fmtag = 0x163;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			wma_cfg.fmtag = 0x167;
+			break;
+
+		default:
+			dev_err(dev, "Unknown WMA profile:%x\n",
+				params->codec.profile);
+			return -EIO;
+		}
+
+		if (wma_v9)
+			ret = q6asm_stream_media_format_block_wma_v9(
+					prtd->audio_client, &wma_cfg);
+		else
+			ret = q6asm_stream_media_format_block_wma_v10(
+					prtd->audio_client, &wma_cfg);
+		if (ret < 0) {
+			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+			return -EIO;
+		}
 	default:
 		break;
 	}
@@ -791,9 +855,10 @@ 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 = 2;
+	caps->num_codecs = 3;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
 	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+	caps->codecs[2] = SND_AUDIOCODEC_WMA;
 
 	return 0;
 }
-- 
2.24.1

[RESEND PATCH v2 5/9] ASoC: qcom: q6asm-dai: add support to wma decoder

[Vinod Koul]

Qualcomm DSPs also supports the wma decoder, so add support for wma
decoder and convert the snd_codec_params to qdsp format.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
 1 file changed, 66 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 8f245d03b6f5..53c250778eea 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -627,10 +627,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	int dir = stream->direction;
 	struct q6asm_dai_data *pdata;
 	struct q6asm_flac_cfg flac_cfg;
+	struct q6asm_wma_cfg wma_cfg;
+	unsigned int wma_v9 = 0;
 	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
+	struct snd_dec_wma *wma;
 
 	codec_options = &(prtd->codec_param.codec.options);
 
@@ -692,6 +695,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 			return -EIO;
 		}
 		break;
+
+	case SND_AUDIOCODEC_WMA:
+		wma = &codec_options->wma_d;
+
+		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+		wma_cfg.sample_rate =  params->codec.sample_rate;
+		wma_cfg.num_channels = params->codec.ch_in;
+		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+		wma_cfg.block_align = params->codec.align;
+		wma_cfg.bits_per_sample = prtd->bits_per_sample;
+		wma_cfg.enc_options = wma->encoder_option;
+		wma_cfg.adv_enc_options = wma->adv_encoder_option;
+		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+		if (wma_cfg.num_channels == 1)
+			wma_cfg.channel_mask = 4; /* Mono Center */
+		else if (wma_cfg.num_channels == 2)
+			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+		else
+			return -EINVAL;
+
+		/* check the codec profile */
+		switch (params->codec.profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			wma_cfg.fmtag = 0x161;
+			wma_v9 = 1;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10:
+			wma_cfg.fmtag = 0x166;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_PRO:
+			wma_cfg.fmtag = 0x162;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+			wma_cfg.fmtag = 0x163;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			wma_cfg.fmtag = 0x167;
+			break;
+
+		default:
+			dev_err(dev, "Unknown WMA profile:%x\n",
+				params->codec.profile);
+			return -EIO;
+		}
+
+		if (wma_v9)
+			ret = q6asm_stream_media_format_block_wma_v9(
+					prtd->audio_client, &wma_cfg);
+		else
+			ret = q6asm_stream_media_format_block_wma_v10(
+					prtd->audio_client, &wma_cfg);
+		if (ret < 0) {
+			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+			return -EIO;
+		}
 	default:
 		break;
 	}
@@ -791,9 +855,10 @@ 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 = 2;
+	caps->num_codecs = 3;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
 	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+	caps->codecs[2] = SND_AUDIOCODEC_WMA;
 
 	return 0;
 }
-- 
2.24.1

Re: [RESEND PATCH v2 5/9] ASoC: qcom: q6asm-dai: add support to wma decoder

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs also supports the wma decoder, so add support for wma
> decoder and convert the snd_codec_params to qdsp format.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>


> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
>   1 file changed, 66 insertions(+), 1 deletion(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index 8f245d03b6f5..53c250778eea 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -627,10 +627,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	int dir = stream->direction;
>   	struct q6asm_dai_data *pdata;
>   	struct q6asm_flac_cfg flac_cfg;
> +	struct q6asm_wma_cfg wma_cfg;
> +	unsigned int wma_v9 = 0;
>   	struct device *dev = c->dev;
>   	int ret;
>   	union snd_codec_options *codec_options;
>   	struct snd_dec_flac *flac;
> +	struct snd_dec_wma *wma;
>   
>   	codec_options = &(prtd->codec_param.codec.options);
>   
> @@ -692,6 +695,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   			return -EIO;
>   		}
>   		break;
> +
> +	case SND_AUDIOCODEC_WMA:
> +		wma = &codec_options->wma_d;
> +
> +		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
> +
> +		wma_cfg.sample_rate =  params->codec.sample_rate;
> +		wma_cfg.num_channels = params->codec.ch_in;
> +		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
> +		wma_cfg.block_align = params->codec.align;
> +		wma_cfg.bits_per_sample = prtd->bits_per_sample;
> +		wma_cfg.enc_options = wma->encoder_option;
> +		wma_cfg.adv_enc_options = wma->adv_encoder_option;
> +		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
> +
> +		if (wma_cfg.num_channels == 1)
> +			wma_cfg.channel_mask = 4; /* Mono Center */
> +		else if (wma_cfg.num_channels == 2)
> +			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
> +		else
> +			return -EINVAL;
> +
> +		/* check the codec profile */
> +		switch (params->codec.profile) {
> +		case SND_AUDIOPROFILE_WMA9:
> +			wma_cfg.fmtag = 0x161;
> +			wma_v9 = 1;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA10:
> +			wma_cfg.fmtag = 0x166;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA9_PRO:
> +			wma_cfg.fmtag = 0x162;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
> +			wma_cfg.fmtag = 0x163;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
> +			wma_cfg.fmtag = 0x167;
> +			break;
> +
> +		default:
> +			dev_err(dev, "Unknown WMA profile:%x\n",
> +				params->codec.profile);
> +			return -EIO;
> +		}
> +
> +		if (wma_v9)
> +			ret = q6asm_stream_media_format_block_wma_v9(
> +					prtd->audio_client, &wma_cfg);
> +		else
> +			ret = q6asm_stream_media_format_block_wma_v10(
> +					prtd->audio_client, &wma_cfg);
> +		if (ret < 0) {
> +			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
> +			return -EIO;
> +		}
>   	default:
>   		break;
>   	}
> @@ -791,9 +855,10 @@ 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 = 2;
> +	caps->num_codecs = 3;
>   	caps->codecs[0] = SND_AUDIOCODEC_MP3;
>   	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
> +	caps->codecs[2] = SND_AUDIOCODEC_WMA;
>   
>   	return 0;
>   }
> 

