Merge "hal: get correspoding sample rate per device when sending cal data" (b0701bdc) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

hal/audio_extn/audio_extn.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -689,6 +689,8 @@ int audio_extn_utils_get_bit_width_from_string(const char *);
		int audio_extn_utils_get_sample_rate_from_string(const char *);
		int audio_extn_utils_get_channels_from_string(const char *);
		void audio_extn_utils_release_snd_device(snd_device_t snd_device);
		int audio_extn_utils_get_app_sample_rate_for_device(struct audio_device *adev,
		struct audio_usecase *usecase, int snd_device);

		#ifdef DS2_DOLBY_DAP_ENABLED
		#define LIB_DS2_DAP_HAL "vendor/lib/libhwdaphal.so"

hal/audio_extn/utils.c

+82 −80

Original line number	Diff line number	Diff line
		/*
		* Copyright (c) 2014-2018, The Linux Foundation. All rights reserved.
		* Copyright (c) 2014-2019, The Linux Foundation. All rights reserved.
		* Not a Contribution.
		*
		* Copyright (C) 2014 The Android Open Source Project
		@@ -892,6 +892,81 @@ void audio_extn_btsco_get_sample_rate(int snd_device, int *sample_rate)
		}
		}

		int audio_extn_utils_get_app_sample_rate_for_device(
		struct audio_device *adev,
		struct audio_usecase *usecase, int snd_device)
		{
		char value[PROPERTY_VALUE_MAX] = {0};
		int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

		if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) {
		property_get("vendor.audio.playback.mch.downsample",value,"");
		if (!strncmp("true", value, sizeof("true"))) {
		if ((popcount(usecase->stream.out->channel_mask) > 2) &&
		(usecase->stream.out->app_type_cfg.sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
		!(usecase->stream.out->flags &
		(audio_output_flags_t)AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		}

		if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
		usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
		} else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
		usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		} else if ((snd_device == SND_DEVICE_OUT_HDMI \|\|
		snd_device == SND_DEVICE_OUT_USB_HEADSET \|\|
		snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
		(usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
		/*
		* To best utlize DSP, check if the stream sample rate is supported/multiple of
		* configured device sample rate, if not update the COPP rate to be equal to the
		* device sample rate, else open COPP at stream sample rate
		*/
		platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
		usecase->stream.out->sample_rate,
		&usecase->stream.out->app_type_cfg.sample_rate);
		} else if (((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
		!audio_is_this_native_usecase(usecase)) &&
		usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) \|\|
		(usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
		/* Reset to default if no native stream is active*/
		usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		} else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
		/*
		* For a2dp playback get encoder sampling rate and set copp sampling rate,
		* for bit width use the stream param only.
		*/
		audio_extn_a2dp_get_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
		ALOGI("%s using %d sample rate rate for A2DP CoPP",
		__func__, usecase->stream.out->app_type_cfg.sample_rate);
		}
		audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
		sample_rate = usecase->stream.out->app_type_cfg.sample_rate;

		if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) \|\|
		(usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC) \|\|
		(usecase->stream.out->format == AUDIO_FORMAT_DOLBY_TRUEHD))
		&& audio_extn_passthru_is_passthrough_stream(usecase->stream.out)
		&& !audio_extn_passthru_is_convert_supported(adev, usecase->stream.out)) {
		sample_rate = sample_rate * 4;
		if (sample_rate > HDMI_PASSTHROUGH_MAX_SAMPLE_RATE)
		sample_rate = HDMI_PASSTHROUGH_MAX_SAMPLE_RATE;
		}
		} else if ((usecase->type == PCM_CAPTURE) && (usecase->stream.in != NULL)) {
		if (usecase->id == USECASE_AUDIO_RECORD_VOIP)
		usecase->stream.in->app_type_cfg.sample_rate = usecase->stream.in->sample_rate;
		if (voice_is_in_call_rec_stream(usecase->stream.in)) {
		audio_extn_btsco_get_sample_rate(usecase->in_snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
		} else {
		audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
		}
		sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
		} else if (usecase->type == TRANSCODE_LOOPBACK) {
		sample_rate = usecase->stream.inout->out_config.sample_rate;
		}
		return sample_rate;
		}

		static int send_app_type_cfg_for_device(struct audio_device *adev,
		struct audio_usecase *usecase,
		int split_snd_device)
		@@ -903,7 +978,6 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
		int pcm_device_id = 0, acdb_dev_id, app_type;
		int snd_device = split_snd_device, snd_device_be_idx = -1;
		int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		char value[PROPERTY_VALUE_MAX] = {0};
		struct streams_io_cfg *s_info = NULL;
		struct listnode *node = NULL;
		int bd_app_type = 0;
		@@ -978,51 +1052,9 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
		snd_device_be_idx);
		}

		if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) {

		property_get("vendor.audio.playback.mch.downsample",value,"");
		if (!strncmp("true", value, sizeof("true"))) {
		if ((popcount(usecase->stream.out->channel_mask) > 2) &&
		(usecase->stream.out->app_type_cfg.sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
		!(usecase->stream.out->flags &
		(audio_output_flags_t)AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		}

