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Commit 83338b5e authored by Viraja Kommaraju's avatar Viraja Kommaraju Committed by Hongtao Peng
Browse files

ASoC: dsp: find CAL block using buffer number 



Use FEdai_id to calculate the buffer number and
use the same to identify the right CAL block for
ASM CAL types.

Change-Id: Iab83892f5c2f2bc54c8fa97ac084d8a04586f712
Signed-off-by: default avatarViraja Kommaraju <virajak@codeaurora.org>
Signed-off-by: default avatarHongtao Peng <hongtaop@codeaurora.org>
parent 965a68a2

asoc/msm-compress-q6-v2.c

+8 −1
Original line number Diff line number Diff line 
// SPDX-License-Identifier: GPL-2.0-only

/* Copyright (c) 2012-2020, The Linux Foundation. All rights reserved.

/* Copyright (c) 2012-2021, The Linux Foundation. All rights reserved.

 */
@@ -1609,6 +1609,9 @@ static int msm_compr_configure_dsp_for_playback 
	else if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE)

		bits_per_sample = 32;



	ac->fedai_id = soc_prtd->dai_link->id;

	ac->stream_type = SNDRV_PCM_STREAM_PLAYBACK;



	if (prtd->compr_passthr != LEGACY_PCM) {

		ret = q6asm_open_write_compressed(ac, prtd->codec,

						  prtd->compr_passthr);
@@ -1764,6 +1767,9 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
		break;

	}



	prtd->audio_client->stream_type = SNDRV_PCM_STREAM_CAPTURE;

	prtd->audio_client->fedai_id = soc_prtd->dai_link->id;



	pr_debug("%s: stream_id %d bits_per_sample %d compr_passthr %d\n",

			__func__, ac->stream_id, bits_per_sample,

			prtd->compr_passthr);
@@ -3029,6 +3035,7 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) 
		pr_debug("%s: open_write stream_id %d bits_per_sample %d",

				__func__, stream_id, bits_per_sample);



		prtd->audio_client->fedai_id = (int)fe_id;

		if (q6core_get_avcs_api_version_per_service(

					APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=

					ADSP_ASM_API_VERSION_V2)

asoc/msm-pcm-loopback-v2.c

+2 −0
Original line number Diff line number Diff line
@@ -343,6 +343,8 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) 
			mutex_unlock(&pcm->lock);

			return -ENOMEM;

		}



		pcm->audio_client->fedai_id = rtd->dai_link->id;

		pcm->session_id = pcm->audio_client->session;

		pcm->audio_client->perf_mode = pdata->perf_mode;

		ret = q6asm_open_loopback_v2(pcm->audio_client,

asoc/msm-pcm-q6-noirq.c

+2 −1
Original line number Diff line number Diff line 
// SPDX-License-Identifier: GPL-2.0-only

/* Copyright (c) 2016-2020, The Linux Foundation. All rights reserved.

/* Copyright (c) 2016-2021, The Linux Foundation. All rights reserved.

 */



#include <linux/init.h>
@@ -367,6 +367,7 @@ static int msm_pcm_hw_params(struct snd_pcm_substream *substream, 
	config.bufsz = params_buffer_bytes(params) / params_periods(params);

	config.bufcnt = params_periods(params);



	prtd->audio_client->fedai_id = soc_prtd->dai_link->id;

	ret = q6asm_open_shared_io(prtd->audio_client, &config, dir,

				   use_default_chmap, chmap);

	if (ret) {

asoc/msm-pcm-q6-v2.c

+6 −0
Original line number Diff line number Diff line
@@ -369,6 +369,9 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream) 
	prtd->audio_client->perf_mode = pdata->perf_mode;

	pr_debug("%s: perf: %x\n", __func__, pdata->perf_mode);



	prtd->audio_client->stream_type = SNDRV_PCM_STREAM_PLAYBACK;

	prtd->audio_client->fedai_id = soc_prtd->dai_link->id;



	switch (params_format(params)) {

	case SNDRV_PCM_FORMAT_S32_LE:

		bits_per_sample = 32;
@@ -532,6 +535,9 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) 
				__func__, params_channels(params),

				prtd->audio_client->perf_mode);



		prtd->audio_client->stream_type = SNDRV_PCM_STREAM_CAPTURE;

		prtd->audio_client->fedai_id = soc_prtd->dai_link->id;



		if ((q6core_get_avcs_api_version_per_service(

				APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=

				ADSP_ASM_API_VERSION_V2))

asoc/msm-transcode-loopback-q6-v2.c

+1 −0
Original line number Diff line number Diff line
@@ -540,6 +540,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, 
						loopback_event_handler);

		trans->session_id = trans->audio_client->session;

		trans->audio_client->perf_mode = trans->sink.perf_mode;

		trans->audio_client->fedai_id = rtd->dai_link->id;

		ret = q6asm_open_transcode_loopback(trans->audio_client,

					bit_width,

					trans->source.codec_format,