Re: [RESEND PATCH v2 5/9] ASoC: qcom: q6asm-dai: add support to wma decoder

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs also supports the wma decoder, so add support for wma
> decoder and convert the snd_codec_params to qdsp format.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>


> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
>   1 file changed, 66 insertions(+), 1 deletion(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index 8f245d03b6f5..53c250778eea 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -627,10 +627,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	int dir = stream->direction;
>   	struct q6asm_dai_data *pdata;
>   	struct q6asm_flac_cfg flac_cfg;
> +	struct q6asm_wma_cfg wma_cfg;
> +	unsigned int wma_v9 = 0;
>   	struct device *dev = c->dev;
>   	int ret;
>   	union snd_codec_options *codec_options;
>   	struct snd_dec_flac *flac;
> +	struct snd_dec_wma *wma;
>   
>   	codec_options = &(prtd->codec_param.codec.options);
>   
> @@ -692,6 +695,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   			return -EIO;
>   		}
>   		break;
> +
> +	case SND_AUDIOCODEC_WMA:
> +		wma = &codec_options->wma_d;
> +
> +		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
> +
> +		wma_cfg.sample_rate =  params->codec.sample_rate;
> +		wma_cfg.num_channels = params->codec.ch_in;
> +		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
> +		wma_cfg.block_align = params->codec.align;
> +		wma_cfg.bits_per_sample = prtd->bits_per_sample;
> +		wma_cfg.enc_options = wma->encoder_option;
> +		wma_cfg.adv_enc_options = wma->adv_encoder_option;
> +		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
> +
> +		if (wma_cfg.num_channels == 1)
> +			wma_cfg.channel_mask = 4; /* Mono Center */
> +		else if (wma_cfg.num_channels == 2)
> +			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
> +		else
> +			return -EINVAL;
> +
> +		/* check the codec profile */
> +		switch (params->codec.profile) {
> +		case SND_AUDIOPROFILE_WMA9:
> +			wma_cfg.fmtag = 0x161;
> +			wma_v9 = 1;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA10:
> +			wma_cfg.fmtag = 0x166;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA9_PRO:
> +			wma_cfg.fmtag = 0x162;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
> +			wma_cfg.fmtag = 0x163;
> +			break;
> +
> +		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
> +			wma_cfg.fmtag = 0x167;
> +			break;
> +
> +		default:
> +			dev_err(dev, "Unknown WMA profile:%x\n",
> +				params->codec.profile);
> +			return -EIO;
> +		}
> +
> +		if (wma_v9)
> +			ret = q6asm_stream_media_format_block_wma_v9(
> +					prtd->audio_client, &wma_cfg);
> +		else
> +			ret = q6asm_stream_media_format_block_wma_v10(
> +					prtd->audio_client, &wma_cfg);
> +		if (ret < 0) {
> +			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
> +			return -EIO;
> +		}
>   	default:
>   		break;
>   	}
> @@ -791,9 +855,10 @@ 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 = 2;
> +	caps->num_codecs = 3;
>   	caps->codecs[0] = SND_AUDIOCODEC_MP3;
>   	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
> +	caps->codecs[2] = SND_AUDIOCODEC_WMA;
>   
>   	return 0;
>   }
> 

[RESEND PATCH v2 6/9] ALSA: compress: add alac & ape decoder params

[Vinod Koul]

Add ALAC (Apple Lossless Audio Codec) and APE (Monkey's Lossless Audio
Codec) defines and parameters required to configure these.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 26 +++++++++++++++++++++++++-
 1 file changed, 25 insertions(+), 1 deletion(-)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index bf6f7155e775..79b14389ae41 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -75,7 +75,9 @@
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
 #define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_BESPOKE
+#define SND_AUDIOCODEC_ALAC                  ((__u32) 0x0000000F)
+#define SND_AUDIOCODEC_APE                   ((__u32) 0x00000010)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_APE
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
@@ -336,6 +338,26 @@ struct snd_dec_wma {
 	__u32 reserved;
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_alac {
+	__u32 frame_length;
+	__u8 compatible_version;
+	__u8 pb;
+	__u8 mb;
+	__u8 kb;
+	__u32 max_run;
+	__u32 max_frame_bytes;
+} __attribute__((packed, aligned(4)));
+
+struct snd_dec_ape {
+	__u16 compatible_version;
+	__u16 compression_level;
+	__u32 format_flags;
+	__u32 blocks_per_frame;
+	__u32 final_frame_blocks;
+	__u32 total_frames;
+	__u32 seek_table_present;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -344,6 +366,8 @@ union snd_codec_options {
 	struct snd_enc_generic generic;
 	struct snd_dec_flac flac_d;
 	struct snd_dec_wma wma_d;
+	struct snd_dec_alac alac_d;
+	struct snd_dec_ape ape_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.24.1

