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Message-ID: <c4d934c1-0218-4147-882f-279795bcd1f4@oss.qualcomm.com>
Date: Wed, 18 Jun 2025 13:34:04 +0100
From: Srinivas Kandagatla <srinivas.kandagatla@....qualcomm.com>
To: Alexey Klimov <alexey.klimov@...aro.org>, Vinod Koul <vkoul@...nel.org>,
Jaroslav Kysela <perex@...ex.cz>, Takashi Iwai <tiwai@...e.com>,
Srinivas Kandagatla <srini@...nel.org>,
Liam Girdwood <lgirdwood@...il.com>, Mark Brown <broonie@...nel.org>
Cc: Patrick Lai <plai@....qualcomm.com>,
Annemarie Porter <annemari@...cinc.com>, linux-sound@...r.kernel.org,
linux-kernel@...r.kernel.org, linux-arm-msm@...r.kernel.org,
Krzysztof Kozlowski <krzysztof.kozlowski@...aro.org>,
kernel@....qualcomm.com, Ekansh Gupta <ekansh.gupta@....qualcomm.com>,
Pierre-Louis Bossart <pierre-louis.bossart@...ux.dev>
Subject: Re: [PATCH RFC 2/2] ASoC: qcom: qdsp6/audioreach: add support for
offloading raw opus playback
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@...aro.org>
Thanks for picking this up Alexey,
Same, co-dev by should be good attribute for such things.
>
> Cc: Annemarie Porter <annemari@...cinc.com>
> Cc: Srinivas Kandagatla <srini@...nel.org>
> Cc: Vinod Koul <vkoul@...nel.org>
> Signed-off-by: Alexey Klimov <alexey.klimov@...aro.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;
> }
>
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