From: Mohammad Rafi Shaik <quic_mohs@xxxxxxxxxxx>
Add function for setting compress params.
Signed-off-by: Mohammad Rafi Shaik <quic_mohs@xxxxxxxxxxx>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx>
---
sound/soc/qcom/qdsp6/audioreach.c | 139 ++++++++++++++++++++++++++----
sound/soc/qcom/qdsp6/audioreach.h | 28 ++++++
sound/soc/qcom/qdsp6/q6apm-dai.c | 1 +
3 files changed, 149 insertions(+), 19 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 34cbc4d05918..ba262408b27a 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -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);
-
- 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->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;
+
+ 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);
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index c4e03a49ac82..dc089879b501 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -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__ */
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 7f02f5b2c33f..9fff41ee98eb 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -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 */
--
2.21.0
