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Wed, 18 Jun 2025 05:34:06 -0700 (PDT) X-Google-Smtp-Source: AGHT+IFmOq0+HrAD5r/cYf5XwPht6kA6wl8WHkSrFkKeJLapYf8ANWSDAI5kyUuHdTLnUrshNBaNyA== X-Received: by 2002:a05:620a:a20d:b0:7d3:e83c:951d with SMTP id af79cd13be357-7d3e83ce085mr412947885a.21.1750250046135; Wed, 18 Jun 2025 05:34:06 -0700 (PDT) Received: from [192.168.68.111] ([5.133.47.210]) by smtp.googlemail.com with ESMTPSA id 5b1f17b1804b1-4535969b60asm13435595e9.0.2025.06.18.05.34.05 (version=TLS1_3 cipher=TLS_AES_128_GCM_SHA256 bits=128/128); Wed, 18 Jun 2025 05:34:05 -0700 (PDT) Message-ID: Date: Wed, 18 Jun 2025 13:34:04 +0100 Precedence: bulk X-Mailing-List: linux-kernel@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 User-Agent: Mozilla Thunderbird Subject: Re: [PATCH RFC 2/2] ASoC: qcom: qdsp6/audioreach: add support for offloading raw opus playback To: Alexey Klimov , 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 , Pierre-Louis Bossart References: <20250616-opus_codec_rfc_v1-v1-0-1f70b0a41a70@linaro.org> <20250616-opus_codec_rfc_v1-v1-2-1f70b0a41a70@linaro.org> Content-Language: en-US From: Srinivas Kandagatla In-Reply-To: <20250616-opus_codec_rfc_v1-v1-2-1f70b0a41a70@linaro.org> Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 7bit X-Proofpoint-Spam-Details-Enc: AW1haW4tMjUwNjE4MDEwNiBTYWx0ZWRfX9FpSuYGbrQjv VSzvMe82ZAA8LJufC5eFWUZ+p39MYWps+rpkS1IBx/rNaKK+1E98tXsX+Ml/opaR9a+orQxU1os Q+4cUAXfiJ6GFYjIouqefFhMGF4ZJzyvoWCa2mJea3GEiLYPELuT7DAkEB9b7rI5AHzrVWwLmZn 1lINkyU7EVwng9F3DgGTDwc03IVilJvZY+FJ/wi3G0SaZeKHrxqpThc/sfcO6kAdouEtaUpXsO0 RnBbvRgy+2Tc/5K0kP1sbpZsHB7BTTAMNEeWjfJwy1A11djoSUuh8SZ5f4748lJNul5pd9LvYGv 7OVHf3caHWF8h7vVqdrepAuhgCpl08Sn+wxzawkdUyB3+rbv77oNl99vz+qeOfVLLICxvjB5ikQ wMlqYj5mYoD52+uWAR43ym/G3V7+243KvbxE8cLdy9W7uNp71fSibufKzKt+e7bpdMLs0SS8 X-Authority-Analysis: v=2.4 cv=PtaTbxM3 c=1 sm=1 tr=0 ts=6852b240 cx=c_pps a=HLyN3IcIa5EE8TELMZ618Q==:117 a=ZsC4DHZuhs/kKio7QBcDoQ==:17 a=IkcTkHD0fZMA:10 a=6IFa9wvqVegA:10 a=KKAkSRfTAAAA:8 a=COk6AnOGAAAA:8 a=VwQbUJbxAAAA:8 a=9AdGPr7lSesGDd7r4mQA:9 a=QEXdDO2ut3YA:10 a=bTQJ7kPSJx9SKPbeHEYW:22 a=cvBusfyB2V15izCimMoJ:22 a=TjNXssC_j7lpFel5tvFf:22 X-Proofpoint-ORIG-GUID: uwl0w_EXz5iiPs3UiT0JnPq5MGUfo7y2 X-Proofpoint-GUID: uwl0w_EXz5iiPs3UiT0JnPq5MGUfo7y2 X-Proofpoint-Virus-Version: vendor=baseguard engine=ICAP:2.0.293,Aquarius:18.0.1099,Hydra:6.0.736,FMLib:17.12.80.40 definitions=2025-06-18_05,2025-06-18_02,2025-03-28_01 X-Proofpoint-Spam-Details: rule=outbound_notspam policy=outbound score=0 bulkscore=0 clxscore=1015 mlxlogscore=999 suspectscore=0 malwarescore=0 lowpriorityscore=0 impostorscore=0 adultscore=0 spamscore=0 priorityscore=1501 phishscore=0 mlxscore=0 classifier=spam authscore=0 authtc=n/a authcc= route=outbound adjust=0 reason=mlx scancount=1 engine=8.19.0-2505280000 definitions=main-2506180106 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 Thanks for picking this up Alexey, Same, co-dev by should be good attribute for such things. > > 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); 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; > } >