[RESEND PATCH v2 6/9] ALSA: compress: add alac & ape decoder params

[Vinod Koul]

Add ALAC (Apple Lossless Audio Codec) and APE (Monkey's Lossless Audio
Codec) defines and parameters required to configure these.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_params.h | 26 +++++++++++++++++++++++++-
 1 file changed, 25 insertions(+), 1 deletion(-)

diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index bf6f7155e775..79b14389ae41 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -75,7 +75,9 @@
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
 #define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_BESPOKE
+#define SND_AUDIOCODEC_ALAC                  ((__u32) 0x0000000F)
+#define SND_AUDIOCODEC_APE                   ((__u32) 0x00000010)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_APE
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
@@ -336,6 +338,26 @@ struct snd_dec_wma {
 	__u32 reserved;
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_alac {
+	__u32 frame_length;
+	__u8 compatible_version;
+	__u8 pb;
+	__u8 mb;
+	__u8 kb;
+	__u32 max_run;
+	__u32 max_frame_bytes;
+} __attribute__((packed, aligned(4)));
+
+struct snd_dec_ape {
+	__u16 compatible_version;
+	__u16 compression_level;
+	__u32 format_flags;
+	__u32 blocks_per_frame;
+	__u32 final_frame_blocks;
+	__u32 total_frames;
+	__u32 seek_table_present;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -344,6 +366,8 @@ union snd_codec_options {
 	struct snd_enc_generic generic;
 	struct snd_dec_flac flac_d;
 	struct snd_dec_wma wma_d;
+	struct snd_dec_alac alac_d;
+	struct snd_dec_ape ape_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities
-- 
2.24.1

[RESEND PATCH v2 7/9] ASoC: qcom: q6asm: add support for alac and ape configs

[Vinod Koul]

Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
so add the API to program the respective config to the DSP.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h |  32 ++++++++++
 2 files changed, 150 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 4cec95c657ba..0e0e8f7a460a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -48,6 +48,8 @@
 #define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
 #define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+#define ASM_MEDIA_FMT_ALAC			0x00012f31
+#define ASM_MEDIA_FMT_APE			0x00012f32
 
 
 #define ASM_LEGACY_STREAM_SESSION	0
@@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 {
 	u32          advanced_enc_options2;
 } __packed;
 
+struct asm_alac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 			goto err;
 		}
 		break;
+	case SND_AUDIOCODEC_ALAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+		break;
+	case SND_AUDIOCODEC_APE:
+		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
 
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg)
+{
+	struct asm_alac_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->frame_length = cfg->frame_length;
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->bit_depth =  cfg->bit_depth;
+	fmt->num_channels = cfg->num_channels;
+	fmt->max_run = cfg->max_run;
+	fmt->max_frame_bytes = cfg->max_frame_bytes;
+	fmt->avg_bit_rate = cfg->avg_bit_rate;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->channel_layout_tag = cfg->channel_layout_tag;
+	fmt->pb = cfg->pb;
+	fmt->mb = cfg->mb;
+	fmt->kb = cfg->kb;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_cfg *cfg)
+{
+	struct asm_ape_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->compatible_version = cfg->compatible_version;
+	fmt->compression_level = cfg->compression_level;
+	fmt->format_flags = cfg->format_flags;
+	fmt->blocks_per_frame = cfg->blocks_per_frame;
+	fmt->final_frame_blocks = cfg->final_frame_blocks;
+	fmt->total_frames = cfg->total_frames;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->seek_table_present = cfg->seek_table_present;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 5d9fbc75688c..38a207d6cd95 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -58,6 +58,34 @@ struct q6asm_wma_cfg {
 	u32 adv_enc_options2;
 };
 
+struct q6asm_alac_cfg {
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
 					   struct q6asm_wma_cfg *cfg);
 int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 					    struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_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,
-- 
2.24.1

[RESEND PATCH v2 7/9] ASoC: qcom: q6asm: add support for alac and ape configs

[Vinod Koul]

Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
so add the API to program the respective config to the DSP.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6asm.h |  32 ++++++++++
 2 files changed, 150 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 4cec95c657ba..0e0e8f7a460a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -48,6 +48,8 @@
 #define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
 #define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+#define ASM_MEDIA_FMT_ALAC			0x00012f31
+#define ASM_MEDIA_FMT_APE			0x00012f32
 
 
 #define ASM_LEGACY_STREAM_SESSION	0
@@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 {
 	u32          advanced_enc_options2;
 } __packed;
 
+struct asm_alac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 			goto err;
 		}
 		break;
+	case SND_AUDIOCODEC_ALAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+		break;
+	case SND_AUDIOCODEC_APE:
+		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 }
 EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
 
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg)
+{
+	struct asm_alac_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->frame_length = cfg->frame_length;
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->bit_depth =  cfg->bit_depth;
+	fmt->num_channels = cfg->num_channels;
+	fmt->max_run = cfg->max_run;
+	fmt->max_frame_bytes = cfg->max_frame_bytes;
+	fmt->avg_bit_rate = cfg->avg_bit_rate;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->channel_layout_tag = cfg->channel_layout_tag;
+	fmt->pb = cfg->pb;
+	fmt->mb = cfg->mb;
+	fmt->kb = cfg->kb;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_cfg *cfg)
+{
+	struct asm_ape_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->compatible_version = cfg->compatible_version;
+	fmt->compression_level = cfg->compression_level;
+	fmt->format_flags = cfg->format_flags;
+	fmt->blocks_per_frame = cfg->blocks_per_frame;
+	fmt->final_frame_blocks = cfg->final_frame_blocks;
+	fmt->total_frames = cfg->total_frames;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->seek_table_present = cfg->seek_table_present;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 5d9fbc75688c..38a207d6cd95 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -58,6 +58,34 @@ struct q6asm_wma_cfg {
 	u32 adv_enc_options2;
 };
 
