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ASoC: q6dsp: audioreach: Add support to set compress format params

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Add function for setting compress params.

Signed-off-by: Mohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230619101653.9750-6-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Mohammad Rafi Shaik 2023-06-19 11:16:47 +01:00 committed by Mark Brown
parent c7548f5990
commit e41521b6e2
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
3 changed files with 147 additions and 17 deletions
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135
sound/soc/qcom/qdsp6/audioreach.c
View File

@ -834,6 +834,99 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
return rc;
}
static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
void *p, struct audioreach_module_config *mcfg)
{
struct payload_media_fmt_aac_t *aac_cfg;
struct payload_media_fmt_pcm *mp3_cfg;
struct payload_media_fmt_flac_t *flac_cfg;
switch (mcfg->fmt) {
case SND_AUDIOCODEC_MP3:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_MP3;
media_fmt_hdr->payload_size = 0;
p = p + sizeof(*media_fmt_hdr);
mp3_cfg = p;
mp3_cfg->sample_rate = mcfg->sample_rate;
mp3_cfg->bit_width = mcfg->bit_width;
mp3_cfg->alignment = PCM_LSB_ALIGNED;
mp3_cfg->bits_per_sample = mcfg->bit_width;
mp3_cfg->q_factor = mcfg->bit_width - 1;
mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
mp3_cfg->num_channels = mcfg->num_channels;
if (mcfg->num_channels == 1) {
mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
} else if (mcfg->num_channels == 2) {
mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R;
}
break;
case SND_AUDIOCODEC_AAC:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_AAC;
media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_aac_t);
p = p + sizeof(*media_fmt_hdr);
aac_cfg = p;
aac_cfg->aac_fmt_flag = 0;
aac_cfg->audio_obj_type = 5;
aac_cfg->num_channels = mcfg->num_channels;
aac_cfg->total_size_of_PCE_bits = 0;
aac_cfg->sample_rate = mcfg->sample_rate;
break;
case SND_AUDIOCODEC_FLAC:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_FLAC;
media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_flac_t);
p = p + sizeof(*media_fmt_hdr);
flac_cfg = p;
flac_cfg->sample_size = mcfg->codec.options.flac_d.sample_size;
flac_cfg->num_channels = mcfg->num_channels;
flac_cfg->min_blk_size = mcfg->codec.options.flac_d.min_blk_size;
flac_cfg->max_blk_size = mcfg->codec.options.flac_d.max_blk_size;
flac_cfg->sample_rate = mcfg->sample_rate;
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;
default:
return -EINVAL;
}
return 0;
}
int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
{
struct media_format *header;
struct gpr_pkt *pkt;
int iid, payload_size, rc;
void *p;
payload_size = sizeof(struct apm_sh_module_media_fmt_cmd);
iid = q6apm_graph_get_rx_shmem_module_iid(graph);
pkt = audioreach_alloc_cmd_pkt(payload_size, DATA_CMD_WR_SH_MEM_EP_MEDIA_FORMAT,
0, graph->port->id, iid);
if (IS_ERR(pkt))
return -ENOMEM;
p = (void *)pkt + GPR_HDR_SIZE;
header = p;
rc = audioreach_set_compr_media_format(header, p, mcfg);
if (rc) {
kfree(pkt);
return rc;
}
rc = gpr_send_port_pkt(graph->port, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(audioreach_compr_set_param);
static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
@ -1037,25 +1130,33 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
p = p + APM_MODULE_PARAM_DATA_SIZE;
header = p;
header->data_format = DATA_FORMAT_FIXED_POINT;
header->fmt_id = MEDIA_FMT_ID_PCM;
header->payload_size = payload_size - sizeof(*header);
if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
header->data_format = DATA_FORMAT_FIXED_POINT;
header->fmt_id = MEDIA_FMT_ID_PCM;
header->payload_size = payload_size - sizeof(*header);
p = p + sizeof(*header);
cfg = p;
cfg->sample_rate = mcfg->sample_rate;
cfg->bit_width = mcfg->bit_width;
cfg->alignment = PCM_LSB_ALIGNED;
cfg->bits_per_sample = mcfg->bit_width;
cfg->q_factor = mcfg->bit_width - 1;
cfg->endianness = PCM_LITTLE_ENDIAN;
cfg->num_channels = mcfg->num_channels;
p = p + sizeof(*header);
cfg = p;
cfg->sample_rate = mcfg->sample_rate;
cfg->bit_width = mcfg->bit_width;
cfg->alignment = PCM_LSB_ALIGNED;
cfg->bits_per_sample = mcfg->bit_width;
cfg->q_factor = mcfg->bit_width - 1;
cfg->endianness = PCM_LITTLE_ENDIAN;
cfg->num_channels = mcfg->num_channels;
if (mcfg->num_channels == 1) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
} else if (num_channels == 2) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
cfg->channel_mapping[1] = PCM_CHANNEL_R;
if (mcfg->num_channels == 1)
cfg->channel_mapping[0] = PCM_CHANNEL_L;
else if (num_channels == 2) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
cfg->channel_mapping[1] = PCM_CHANNEL_R;
}
} else {
rc = audioreach_set_compr_media_format(header, p, mcfg);
if (rc) {
kfree(pkt);
return rc;
}
}
rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);

