Unverified Commit e41521b6 authored by Mohammad Rafi Shaik's avatar Mohammad Rafi Shaik Committed by Mark Brown
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ASoC: q6dsp: audioreach: Add support to set compress format params 



Add function for setting compress params.

Signed-off-by: default avatarMohammad Rafi Shaik <quic_mohs@quicinc.com>
Co-developed-by: default avatarSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: default avatarSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230619101653.9750-6-srinivas.kandagatla@linaro.org


Signed-off-by: default avatarMark Brown <broonie@kernel.org>
parent c7548f59

sound/soc/qcom/qdsp6/audioreach.c

+120 −19
Original line number Diff line number Diff line
@@ -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)
@@ -1037,6 +1130,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, 
	p = p + APM_MODULE_PARAM_DATA_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 = payload_size - sizeof(*header);
@@ -1051,12 +1145,19 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, 
		cfg->endianness = PCM_LITTLE_ENDIAN;

		cfg->num_channels = mcfg->num_channels;



	if (mcfg->num_channels == 1) {

		if (mcfg->num_channels == 1)

			cfg->channel_mapping[0] =  PCM_CHANNEL_L;

	} else if (num_channels == 2) {

		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);

sound/soc/qcom/qdsp6/audioreach.h

+28 −0
Original line number Diff line number Diff line
@@ -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__ */

sound/soc/qcom/qdsp6/q6apm-dai.c

+1 −0
Original line number Diff line number Diff line
@@ -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  */