+struct q6asm_alac_cfg {
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
@@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
 					   struct q6asm_wma_cfg *cfg);
 int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 					    struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_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,
-- 
2.24.1

Re: [RESEND PATCH v2 7/9] ASoC: qcom: q6asm: add support for alac and ape configs

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
> so add the API to program the respective config to the DSP.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>


Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>


> ---
>   sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++
>   sound/soc/qcom/qdsp6/q6asm.h |  32 ++++++++++
>   2 files changed, 150 insertions(+)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 4cec95c657ba..0e0e8f7a460a 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -48,6 +48,8 @@
>   #define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
>   #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
>   #define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
> +#define ASM_MEDIA_FMT_ALAC			0x00012f31
> +#define ASM_MEDIA_FMT_APE			0x00012f32
>   
>   
>   #define ASM_LEGACY_STREAM_SESSION	0
> @@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 {
>   	u32          advanced_enc_options2;
>   } __packed;
>   
> +struct asm_alac_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u32 frame_length;
> +	u8 compatible_version;
> +	u8 bit_depth;
> +	u8 pb;
> +	u8 mb;
> +	u8 kb;
> +	u8 num_channels;
> +	u16 max_run;
> +	u32 max_frame_bytes;
> +	u32 avg_bit_rate;
> +	u32 sample_rate;
> +	u32 channel_layout_tag;
> +} __packed;
> +
> +struct asm_ape_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16 compatible_version;
> +	u16 compression_level;
> +	u32 format_flags;
> +	u32 blocks_per_frame;
> +	u32 final_frame_blocks;
> +	u32 total_frames;
> +	u16 bits_per_sample;
> +	u16 num_channels;
> +	u32 sample_rate;
> +	u32 seek_table_present;
> +} __packed;
> +
>   struct asm_stream_cmd_set_encdec_param {
>   	u32                  param_id;
>   	u32                  param_size;
> @@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
>   			goto err;
>   		}
>   		break;
> +	case SND_AUDIOCODEC_ALAC:
> +		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
> +		break;
> +	case SND_AUDIOCODEC_APE:
> +		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
> +		break;
>   	default:
>   		dev_err(ac->dev, "Invalid format 0x%x\n", format);
>   		rc = -EINVAL;
> @@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
>   }
>   EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
>   
> +int q6asm_stream_media_format_block_alac(struct audio_client *ac,
> +					 struct q6asm_alac_cfg *cfg)
> +{
> +	struct asm_alac_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->frame_length = cfg->frame_length;
> +	fmt->compatible_version = cfg->compatible_version;
> +	fmt->bit_depth =  cfg->bit_depth;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->max_run = cfg->max_run;
> +	fmt->max_frame_bytes = cfg->max_frame_bytes;
> +	fmt->avg_bit_rate = cfg->avg_bit_rate;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->channel_layout_tag = cfg->channel_layout_tag;
> +	fmt->pb = cfg->pb;
> +	fmt->mb = cfg->mb;
> +	fmt->kb = cfg->kb;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
> +
> +int q6asm_stream_media_format_block_ape(struct audio_client *ac,
> +					struct q6asm_ape_cfg *cfg)
> +{
> +	struct asm_ape_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->compatible_version = cfg->compatible_version;
> +	fmt->compression_level = cfg->compression_level;
> +	fmt->format_flags = cfg->format_flags;
> +	fmt->blocks_per_frame = cfg->blocks_per_frame;
> +	fmt->final_frame_blocks = cfg->final_frame_blocks;
> +	fmt->total_frames = cfg->total_frames;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->seek_table_present = cfg->seek_table_present;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
> +
>   /**
>    * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
>    *
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 5d9fbc75688c..38a207d6cd95 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -58,6 +58,34 @@ struct q6asm_wma_cfg {
>   	u32 adv_enc_options2;
>   };
>   
> +struct q6asm_alac_cfg {
> +	u32 frame_length;
> +	u8 compatible_version;
> +	u8 bit_depth;
> +	u8 pb;
> +	u8 mb;
> +	u8 kb;
> +	u8 num_channels;
> +	u16 max_run;
> +	u32 max_frame_bytes;
> +	u32 avg_bit_rate;
> +	u32 sample_rate;
> +	u32 channel_layout_tag;
> +};
> +
> +struct q6asm_ape_cfg {
> +	u16 compatible_version;
> +	u16 compression_level;
> +	u32 format_flags;
> +	u32 blocks_per_frame;
> +	u32 final_frame_blocks;
> +	u32 total_frames;
> +	u16 bits_per_sample;
> +	u16 num_channels;
> +	u32 sample_rate;
> +	u32 seek_table_present;
> +};
> +
>   typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
>   			  void *payload, void *priv);
>   struct audio_client;
> @@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
>   					   struct q6asm_wma_cfg *cfg);
>   int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
>   					    struct q6asm_wma_cfg *cfg);
> +int q6asm_stream_media_format_block_alac(struct audio_client *ac,
> +					 struct q6asm_alac_cfg *cfg);
> +int q6asm_stream_media_format_block_ape(struct audio_client *ac,
> +					struct q6asm_ape_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,
> 