28
sound/soc/qcom/qdsp6/audioreach.h
View File

@ -148,12 +148,15 @@ struct param_id_enc_bitrate_param {
} __packed;
#define DATA_FORMAT_FIXED_POINT 1
#define DATA_FORMAT_GENERIC_COMPRESSED 5
#define DATA_FORMAT_RAW_COMPRESSED 6
#define PCM_LSB_ALIGNED 1
#define PCM_MSB_ALIGNED 2
#define PCM_LITTLE_ENDIAN 1
#define PCM_BIT_ENDIAN 2
#define MEDIA_FMT_ID_PCM 0x09001000
#define MEDIA_FMT_ID_MP3 0x09001009
#define PCM_CHANNEL_L 1
#define PCM_CHANNEL_R 2
#define SAMPLE_RATE_48K 48000
@ -231,6 +234,28 @@ struct apm_media_format {
uint32_t payload_size;
} __packed;
#define MEDIA_FMT_ID_FLAC 0x09001004
struct payload_media_fmt_flac_t {
uint16_t num_channels;
uint16_t sample_size;
uint16_t min_blk_size;
uint16_t max_blk_size;
uint32_t sample_rate;
uint32_t min_frame_size;
uint32_t max_frame_size;
} __packed;
#define MEDIA_FMT_ID_AAC 0x09001001
struct payload_media_fmt_aac_t {
uint16_t aac_fmt_flag;
uint16_t audio_obj_type;
uint16_t num_channels;
uint16_t total_size_of_PCE_bits;
uint32_t sample_rate;
} __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
@ -730,6 +755,7 @@ struct audioreach_module_config {
u32 channel_allocation;
u32 sd_line_mask;
int fmt;
struct snd_codec codec;
u8 channel_map[AR_PCM_MAX_NUM_CHANNEL];
};
@ -768,4 +794,6 @@ int audioreach_gain_set_vol_ctrl(struct q6apm *apm,
struct audioreach_module *module, int vol);
int audioreach_send_u32_param(struct q6apm_graph *graph, struct audioreach_module *module,
uint32_t param_id, uint32_t param_val);
int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
#endif /* __AUDIOREACH_H__ */

1
sound/soc/qcom/qdsp6/q6apm-dai.c
View File

@ -155,6 +155,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
cfg.sample_rate = runtime->rate;
cfg.num_channels = runtime->channels;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = SND_AUDIOCODEC_PCM;
if (prtd->state) {
/* clear the previous setup if any */