@@ -1314,3 +1314,54 @@ int audioreach_enable_module(struct q6apm *apm, struct audioreach_module *module
return audioreach_send_u32_param(apm, module, PARAM_ID_MODULE_ENABLE, en);
}
EXPORT_SYMBOL_GPL(audioreach_enable_module);
+
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
+{
+ struct media_format *header;
+ struct gpr_pkt *pkt;
+ struct payload_media_fmt_pcm *cfg;
+ uint32_t num_channels = mcfg->num_channels;
+ 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;
+
+ if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
+ header->data_format = DATA_FORMAT_FIXED_POINT;
+ header->fmt_id = MEDIA_FMT_ID_PCM;
+ header->payload_size = sizeof(*cfg);
+
+ 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;
+ }
+ }
+
+ rc = gpr_send_port_pkt(graph->port, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(audioreach_compr_set_param);
@@ -786,4 +786,5 @@ int audioreach_remove_initial_silence(struct q6apm *apm, struct audioreach_modul
uint32_t initial_samples);
int audioreach_remove_trailing_silence(struct q6apm *apm, struct audioreach_module *module,
uint32_t trailing_samples);
+int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
#endif /* __AUDIOREACH_H__ */
@@ -146,6 +146,7 @@ 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 */