Re: [RESEND PATCH v2 7/9] ASoC: qcom: q6asm: add support for alac and ape configs

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
> so add the API to program the respective config to the DSP.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>


Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>


> ---
>   sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++
>   sound/soc/qcom/qdsp6/q6asm.h |  32 ++++++++++
>   2 files changed, 150 insertions(+)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 4cec95c657ba..0e0e8f7a460a 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -48,6 +48,8 @@
>   #define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
>   #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
>   #define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
> +#define ASM_MEDIA_FMT_ALAC			0x00012f31
> +#define ASM_MEDIA_FMT_APE			0x00012f32
>   
>   
>   #define ASM_LEGACY_STREAM_SESSION	0
> @@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 {
>   	u32          advanced_enc_options2;
>   } __packed;
>   
> +struct asm_alac_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u32 frame_length;
> +	u8 compatible_version;
> +	u8 bit_depth;
> +	u8 pb;
> +	u8 mb;
> +	u8 kb;
> +	u8 num_channels;
> +	u16 max_run;
> +	u32 max_frame_bytes;
> +	u32 avg_bit_rate;
> +	u32 sample_rate;
> +	u32 channel_layout_tag;
> +} __packed;
> +
> +struct asm_ape_fmt_blk_v2 {
> +	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
> +	u16 compatible_version;
> +	u16 compression_level;
> +	u32 format_flags;
> +	u32 blocks_per_frame;
> +	u32 final_frame_blocks;
> +	u32 total_frames;
> +	u16 bits_per_sample;
> +	u16 num_channels;
> +	u32 sample_rate;
> +	u32 seek_table_present;
> +} __packed;
> +
>   struct asm_stream_cmd_set_encdec_param {
>   	u32                  param_id;
>   	u32                  param_size;
> @@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
>   			goto err;
>   		}
>   		break;
> +	case SND_AUDIOCODEC_ALAC:
> +		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
> +		break;
> +	case SND_AUDIOCODEC_APE:
> +		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
> +		break;
>   	default:
>   		dev_err(ac->dev, "Invalid format 0x%x\n", format);
>   		rc = -EINVAL;
> @@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
>   }
>   EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
>   
> +int q6asm_stream_media_format_block_alac(struct audio_client *ac,
> +					 struct q6asm_alac_cfg *cfg)
> +{
> +	struct asm_alac_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->frame_length = cfg->frame_length;
> +	fmt->compatible_version = cfg->compatible_version;
> +	fmt->bit_depth =  cfg->bit_depth;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->max_run = cfg->max_run;
> +	fmt->max_frame_bytes = cfg->max_frame_bytes;
> +	fmt->avg_bit_rate = cfg->avg_bit_rate;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->channel_layout_tag = cfg->channel_layout_tag;
> +	fmt->pb = cfg->pb;
> +	fmt->mb = cfg->mb;
> +	fmt->kb = cfg->kb;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
> +
> +int q6asm_stream_media_format_block_ape(struct audio_client *ac,
> +					struct q6asm_ape_cfg *cfg)
> +{
> +	struct asm_ape_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->compatible_version = cfg->compatible_version;
> +	fmt->compression_level = cfg->compression_level;
> +	fmt->format_flags = cfg->format_flags;
> +	fmt->blocks_per_frame = cfg->blocks_per_frame;
> +	fmt->final_frame_blocks = cfg->final_frame_blocks;
> +	fmt->total_frames = cfg->total_frames;
> +	fmt->bits_per_sample = cfg->bits_per_sample;
> +	fmt->num_channels = cfg->num_channels;
> +	fmt->sample_rate = cfg->sample_rate;
> +	fmt->seek_table_present = cfg->seek_table_present;
> +
> +	rc = q6asm_ac_send_cmd_sync(ac, pkt);
> +	kfree(pkt);
> +
> +	return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
> +
>   /**
>    * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
>    *
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 5d9fbc75688c..38a207d6cd95 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -58,6 +58,34 @@ struct q6asm_wma_cfg {
>   	u32 adv_enc_options2;
>   };
>   
> +struct q6asm_alac_cfg {
> +	u32 frame_length;
> +	u8 compatible_version;
> +	u8 bit_depth;
> +	u8 pb;
> +	u8 mb;
> +	u8 kb;
> +	u8 num_channels;
> +	u16 max_run;
> +	u32 max_frame_bytes;
> +	u32 avg_bit_rate;
> +	u32 sample_rate;
> +	u32 channel_layout_tag;
> +};
> +
> +struct q6asm_ape_cfg {
> +	u16 compatible_version;
> +	u16 compression_level;
> +	u32 format_flags;
> +	u32 blocks_per_frame;
> +	u32 final_frame_blocks;
> +	u32 total_frames;
> +	u16 bits_per_sample;
> +	u16 num_channels;
> +	u32 sample_rate;
> +	u32 seek_table_present;
> +};
> +
>   typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
>   			  void *payload, void *priv);
>   struct audio_client;
> @@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
>   					   struct q6asm_wma_cfg *cfg);
>   int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
>   					    struct q6asm_wma_cfg *cfg);
> +int q6asm_stream_media_format_block_alac(struct audio_client *ac,
> +					 struct q6asm_alac_cfg *cfg);
> +int q6asm_stream_media_format_block_ape(struct audio_client *ac,
> +					struct q6asm_ape_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,
> 

[RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Vinod Koul]

