Merge "asoc: enable app type config for transcode loopback" (f1de3e72) · Commits · e / devices / android_kernel_fairphone_FP4

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

+146 −11

Original line number	Diff line number	Diff line
		@@ -40,6 +40,11 @@

		#define LOOPBACK_SESSION_MAX_NUM_STREAMS 2

		#define APP_TYPE_CONFIG_IDX_APP_TYPE 0
		#define APP_TYPE_CONFIG_IDX_ACDB_ID 1
		#define APP_TYPE_CONFIG_IDX_SAMPLE_RATE 2
		#define APP_TYPE_CONFIG_IDX_BE_ID 3

		static DEFINE_MUTEX(transcode_loopback_session_lock);

		struct trans_loopback_pdata {
		@@ -99,7 +104,7 @@ static void loopback_event_handler(uint32_t opcode,
		return;
		}

		cstream = trans->source.cstream;
		cstream = trans->sink.cstream;
		ac = trans->audio_client;

		/*
		@@ -111,7 +116,8 @@ static void loopback_event_handler(uint32_t opcode,
		switch (opcode) {
		case ASM_STREAM_CMD_ENCDEC_EVENTS:
		case ASM_IEC_61937_MEDIA_FMT_EVENT:
		pr_debug("%s: ASM_IEC_61937_MEDIA_FMT_EVENT\n", __func__);
		pr_debug("%s: Handling stream event : 0X%x\n",
		__func__, opcode);
		rtd = cstream->private_data;
		if (!rtd) {
		pr_err("%s: rtd is NULL\n", __func__);
		@@ -221,6 +227,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream)
		ret = -EINVAL;
		goto exit;
		}
		msm_adsp_init_mixer_ctl_pp_event_queue(rtd);
		}

		pr_debug("%s: num stream%d, stream name %s\n", __func__,
		@@ -235,8 +242,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream)
		}

		runtime->private_data = trans;
		if (trans->num_streams == 1)
		msm_adsp_init_mixer_ctl_pp_event_queue(rtd);

		exit:
		mutex_unlock(&trans->lock);
		return ret;
		@@ -281,14 +287,14 @@ static int msm_transcode_loopback_free(struct snd_compr_stream *cstream)
		trans->num_streams--;
		stop_transcoding(trans);

		if (cstream->direction == SND_COMPRESS_PLAYBACK)
		if (cstream->direction == SND_COMPRESS_PLAYBACK) {
		memset(&trans->sink, 0, sizeof(struct loopback_stream));
		else if (cstream->direction == SND_COMPRESS_CAPTURE)
		msm_adsp_clean_mixer_ctl_pp_event_queue(rtd);
		} else if (cstream->direction == SND_COMPRESS_CAPTURE) {
		memset(&trans->source, 0, sizeof(struct loopback_stream));
		}

		trans->session_state = LOOPBACK_SESSION_CLOSE;
		if (trans->num_streams == 1)
		msm_adsp_clean_mixer_ctl_pp_event_queue(rtd);
		mutex_unlock(&trans->lock);
		return ret;
		}
		@@ -448,17 +454,17 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream,
		trans->audio_client->perf_mode,
		trans->session_id,
		SNDRV_PCM_STREAM_CAPTURE,
		true);
		COMPRESSED_PASSTHROUGH_GEN);
		else
		msm_pcm_routing_reg_phy_stream(
		soc_pcm_tx->dai_link->id,
		trans->audio_client->perf_mode,
		trans->session_id,
		SNDRV_PCM_STREAM_CAPTURE);

		/* Opening Rx ADM in LOW_LATENCY mode by default */
		msm_pcm_routing_reg_phy_stream(
		soc_pcm_rx->dai_link->id,
		trans->audio_client->perf_mode,
		true,
		trans->session_id,
		SNDRV_PCM_STREAM_PLAYBACK);
		pr_debug("%s: Successfully opened ADM sessions\n", __func__);
		@@ -661,6 +667,67 @@ static int msm_transcode_rtic_event_ack_put(struct snd_kcontrol *kcontrol,
		return ret;
		}

		static int msm_transcode_playback_app_type_cfg_put(
		struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_value *ucontrol)
		{
		u64 fe_id = kcontrol->private_value;
		int session_type = SESSION_TYPE_RX;
		int be_id = ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID];
		struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000};
		int ret = 0;

