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Mon, 16 Jun 2025 08:26:33 -0700 (PDT) From: Alexey Klimov Date: Mon, 16 Jun 2025 16:26:23 +0100 Subject: [PATCH RFC 2/2] ASoC: qcom: qdsp6/audioreach: add support for offloading raw opus playback Precedence: bulk X-Mailing-List: linux-arm-msm@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 Message-Id: <20250616-opus_codec_rfc_v1-v1-2-1f70b0a41a70@linaro.org> References: <20250616-opus_codec_rfc_v1-v1-0-1f70b0a41a70@linaro.org> In-Reply-To: <20250616-opus_codec_rfc_v1-v1-0-1f70b0a41a70@linaro.org> To: Vinod Koul , Jaroslav Kysela , Takashi Iwai , Srinivas Kandagatla , Liam Girdwood , Mark Brown Cc: Patrick Lai , Annemarie Porter , linux-sound@vger.kernel.org, linux-kernel@vger.kernel.org, linux-arm-msm@vger.kernel.org, Krzysztof Kozlowski , kernel@oss.qualcomm.com, Ekansh Gupta , Alexey Klimov , Pierre-Louis Bossart X-Mailer: b4 0.14.2 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 Cc: Annemarie Porter Cc: Srinivas Kandagatla Cc: Vinod Koul Signed-off-by: Alexey Klimov --- 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); + 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; + } + + 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; }