Qualcomm DSPs also supports the ALAC and APE decoders, so add support
for these and convert the snd_codec_params to qdsp format.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
 1 file changed, 66 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 53c250778eea..948710759824 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	struct q6asm_dai_data *pdata;
 	struct q6asm_flac_cfg flac_cfg;
 	struct q6asm_wma_cfg wma_cfg;
+	struct q6asm_alac_cfg alac_cfg;
+	struct q6asm_ape_cfg ape_cfg;
 	unsigned int wma_v9 = 0;
 	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
 	struct snd_dec_wma *wma;
+	struct snd_dec_alac *alac;
+	struct snd_dec_ape *ape;
 
 	codec_options = &(prtd->codec_param.codec.options);
 
@@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
 			return -EIO;
 		}
+		break;
+
+	case SND_AUDIOCODEC_ALAC:
+		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+		alac = &codec_options->alac_d;
+
+		alac_cfg.sample_rate = params->codec.sample_rate;
+		alac_cfg.avg_bit_rate = params->codec.bit_rate;
+		alac_cfg.bit_depth = prtd->bits_per_sample;
+		alac_cfg.num_channels = params->codec.ch_in;
+
+		alac_cfg.frame_length = alac->frame_length;
+		alac_cfg.pb = alac->pb;
+		alac_cfg.mb = alac->mb;
+		alac_cfg.kb = alac->kb;
+		alac_cfg.max_run = alac->max_run;
+		alac_cfg.compatible_version = alac->compatible_version;
+		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+		switch (params->codec.ch_in) {
+		case 1:
+			alac_cfg.channel_layout_tag = (100 << 16) | 1;
+			break;
+		case 2:
+			alac_cfg.channel_layout_tag = (101 << 16) | 2;
+			break;
+		}
+		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+							   &alac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_APE:
+		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+		ape = &codec_options->ape_d;
+
+		ape_cfg.sample_rate = params->codec.sample_rate;
+		ape_cfg.num_channels = params->codec.ch_in;
+		ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+		ape_cfg.compatible_version = ape->compatible_version;
+		ape_cfg.compression_level = ape->compression_level;
+		ape_cfg.format_flags = ape->format_flags;
+		ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+		ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+		ape_cfg.total_frames = ape->total_frames;
+		ape_cfg.seek_table_present = ape->seek_table_present;
+
+		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+							  &ape_cfg);
+		if (ret < 0) {
+			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
 	default:
 		break;
 	}
@@ -855,10 +918,12 @@ 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 = 3;
+	caps->num_codecs = 5;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
 	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
 	caps->codecs[2] = SND_AUDIOCODEC_WMA;
+	caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+	caps->codecs[4] = SND_AUDIOCODEC_APE;
 
 	return 0;
 }
-- 
2.24.1

[RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Vinod Koul]

Qualcomm DSPs also supports the ALAC and APE decoders, so add support
for these and convert the snd_codec_params to qdsp format.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
 1 file changed, 66 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 53c250778eea..948710759824 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	struct q6asm_dai_data *pdata;
 	struct q6asm_flac_cfg flac_cfg;
 	struct q6asm_wma_cfg wma_cfg;
+	struct q6asm_alac_cfg alac_cfg;
+	struct q6asm_ape_cfg ape_cfg;
 	unsigned int wma_v9 = 0;
 	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
 	struct snd_dec_wma *wma;
+	struct snd_dec_alac *alac;
+	struct snd_dec_ape *ape;
 
 	codec_options = &(prtd->codec_param.codec.options);
 
@@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
 			return -EIO;
 		}
+		break;
+
+	case SND_AUDIOCODEC_ALAC:
+		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+		alac = &codec_options->alac_d;
+
+		alac_cfg.sample_rate = params->codec.sample_rate;
+		alac_cfg.avg_bit_rate = params->codec.bit_rate;
+		alac_cfg.bit_depth = prtd->bits_per_sample;
+		alac_cfg.num_channels = params->codec.ch_in;
+
+		alac_cfg.frame_length = alac->frame_length;
+		alac_cfg.pb = alac->pb;
+		alac_cfg.mb = alac->mb;
+		alac_cfg.kb = alac->kb;
+		alac_cfg.max_run = alac->max_run;
+		alac_cfg.compatible_version = alac->compatible_version;
+		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+		switch (params->codec.ch_in) {
+		case 1:
+			alac_cfg.channel_layout_tag = (100 << 16) | 1;
+			break;
+		case 2:
+			alac_cfg.channel_layout_tag = (101 << 16) | 2;
+			break;
+		}
+		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+							   &alac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_APE:
+		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+		ape = &codec_options->ape_d;
+
+		ape_cfg.sample_rate = params->codec.sample_rate;
+		ape_cfg.num_channels = params->codec.ch_in;
+		ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+		ape_cfg.compatible_version = ape->compatible_version;
+		ape_cfg.compression_level = ape->compression_level;
+		ape_cfg.format_flags = ape->format_flags;
+		ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+		ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+		ape_cfg.total_frames = ape->total_frames;
+		ape_cfg.seek_table_present = ape->seek_table_present;
+
+		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+							  &ape_cfg);
+		if (ret < 0) {
+			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
 	default:
 		break;
 	}
@@ -855,10 +918,12 @@ 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 = 3;
+	caps->num_codecs = 5;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
 	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
 	caps->codecs[2] = SND_AUDIOCODEC_WMA;
+	caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+	caps->codecs[4] = SND_AUDIOCODEC_APE;
 
 	return 0;
 }
-- 
2.24.1

Re: [RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs also supports the ALAC and APE decoders, so add support
> for these and convert the snd_codec_params to qdsp format.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>
One minor nit, other that,

