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Commit bfef6ec1 authored by Linux Build Service Account's avatar Linux Build Service Account Committed by Gerrit - the friendly Code Review server
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Merge changes I2a570ec9,I563c59eb into audio-drivers.lnx.3.0 

* changes:
  asoc: sm8150: Add MultiMedia17 FE with capture support
  dsp: Add support for custom encoder
parents deb6842f 3f030f19

asoc/msm-compress-q6-v2.c

+25 −5
Original line number Diff line number Diff line
@@ -1432,6 +1432,7 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
	int dir = OUT, ret = 0;

	struct audio_client *ac = prtd->audio_client;

	uint32_t stream_index;

	uint32_t enc_cfg_id = ENC_CFG_ID_NONE;



	switch (prtd->codec_param.codec.format) {

	case SNDRV_PCM_FORMAT_S24_LE:
@@ -1450,6 +1451,9 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
	default:

		bits_per_sample = 16;

		sample_word_size = 16;

		if (prtd->codec == FORMAT_BESPOKE)

			enc_cfg_id =

			prtd->codec_param.codec.options.generic.reserved[0];

		break;

	}
@@ -1457,11 +1461,11 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
			__func__, ac->stream_id, bits_per_sample);



	if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) {

		ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,

			bits_per_sample, true);

		ret = q6asm_open_read_v4(prtd->audio_client, prtd->codec,

			bits_per_sample, true, enc_cfg_id);

	} else {

		ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,

			bits_per_sample, false);

		ret = q6asm_open_read_v4(prtd->audio_client, prtd->codec,

			bits_per_sample, false, enc_cfg_id);

	}

	if (ret < 0) {

		pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret);
@@ -1521,10 +1525,20 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
	pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",

			__func__, prtd->sample_rate, prtd->num_channels,

					 bits_per_sample, sample_word_size);

	if (prtd->codec == FORMAT_BESPOKE) {

		/*

		 * For BESPOKE codec, encoder specific config params are

		 * included as part of generic.

		 */

		ret = q6asm_enc_cfg_blk_custom(prtd->audio_client, prtd->sample_rate,

			prtd->num_channels, prtd->codec,

			(void *)&prtd->codec_param.codec.options.generic);

	} else {

		ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,

					prtd->sample_rate, prtd->num_channels,

					bits_per_sample, sample_word_size,

					ASM_LITTLE_ENDIAN, DEFAULT_QF);

	}



	return ret;

}
@@ -2043,6 +2057,12 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream, 
		break;

	}



	case SND_AUDIOCODEC_BESPOKE: {

		pr_debug("%s: SND_AUDIOCODEC_BESPOKE\n", __func__);

		prtd->codec = FORMAT_BESPOKE;

		break;

	}



	default:

		pr_err("codec not supported, id =%d\n", params->codec.id);

		return -EINVAL;

asoc/msm-pcm-q6-v2.c

+1 −1
Original line number Diff line number Diff line
@@ -490,7 +490,7 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
				prtd->audio_client->perf_mode);



		ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,

				bits_per_sample, false);

				bits_per_sample, false, ENC_CFG_ID_NONE);

		if (ret < 0) {

			pr_err("%s: q6asm_open_read failed\n", __func__);

			q6asm_audio_client_free(prtd->audio_client);

asoc/sm8150.c

+15 −0
Original line number Diff line number Diff line
@@ -5668,6 +5668,21 @@ static struct snd_soc_dai_link msm_common_misc_fe_dai_links[] = { 
		.codec_dai_name = "snd-soc-dummy-dai",