		cfg_data.app_type = ucontrol->value.integer.value[
		APP_TYPE_CONFIG_IDX_APP_TYPE];
		cfg_data.acdb_dev_id = ucontrol->value.integer.value[
		APP_TYPE_CONFIG_IDX_ACDB_ID];
		if (ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] != 0)
		cfg_data.sample_rate = ucontrol->value.integer.value[
		APP_TYPE_CONFIG_IDX_SAMPLE_RATE];
		pr_debug("%s: fe_id %llu session_type %d be_id %d app_type %d acdb_dev_id %d sample_rate- %d\n",
		__func__, fe_id, session_type, be_id,
		cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
		ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type,
		be_id, &cfg_data);
		if (ret < 0)
		pr_err("%s: msm_transcode_playback_stream_app_type_cfg set failed returned %d\n",
		__func__, ret);

		return ret;
		}

		static int msm_transcode_playback_app_type_cfg_get(
		struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_value *ucontrol)
		{
		u64 fe_id = kcontrol->private_value;
		int session_type = SESSION_TYPE_RX;
		int be_id = 0;
		struct msm_pcm_stream_app_type_cfg cfg_data = {0};
		int ret = 0;

		ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type,
		&be_id, &cfg_data);
		if (ret < 0) {
		pr_err("%s: msm_transcode_playback_stream_app_type_cfg get failed returned %d\n",
		__func__, ret);
		goto done;
		}

		ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_APP_TYPE] =
		cfg_data.app_type;
		ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_ACDB_ID] =
		cfg_data.acdb_dev_id;
		ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] =
		cfg_data.sample_rate;
		ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID] = be_id;
		pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n",
		__func__, fe_id, session_type, be_id,
		cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
		done:
		return ret;
		}

		static int msm_transcode_stream_cmd_control(
		struct snd_soc_pcm_runtime *rtd)
		{
		@@ -862,6 +929,58 @@ static int msm_transcode_add_event_ack_cmd_control(
		return ret;
		}

		static int msm_transcode_app_type_cfg_info(struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_info *uinfo)
		{
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
		uinfo->count = 5;
		uinfo->value.integer.min = 0;
		uinfo->value.integer.max = 0xFFFFFFFF;
		return 0;
		}

		static int msm_transcode_add_app_type_cfg_control(
		struct snd_soc_pcm_runtime *rtd)
		{
		char mixer_str[32];
		struct snd_kcontrol_new fe_app_type_cfg_control[1] = {
		{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
		.info = msm_transcode_app_type_cfg_info,
		.put = msm_transcode_playback_app_type_cfg_put,
		.get = msm_transcode_playback_app_type_cfg_get,
		.private_value = 0,
		}
		};

		if (!rtd) {
		pr_err("%s NULL rtd\n", __func__);
		return -EINVAL;
		}

		if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) {
		snprintf(mixer_str, sizeof(mixer_str),
		"Audio Stream %d App Type Cfg",
		rtd->pcm->device);

		fe_app_type_cfg_control[0].name = mixer_str;
		fe_app_type_cfg_control[0].private_value = rtd->dai_link->id;

		fe_app_type_cfg_control[0].put =
		msm_transcode_playback_app_type_cfg_put;
		fe_app_type_cfg_control[0].get =
		msm_transcode_playback_app_type_cfg_get;

		pr_debug("Registering new mixer ctl %s", mixer_str);
		snd_soc_add_platform_controls(rtd->platform,
		fe_app_type_cfg_control,
		ARRAY_SIZE(fe_app_type_cfg_control));
		}

		return 0;
		}

		static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd)
		{
		int rc;
		@@ -885,6 +1004,11 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd)
		pr_err("%s: Could not add transcode event ack Control\n",
		__func__);

		rc = msm_transcode_add_app_type_cfg_control(rtd);
		if (rc)
		pr_err("%s: Could not add Compr App Type Cfg Control\n",
		__func__);

		return 0;
		}

		@@ -912,10 +1036,21 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform)
		return 0;
		}

		static int msm_transcode_loopback_remove(struct snd_soc_platform *platform)
		{
		struct trans_loopback_pdata *pdata = NULL;

		pdata = (struct trans_loopback_pdata *)
		snd_soc_platform_get_drvdata(platform);
		kfree(pdata);
		return 0;
		}

		static struct snd_soc_platform_driver msm_soc_platform = {
		.probe = msm_transcode_loopback_probe,
		.compr_ops = &msm_transcode_loopback_ops,
		.pcm_new = msm_transcode_loopback_new,
		.remove = msm_transcode_loopback_remove,
		};

		static int msm_transcode_dev_probe(struct platform_device *pdev)