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
>   1 file changed, 66 insertions(+), 1 deletion(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index 53c250778eea..948710759824 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	struct q6asm_dai_data *pdata;
>   	struct q6asm_flac_cfg flac_cfg;
>   	struct q6asm_wma_cfg wma_cfg;
> +	struct q6asm_alac_cfg alac_cfg;
> +	struct q6asm_ape_cfg ape_cfg;
>   	unsigned int wma_v9 = 0;
>   	struct device *dev = c->dev;
>   	int ret;
>   	union snd_codec_options *codec_options;
>   	struct snd_dec_flac *flac;
>   	struct snd_dec_wma *wma;
> +	struct snd_dec_alac *alac;
> +	struct snd_dec_ape *ape;
>   
>   	codec_options = &(prtd->codec_param.codec.options);
>   
> @@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
>   			return -EIO;
>   		}
> +		break;
> +
> +	case SND_AUDIOCODEC_ALAC:
> +		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
> +		alac = &codec_options->alac_d;
> +
> +		alac_cfg.sample_rate = params->codec.sample_rate;
> +		alac_cfg.avg_bit_rate = params->codec.bit_rate;
> +		alac_cfg.bit_depth = prtd->bits_per_sample;
> +		alac_cfg.num_channels = params->codec.ch_in;
> +
> +		alac_cfg.frame_length = alac->frame_length;
> +		alac_cfg.pb = alac->pb;
> +		alac_cfg.mb = alac->mb;
> +		alac_cfg.kb = alac->kb;
> +		alac_cfg.max_run = alac->max_run;
> +		alac_cfg.compatible_version = alac->compatible_version;
> +		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
> +
> +		switch (params->codec.ch_in) {
> +		case 1:
> +			alac_cfg.channel_layout_tag = (100 << 16) | 1;

We should probably define this layout tag in asm.h
something like:

#define ALAC_CHANNEL_LAYOUT_TAG_Mono (100<<16) | 1
#define ALAC_CHANNEL_LAYOUT_TAG_STEREO (100<<16) | 2

--srini

> +			break;
> +		case 2:
> +			alac_cfg.channel_layout_tag = (101 << 16) | 2;
> +			break;
> +		}

Re: [RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Srinivas Kandagatla]

On 13/03/2020 10:16, Vinod Koul wrote:
> Qualcomm DSPs also supports the ALAC and APE decoders, so add support
> for these and convert the snd_codec_params to qdsp format.
> 
> Signed-off-by: Vinod Koul <vkoul@kernel.org>
One minor nit, other that,

Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

> ---
>   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
>   1 file changed, 66 insertions(+), 1 deletion(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index 53c250778eea..948710759824 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   	struct q6asm_dai_data *pdata;
>   	struct q6asm_flac_cfg flac_cfg;
>   	struct q6asm_wma_cfg wma_cfg;
> +	struct q6asm_alac_cfg alac_cfg;
> +	struct q6asm_ape_cfg ape_cfg;
>   	unsigned int wma_v9 = 0;
>   	struct device *dev = c->dev;
>   	int ret;
>   	union snd_codec_options *codec_options;
>   	struct snd_dec_flac *flac;
>   	struct snd_dec_wma *wma;
> +	struct snd_dec_alac *alac;
> +	struct snd_dec_ape *ape;
>   
>   	codec_options = &(prtd->codec_param.codec.options);
>   
> @@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
>   			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
>   			return -EIO;
>   		}
> +		break;
> +
> +	case SND_AUDIOCODEC_ALAC:
> +		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
> +		alac = &codec_options->alac_d;
> +
> +		alac_cfg.sample_rate = params->codec.sample_rate;
> +		alac_cfg.avg_bit_rate = params->codec.bit_rate;
> +		alac_cfg.bit_depth = prtd->bits_per_sample;
> +		alac_cfg.num_channels = params->codec.ch_in;
> +
> +		alac_cfg.frame_length = alac->frame_length;
> +		alac_cfg.pb = alac->pb;
> +		alac_cfg.mb = alac->mb;
> +		alac_cfg.kb = alac->kb;
> +		alac_cfg.max_run = alac->max_run;
> +		alac_cfg.compatible_version = alac->compatible_version;
> +		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
> +
> +		switch (params->codec.ch_in) {
> +		case 1:
> +			alac_cfg.channel_layout_tag = (100 << 16) | 1;

We should probably define this layout tag in asm.h
something like:

#define ALAC_CHANNEL_LAYOUT_TAG_Mono (100<<16) | 1
#define ALAC_CHANNEL_LAYOUT_TAG_STEREO (100<<16) | 2

--srini

> +			break;
> +		case 2:
> +			alac_cfg.channel_layout_tag = (101 << 16) | 2;
> +			break;
> +		}

Re: [RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Vinod Koul]

On 13-03-20, 12:15, Srinivas Kandagatla wrote:
> 
> 
> On 13/03/2020 10:16, Vinod Koul wrote:
> > Qualcomm DSPs also supports the ALAC and APE decoders, so add support
> > for these and convert the snd_codec_params to qdsp format.
> > 
> > Signed-off-by: Vinod Koul <vkoul@kernel.org>
> One minor nit, other that,
> 
> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Thanks Srini for the reviews