		.codec_name = "snd-soc-dummy",

	},

	{

		.name = "Compress Capture",

		.stream_name = "Compress9",

		.cpu_dai_name = "MultiMedia17",

		.platform_name = "msm-compress-dsp",

		.dynamic = 1,

		.dpcm_capture = 1,

		.trigger = {SND_SOC_DPCM_TRIGGER_POST,

			    SND_SOC_DPCM_TRIGGER_POST},

		.codec_dai_name = "snd-soc-dummy-dai",

		.codec_name = "snd-soc-dummy",

		.ignore_suspend = 1,

		.ignore_pmdown_time = 1,

		.id = MSM_FRONTEND_DAI_MULTIMEDIA17,

	},

};



static struct snd_soc_dai_link msm_common_be_dai_links[] = {

dsp/q6asm.c

+104 −6
Original line number Diff line number Diff line
@@ -46,6 +46,8 @@ 
#define FALSE       0x00

#define SESSION_MAX 8



#define ENC_FRAMES_PER_BUFFER 0x01



enum {

	ASM_TOPOLOGY_CAL = 0,

	ASM_CUSTOM_TOP_CAL,
@@ -2812,7 +2814,7 @@ EXPORT_SYMBOL(q6asm_set_soft_volume_module_instance_ids); 
static int __q6asm_open_read(struct audio_client *ac,

			     uint32_t format, uint16_t bits_per_sample,

			     uint32_t pcm_format_block_ver,

			     bool ts_mode)

			     bool ts_mode, uint32_t enc_cfg_id)

{

	int rc = 0x00;

	struct asm_stream_cmd_open_read_v3 open;
@@ -2888,6 +2890,12 @@ static int __q6asm_open_read(struct audio_client *ac, 
		open.mode_flags |= BUFFER_META_ENABLE;

		open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;

		break;

	case FORMAT_BESPOKE:

		open.mode_flags |= BUFFER_META_ENABLE;

		open.enc_cfg_id = enc_cfg_id;

		if (ts_mode)

			open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE;

		break;

	default:

		pr_err("%s: Invalid format 0x%x\n",

			__func__, format);
@@ -2939,7 +2947,7 @@ int q6asm_open_read(struct audio_client *ac, 
{

	return __q6asm_open_read(ac, format, 16,

				PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,

				false/*ts_mode*/);

				false/*ts_mode*/, ENC_CFG_ID_NONE);

}

EXPORT_SYMBOL(q6asm_open_read);
@@ -2948,7 +2956,7 @@ int q6asm_open_read_v2(struct audio_client *ac, uint32_t format, 
{

	return __q6asm_open_read(ac, format, bits_per_sample,

				 PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,

				 false/*ts_mode*/);

				 false/*ts_mode*/, ENC_CFG_ID_NONE);

}



/*
@@ -2963,7 +2971,7 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format, 
{

	return __q6asm_open_read(ac, format, bits_per_sample,

				 PCM_MEDIA_FORMAT_V3/*media fmt block ver*/,

				 false/*ts_mode*/);

				 false/*ts_mode*/, ENC_CFG_ID_NONE);

}

EXPORT_SYMBOL(q6asm_open_read_v3);
@@ -2976,11 +2984,12 @@ EXPORT_SYMBOL(q6asm_open_read_v3); 
 * @ts_mode: timestamp mode

 */

int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,

			uint16_t bits_per_sample, bool ts_mode)

			uint16_t bits_per_sample, bool ts_mode,

			uint32_t enc_cfg_id)

{

	return __q6asm_open_read(ac, format, bits_per_sample,

				 PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/,

				 ts_mode);

				 ts_mode, enc_cfg_id);

}

EXPORT_SYMBOL(q6asm_open_read_v4);
@@ -4287,6 +4296,95 @@ int q6asm_stream_run_nowait(struct audio_client *ac, uint32_t flags, 
	return __q6asm_run_nowait(ac, flags, msw_ts, lsw_ts, stream_id);

}



/**

 * q6asm_enc_cfg_blk_custom -

 *       command to set encode cfg block for custom

 *

 * @ac: Audio client handle

 * @sample_rate: Sample rate

 * @channels: number of ASM channels

 * @format: custom format flag

 * @cfg: generic encoder config

 *

 * Returns 0 on success or error on failure

 */

int q6asm_enc_cfg_blk_custom(struct audio_client *ac,

			uint32_t sample_rate, uint32_t channels,

			uint32_t format, void *cfg)

