From: Mohammad Rafi Shaik quic_mohs@quicinc.com
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 --- 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 0acd4a75d5cd..6d0f4c8505f1 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 */