Re: [PATCH RFC 2/2] ASoC: qcom: qdsp6/audioreach: add support for offloading raw opus playback

[Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>]

On 6/16/25 4:26 PM, Alexey Klimov wrote:
> Add support for OPUS module, OPUS format ID, media format payload struct
> and make it all recognizable by audioreach compress playback path.
> 
> At this moment this only supports raw or plain OPUS packets not
> encapsulated in container (for instance OGG container). For this usecase
> each OPUS packet needs to be prepended with 4-bytes long length field
> which is expected to be done by userspace applications. This is
> Qualcomm DSP specific requirement.
> > This patch is based on earlier work done by
> Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Thanks for picking this up Alexey,

Same, co-dev by should be good attribute for such things.


> 
> Cc: Annemarie Porter <annemari@quicinc.com>
> Cc: Srinivas Kandagatla <srini@kernel.org>
> Cc: Vinod Koul <vkoul@kernel.org>
> Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
> ---
>  sound/soc/qcom/qdsp6/audioreach.c | 33 +++++++++++++++++++++++++++++++++
>  sound/soc/qcom/qdsp6/audioreach.h | 17 +++++++++++++++++
>  sound/soc/qcom/qdsp6/q6apm-dai.c  |  3 ++-
>  sound/soc/qcom/qdsp6/q6apm.c      |  3 +++
>  4 files changed, 55 insertions(+), 1 deletion(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
> index 4ebaaf736fb98a5a8a58d06416b3ace2504856e1..09e3a4da945d61b6915bf8b6f382c25ae94c5888 100644
> --- a/sound/soc/qcom/qdsp6/audioreach.c
> +++ b/sound/soc/qcom/qdsp6/audioreach.c
> @@ -859,6 +859,7 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
>  	struct payload_media_fmt_aac_t *aac_cfg;
>  	struct payload_media_fmt_pcm *mp3_cfg;
>  	struct payload_media_fmt_flac_t *flac_cfg;
> +	struct payload_media_fmt_opus_t *opus_cfg;
>  
>  	switch (mcfg->fmt) {
>  	case SND_AUDIOCODEC_MP3:
> @@ -901,6 +902,38 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
>  		flac_cfg->min_frame_size = mcfg->codec.options.flac_d.min_frame_size;
>  		flac_cfg->max_frame_size = mcfg->codec.options.flac_d.max_frame_size;
>  		break;
> +	case SND_AUDIOCODEC_OPUS_RAW:
> +		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
> +		media_fmt_hdr->fmt_id = MEDIA_FMT_ID_OPUS;
> +		media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_opus_t);

maybe sizeof(*opus_cfg)?

> +		p = p + sizeof(*media_fmt_hdr);
> +		opus_cfg = p;
> +		/* raw opus packets prepended with 4 bytes of length */
> +		opus_cfg->bitstream_format = 1;
> +		/*
> +		 * payload_type:
> +		 * 0 -- read metadata from opus stream;
> +		 * 1 -- metadata is provided by filling in the struct here.
> +		 */
> +		opus_cfg->payload_type = 1;
> +		opus_cfg->version = mcfg->codec.options.opus_d.version;
> +		opus_cfg->num_channels = mcfg->codec.options.opus_d.num_channels;
> +		opus_cfg->pre_skip = mcfg->codec.options.opus_d.pre_skip;
> +		opus_cfg->sample_rate = mcfg->codec.options.opus_d.sample_rate;
> +		opus_cfg->output_gain = mcfg->codec.options.opus_d.output_gain;
> +		opus_cfg->mapping_family = mcfg->codec.options.opus_d.mapping_family;
> +		opus_cfg->stream_count = mcfg->codec.options.opus_d.stream_count;
> +		opus_cfg->coupled_count = mcfg->codec.options.opus_d.coupled_count;
> +
> +		if (mcfg->codec.options.opus_d.num_channels == 1) {
> +			opus_cfg->channel_mapping[0] = PCM_CHANNEL_FL;
> +		} else if (mcfg->codec.options.opus_d.num_channels == 2) {
> +			opus_cfg->channel_mapping[0] = PCM_CHANNEL_FL;
> +			opus_cfg->channel_mapping[1] = PCM_CHANNEL_FR;
> +		}

Pl use audioreach_set_default_channel_mapping() to fill in the channel
mapping data.

