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>
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);
+ 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;
}
--
2.47.2
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;
> }
>
On Wed Jun 18, 2025 at 1:34 PM BST, Srinivas Kandagatla wrote:
>
>
> 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.
Thanks. I'll update it for the next version.
>> 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)?
Ack.
Best regards,
Alexey
On Wed Jun 18, 2025 at 1:34 PM BST, Srinivas Kandagatla wrote:
>
>
> 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.
I need your Signed-off-by then and/or permission to use your Sign off.
>> 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)?
Yes, I can update that.
>> + 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?
Okay, I was re-reading RFC and can't really get my head around this.
So first I came up with something like this:
switch (opus_cfg->mapping_family) {
case 0:
if (num_chan == 1 || num_chan == 2)
audioreach_set_default_channel_mapping(ch_map, num_chan);
else
/* mapping family 0 allows only 1 or 2 channels */
return -EINVAL;
break;
case 1:
if (num_chan > 8)
return -EINVAL;
if (mcfg->codec.options.opus_d.coupled_count > mcfg->codec.options.opus_d.stream_count)
return -EINVAL;
memcpy(ch_map, mcfg->codec.options.opus_d.channel_map, sizeof(mcfg->codec.options.opus_d.channel_map));
break;
default:
/* mapping family 2..255 shouldn't be allowed to playback */
return -EOPNOTSUPP;
}
but I don't think above is correct at all.
After re-reading the RFC few more times. It looks that channel_mapping in
opus struct has nothing to do with channel mapping that we need to provide
to DSP here. The channel mapping maps "decoded" channels to output channels
and seems to be needed by opus decoder logic and in some sense is internal
thingy to correctly construct sound for output channel from opus stream(s).
In other words if output channel is present and valid then channel_mapping
describes how and which decoded stream or streams (coupled or uncoupled)
to use for producing sound output for that output channel.
This is described in https://www.rfc-editor.org/rfc/rfc7845#section-5.1.1
The number of output channels here actually matters for us. We can construct
mapping for channels that we pass to DSP based just only on the number of
output channels here and let DSP to figure out how to scatter or downmix or
upmix them to its own output channels.
Conclusion from my understanding:
-- we shouldn't mess with opus_cfg->channel_mapping here, it is needed for
the correct operation of decoder, we shouldn't call
audioreach_set_default_channel_mapping() on it;
-- mapping output channels to provide the mapping to DSP might require some
rework or I need to look into this.
Or something else that didn't came up in my mind yet.
Also, I don't have any test files to test mapping_family 1 and some tricky
cases here. As far as I understand, it works just fine right now with
mapping_family 0.
Best regards,
Alexey
On Thu Jul 3, 2025 at 3:33 PM BST, Alexey Klimov wrote:
> On Wed Jun 18, 2025 at 1:34 PM BST, Srinivas Kandagatla wrote:
[...]
>> 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?
>
> Okay, I was re-reading RFC and can't really get my head around this.
>
> So first I came up with something like this:
>
> switch (opus_cfg->mapping_family) {
> case 0:
> if (num_chan == 1 || num_chan == 2)
> audioreach_set_default_channel_mapping(ch_map, num_chan);
> else
> /* mapping family 0 allows only 1 or 2 channels */
> return -EINVAL;
> break;
> case 1:
> if (num_chan > 8)
> return -EINVAL;
> if (mcfg->codec.options.opus_d.coupled_count > mcfg->codec.options.opus_d.stream_count)
> return -EINVAL;
>
> memcpy(ch_map, mcfg->codec.options.opus_d.channel_map, sizeof(mcfg->codec.options.opus_d.channel_map));
> break;
> default:
> /* mapping family 2..255 shouldn't be allowed to playback */
> return -EOPNOTSUPP;
> }
>
> but I don't think above is correct at all.
>
> After re-reading the RFC few more times. It looks that channel_mapping in
> opus struct has nothing to do with channel mapping that we need to provide
> to DSP here. The channel mapping maps "decoded" channels to output channels
> and seems to be needed by opus decoder logic and in some sense is internal
> thingy to correctly construct sound for output channel from opus stream(s).
> In other words if output channel is present and valid then channel_mapping
> describes how and which decoded stream or streams (coupled or uncoupled)
> to use for producing sound output for that output channel.
> This is described in https://www.rfc-editor.org/rfc/rfc7845#section-5.1.1
>
> The number of output channels here actually matters for us. We can construct
> mapping for channels that we pass to DSP based just only on the number of
> output channels here and let DSP to figure out how to scatter or downmix or
> upmix them to its own output channels.
>
> Conclusion from my understanding:
> -- we shouldn't mess with opus_cfg->channel_mapping here, it is needed for
> the correct operation of decoder, we shouldn't call
> audioreach_set_default_channel_mapping() on it;
> -- mapping output channels to provide the mapping to DSP might require some
> rework or I need to look into this.
>
> Or something else that didn't came up in my mind yet.
As discussed during out-of-list chats I'll update it to include mfc module in
compress-playback path that should handle the mapping to output channels.
I already have a change for that locally and it seems to work, at least it
doesn't break playback.
Thanks,
Alexey
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