{

	struct asm_custom_enc_cfg_t_v2 enc_cfg;

	int rc = 0;

	uint32_t custom_size;

	struct snd_enc_generic *enc_generic = (struct snd_enc_generic *) cfg;



	custom_size = enc_generic->reserved[1];



	pr_debug("%s: session[%d] size[%d] res[2]=[%d] res[3]=[%d]\n",

		 __func__, ac->session, custom_size, enc_generic->reserved[2],

		 enc_generic->reserved[3]);



	pr_debug("%s: res[4]=[%d] sr[%d] ch[%d] format[%d]\n",

		 __func__, enc_generic->reserved[4], sample_rate,

		 channels, format);



	memset(&enc_cfg, 0, sizeof(struct asm_custom_enc_cfg_t_v2));

	q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);

	atomic_set(&ac->cmd_state, -1);



	enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;

	enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;

	enc_cfg.encdec.param_size = sizeof(struct asm_custom_enc_cfg_t_v2) -

				sizeof(struct asm_stream_cmd_set_encdec_param);

	enc_cfg.encblk.frames_per_buf = ENC_FRAMES_PER_BUFFER;

	enc_cfg.encblk.enc_cfg_blk_size  = enc_cfg.encdec.param_size -

				sizeof(struct asm_enc_cfg_blk_param_v2);



	enc_cfg.num_channels = channels;

	enc_cfg.sample_rate = sample_rate;



	if (q6asm_map_channels(enc_cfg.channel_mapping, channels, false)) {

		pr_err("%s: map channels failed %d\n",

		       __func__, channels);

		rc = -EINVAL;

		goto fail_cmd;

	}



	if (format == FORMAT_BESPOKE && custom_size &&

		custom_size <= sizeof(enc_cfg.custom_data)) {

		memcpy(enc_cfg.custom_data, &enc_generic->reserved[2],

			   custom_size);

		enc_cfg.custom_size = custom_size;

	}



	rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);

	if (rc < 0) {

		pr_err("%s: Comamnd %d failed %d\n",

		       __func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);

		rc = -EINVAL;

		goto fail_cmd;

	}

	rc = wait_event_timeout(ac->cmd_wait,

			(atomic_read(&ac->cmd_state) >= 0), 5*HZ);

	if (!rc) {

		pr_err("%s: timeout. waited for FORMAT_UPDATE\n",

			__func__);

		rc = -ETIMEDOUT;

		goto fail_cmd;

	}

	if (atomic_read(&ac->cmd_state) > 0) {

		pr_err("%s: DSP returned error[%s]\n",

					__func__, adsp_err_get_err_str(

					atomic_read(&ac->cmd_state)));

		rc = adsp_err_get_lnx_err_code(

				atomic_read(&ac->cmd_state));

		goto fail_cmd;

	}

	return 0;

fail_cmd:

	return rc;

}

EXPORT_SYMBOL(q6asm_enc_cfg_blk_custom);



/**

 * q6asm_enc_cfg_blk_aac -

 *       command to set encode cfg block for aac

include/dsp/apr_audio-v2.h

+16 −0
Original line number Diff line number Diff line
@@ -4743,6 +4743,22 @@ struct asm_enc_cfg_blk_param_v2 { 


} __packed;



struct asm_custom_enc_cfg_t_v2 {

	struct apr_hdr hdr;

	struct asm_stream_cmd_set_encdec_param encdec;

	struct asm_enc_cfg_blk_param_v2 encblk;

	uint32_t sample_rate;



	uint16_t num_channels;

	uint16_t reserved;

	/* num_ch == 1, then PCM_CHANNEL_C,

	 * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}

	 */

	uint8_t  channel_mapping[8];

	uint32_t  custom_size;

	uint8_t  custom_data[15];

} __packed;



/* @brief Dolby Digital Plus end point configuration structure

 */

struct asm_dec_ddp_endp_param_v2 {