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[13/14] ASoC: qdsp6: audioreach: Add MP3, AAC and FLAC compress format support

Message ID 20230201134947.1638197-14-quic_mohs@quicinc.com
State New
Headers show
Series Add support for compress offload and gapless playback. | expand

Commit Message

Mohammad Rafi Shaik Feb. 1, 2023, 1:49 p.m. UTC 
Add support for handling compressed formats such as MP3, AAC and FLAC.

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 | 106 ++++++++++++++++++++++++------
 1 file changed, 86 insertions(+), 20 deletions(-)
diff mbox series

Patch

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 7c45c36e9156..250ed828c7d3 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -852,6 +852,68 @@  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;
+	int ret = 0;
+
+	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 ret;
+}
+
 static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
 					   struct audioreach_module *module,
 					   struct audioreach_module_config *cfg)
@@ -1055,26 +1117,29 @@  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;
-
-	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;
-	}
+		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
+		audioreach_set_compr_media_format(header, p, mcfg);
 
 	rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);
 
@@ -1401,7 +1466,8 @@  int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_modu
 			cfg->channel_mapping[0] =  PCM_CHANNEL_L;
 			cfg->channel_mapping[1] =  PCM_CHANNEL_R;
 		}
-	}
+	} else
+		audioreach_set_compr_media_format(header, p, mcfg);
 
 	rc = gpr_send_port_pkt(graph->port, pkt);
 	kfree(pkt);