Why are we not using channel mapping info from the snd_dec_opus struct here?


> +
> +		opus_cfg->reserved[0] = opus_cfg->reserved[1] = opus_cfg->reserved[2] = 0;
> +		break;
>  	default:
>  		return -EINVAL;
>  	}
> diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
> index 61a69df4f50f6cc90b56697c272ade6f1411bbf2..512ea24fd402c95f98db790313b756a5ba3dcca1 100644
> --- a/sound/soc/qcom/qdsp6/audioreach.h
> +++ b/sound/soc/qcom/qdsp6/audioreach.h
> @@ -29,6 +29,7 @@ struct q6apm_graph;
>  #define MODULE_ID_MP3_DECODE		0x0700103B
>  #define MODULE_ID_GAPLESS		0x0700104D
>  #define MODULE_ID_DISPLAY_PORT_SINK	0x07001069
> +#define MODULE_ID_OPUS_DEC		0x07001174
>  
>  #define APM_CMD_GET_SPF_STATE		0x01001021
>  #define APM_CMD_RSP_GET_SPF_STATE	0x02001007
> @@ -255,6 +256,22 @@ struct payload_media_fmt_aac_t {
>  	uint32_t sample_rate;
>  } __packed;
>  
> +#define MEDIA_FMT_ID_OPUS	0x09001039
> +struct payload_media_fmt_opus_t {
> +	uint16_t bitstream_format;
> +	uint16_t payload_type;
> +	uint8_t version;
> +	uint8_t num_channels;
> +	uint16_t pre_skip;
> +	uint32_t sample_rate;
> +	uint16_t output_gain;
> +	uint8_t mapping_family;
> +	uint8_t stream_count;
> +	uint8_t coupled_count;
> +	uint8_t channel_mapping[8];
> +	uint8_t reserved[3];
> +} __packed;
> +
>  #define DATA_CMD_WR_SH_MEM_EP_EOS			0x04001002
>  #define WR_SH_MEM_EP_EOS_POLICY_LAST	1
>  #define WR_SH_MEM_EP_EOS_POLICY_EACH	2
> diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
> index 2cd522108221a2ec5c7bbbb63f7d4ae4f8001cf6..7da91ed297f4a5ed39ca0747804cabd579634b50 100644
> --- a/sound/soc/qcom/qdsp6/q6apm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
> @@ -550,10 +550,11 @@ static int q6apm_dai_compr_get_caps(struct snd_soc_component *component,
>  	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 = 4;
>  	caps->codecs[0] = SND_AUDIOCODEC_MP3;
>  	caps->codecs[1] = SND_AUDIOCODEC_AAC;
>  	caps->codecs[2] = SND_AUDIOCODEC_FLAC;
> +	caps->codecs[3] = SND_AUDIOCODEC_OPUS_RAW;
>  
>  	return 0;
>  }
> diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
> index b4ffa0f0b188e2c32fdfb863b9130d1d11e578dd..0e667a7eb5467bdd65326099132e8ba9dfefa21e 100644
> --- a/sound/soc/qcom/qdsp6/q6apm.c
> +++ b/sound/soc/qcom/qdsp6/q6apm.c
> @@ -354,6 +354,9 @@ int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph,
>  	case SND_AUDIOCODEC_FLAC:
>  		module_id = MODULE_ID_FLAC_DEC;
>  		break;
> +	case SND_AUDIOCODEC_OPUS_RAW:
> +		module_id = MODULE_ID_OPUS_DEC;
> +		break;
>  	default:
>  		return -EINVAL;
>  	}
>