> > ---
> >   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
> >   1 file changed, 66 insertions(+), 1 deletion(-)
> > 
> > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> > index 53c250778eea..948710759824 100644
> > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> > @@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
> >   	struct q6asm_dai_data *pdata;
> >   	struct q6asm_flac_cfg flac_cfg;
> >   	struct q6asm_wma_cfg wma_cfg;
> > +	struct q6asm_alac_cfg alac_cfg;
> > +	struct q6asm_ape_cfg ape_cfg;
> >   	unsigned int wma_v9 = 0;
> >   	struct device *dev = c->dev;
> >   	int ret;
> >   	union snd_codec_options *codec_options;
> >   	struct snd_dec_flac *flac;
> >   	struct snd_dec_wma *wma;
> > +	struct snd_dec_alac *alac;
> > +	struct snd_dec_ape *ape;
> >   	codec_options = &(prtd->codec_param.codec.options);
> > @@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
> >   			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
> >   			return -EIO;
> >   		}
> > +		break;
> > +
> > +	case SND_AUDIOCODEC_ALAC:
> > +		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
> > +		alac = &codec_options->alac_d;
> > +
> > +		alac_cfg.sample_rate = params->codec.sample_rate;
> > +		alac_cfg.avg_bit_rate = params->codec.bit_rate;
> > +		alac_cfg.bit_depth = prtd->bits_per_sample;
> > +		alac_cfg.num_channels = params->codec.ch_in;
> > +
> > +		alac_cfg.frame_length = alac->frame_length;
> > +		alac_cfg.pb = alac->pb;
> > +		alac_cfg.mb = alac->mb;
> > +		alac_cfg.kb = alac->kb;
> > +		alac_cfg.max_run = alac->max_run;
> > +		alac_cfg.compatible_version = alac->compatible_version;
> > +		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
> > +
> > +		switch (params->codec.ch_in) {
> > +		case 1:
> > +			alac_cfg.channel_layout_tag = (100 << 16) | 1;
> 
> We should probably define this layout tag in asm.h
> something like:
> 
> #define ALAC_CHANNEL_LAYOUT_TAG_Mono (100<<16) | 1
> #define ALAC_CHANNEL_LAYOUT_TAG_STEREO (100<<16) | 2

Sure I will add these

-- 
~Vinod

Re: [RESEND PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders

[Vinod Koul]

On 13-03-20, 12:15, Srinivas Kandagatla wrote:
> 
> 
> On 13/03/2020 10:16, Vinod Koul wrote:
> > Qualcomm DSPs also supports the ALAC and APE decoders, so add support
> > for these and convert the snd_codec_params to qdsp format.
> > 
> > Signed-off-by: Vinod Koul <vkoul@kernel.org>
> One minor nit, other that,
> 
> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Thanks Srini for the reviews

> > ---
> >   sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
> >   1 file changed, 66 insertions(+), 1 deletion(-)
> > 
> > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> > index 53c250778eea..948710759824 100644
> > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> > @@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
> >   	struct q6asm_dai_data *pdata;
> >   	struct q6asm_flac_cfg flac_cfg;
> >   	struct q6asm_wma_cfg wma_cfg;
> > +	struct q6asm_alac_cfg alac_cfg;
> > +	struct q6asm_ape_cfg ape_cfg;
> >   	unsigned int wma_v9 = 0;
> >   	struct device *dev = c->dev;
> >   	int ret;
> >   	union snd_codec_options *codec_options;
> >   	struct snd_dec_flac *flac;
> >   	struct snd_dec_wma *wma;
> > +	struct snd_dec_alac *alac;
> > +	struct snd_dec_ape *ape;
> >   	codec_options = &(prtd->codec_param.codec.options);
> > @@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
> >   			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
> >   			return -EIO;
> >   		}
> > +		break;
> > +
> > +	case SND_AUDIOCODEC_ALAC:
> > +		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
> > +		alac = &codec_options->alac_d;
> > +
> > +		alac_cfg.sample_rate = params->codec.sample_rate;
> > +		alac_cfg.avg_bit_rate = params->codec.bit_rate;
> > +		alac_cfg.bit_depth = prtd->bits_per_sample;
> > +		alac_cfg.num_channels = params->codec.ch_in;
> > +
> > +		alac_cfg.frame_length = alac->frame_length;
> > +		alac_cfg.pb = alac->pb;
> > +		alac_cfg.mb = alac->mb;
> > +		alac_cfg.kb = alac->kb;
> > +		alac_cfg.max_run = alac->max_run;
> > +		alac_cfg.compatible_version = alac->compatible_version;
> > +		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
> > +
> > +		switch (params->codec.ch_in) {
> > +		case 1:
> > +			alac_cfg.channel_layout_tag = (100 << 16) | 1;
> 
> We should probably define this layout tag in asm.h
> something like:
> 
> #define ALAC_CHANNEL_LAYOUT_TAG_Mono (100<<16) | 1
> #define ALAC_CHANNEL_LAYOUT_TAG_STEREO (100<<16) | 2

Sure I will add these

-- 
~Vinod

[RESEND PATCH v2 9/9] ALSA: compress: bump the version

[Vinod Koul]

We have added support for bunch of new decoders and parameters for
decoders. To help users find the support bump the version up to 0,2,0.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_offload.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 56d95673ce0f..7184265c0b0d 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -31,7 +31,7 @@
 #include <sound/compress_params.h>
 
 
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0)
 /**
  * struct snd_compressed_buffer - compressed buffer
  * @fragment_size: size of buffer fragment in bytes
-- 
2.24.1

[RESEND PATCH v2 9/9] ALSA: compress: bump the version

[Vinod Koul]

We have added support for bunch of new decoders and parameters for
decoders. To help users find the support bump the version up to 0,2,0.

Signed-off-by: Vinod Koul <vkoul@kernel.org>
---
 include/uapi/sound/compress_offload.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 56d95673ce0f..7184265c0b0d 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -31,7 +31,7 @@
 #include <sound/compress_params.h>
 
 
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0)
 /**
  * struct snd_compressed_buffer - compressed buffer
  * @fragment_size: size of buffer fragment in bytes
-- 
2.24.1