		if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
		usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
		} else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
		usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		} else if ((snd_device == SND_DEVICE_OUT_HDMI \|\|
		snd_device == SND_DEVICE_OUT_USB_HEADSET \|\|
		snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
		(usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
		/*
		* To best utlize DSP, check if the stream sample rate is supported/multiple of
		* configured device sample rate, if not update the COPP rate to be equal to the
		* device sample rate, else open COPP at stream sample rate
		*/
		platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
		usecase->stream.out->sample_rate,
		&usecase->stream.out->app_type_cfg.sample_rate);
		} else if (((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
		!audio_is_this_native_usecase(usecase)) &&
		usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) \|\|
		(usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
		/* Reset to default if no native stream is active*/
		usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		} else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
		/*
		* For a2dp playback get encoder sampling rate and set copp sampling rate,
		* for bit width use the stream param only.
		*/
		audio_extn_a2dp_get_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
		ALOGI("%s using %d sample rate rate for A2DP CoPP",
		__func__, usecase->stream.out->app_type_cfg.sample_rate);
		}
		audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
		sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
		sample_rate = audio_extn_utils_get_app_sample_rate_for_device(adev, usecase, snd_device);

		if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) {
		/* Interactive streams are supported with only direct app type id.
		* Get Direct profile app type and use it for interactive streams
		*/
		@@ -1039,16 +1071,6 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
		app_type = usecase->stream.out->app_type_cfg.app_type;
		app_type_cfg[len++] = app_type;
		app_type_cfg[len++] = acdb_dev_id;
		if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) \|\|
		(usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC) \|\|
		(usecase->stream.out->format == AUDIO_FORMAT_DOLBY_TRUEHD))
		&& audio_extn_passthru_is_passthrough_stream(usecase->stream.out)
		&& !audio_extn_passthru_is_convert_supported(adev, usecase->stream.out)) {

		sample_rate = sample_rate * 4;
		if (sample_rate > HDMI_PASSTHROUGH_MAX_SAMPLE_RATE)
		sample_rate = HDMI_PASSTHROUGH_MAX_SAMPLE_RATE;
		}
		app_type_cfg[len++] = sample_rate;

		if (snd_device_be_idx > 0)
		@@ -1061,14 +1083,6 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
		app_type = usecase->stream.in->app_type_cfg.app_type;
		app_type_cfg[len++] = app_type;
		app_type_cfg[len++] = acdb_dev_id;
		if (usecase->id == USECASE_AUDIO_RECORD_VOIP)
		usecase->stream.in->app_type_cfg.sample_rate = usecase->stream.in->sample_rate;
		if (voice_is_in_call_rec_stream(usecase->stream.in)) {
		audio_extn_btsco_get_sample_rate(usecase->in_snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
		} else {
		audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
		}
		sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
		app_type_cfg[len++] = sample_rate;
		if (snd_device_be_idx > 0)
		app_type_cfg[len++] = snd_device_be_idx;
		@@ -1077,7 +1091,6 @@ static int send_app_type_cfg_for_device(struct audio_device *adev,
		} else {
		app_type = platform_get_default_app_type_v2(adev->platform, usecase->type);
		if(usecase->type == TRANSCODE_LOOPBACK) {
		sample_rate = usecase->stream.inout->out_config.sample_rate;
		app_type = usecase->stream.inout->out_app_type_cfg.app_type;
		}
		app_type_cfg[len++] = app_type;
		@@ -1433,29 +1446,18 @@ void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
		int type = usecase->type;

		if (type == PCM_PLAYBACK && usecase->stream.out != NULL) {
		struct stream_out *out = usecase->stream.out;
		int snd_device = usecase->out_snd_device;
		snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
		platform_get_spkr_prot_snd_device(snd_device) : snd_device;
		platform_send_audio_calibration(adev->platform, usecase,
		out->app_type_cfg.app_type,
		usecase->stream.out->app_type_cfg.sample_rate);
		usecase->stream.out->app_type_cfg.app_type);
		} else if (type == PCM_CAPTURE && usecase->stream.in != NULL) {
		platform_send_audio_calibration(adev->platform, usecase,
		usecase->stream.in->app_type_cfg.app_type,
		usecase->stream.in->app_type_cfg.sample_rate);
		usecase->stream.in->app_type_cfg.app_type);
		} else if (type == PCM_HFP_CALL \|\| type == PCM_CAPTURE) {
		/* when app type is default. the sample rate is not used to send cal */
		platform_send_audio_calibration(adev->platform, usecase,
		platform_get_default_app_type_v2(adev->platform, usecase->type),
		48000);
		platform_get_default_app_type_v2(adev->platform, usecase->type));
		} else if (type == TRANSCODE_LOOPBACK && usecase->stream.inout != NULL) {
		int snd_device = usecase->out_snd_device;
		snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
		platform_get_spkr_prot_snd_device(snd_device) : snd_device;
		platform_send_audio_calibration(adev->platform, usecase,
		platform_get_default_app_type_v2(adev->platform, usecase->type),
		usecase->stream.inout->out_config.sample_rate);
		platform_get_default_app_type_v2(adev->platform, usecase->type));
		} else {
		/* No need to send audio calibration for voice and voip call usecases */
		if ((type != VOICE_CALL) && (type != VOIP_CALL))

hal/msm8916/platform.c

+9 −3

Original line number	Diff line number	Diff line
		@@ -3538,7 +3538,7 @@ int platform_get_backend_index(snd_device_t snd_device)
		}

		int platform_send_audio_calibration(void platform, struct audio_usecase usecase,
		int app_type, int sample_rate)
		int app_type)
		{
		struct platform_data my_data = (struct platform_data )platform;
		int acdb_dev_id, acdb_dev_type;
		@@ -3547,6 +3547,7 @@ int platform_send_audio_calibration(void platform, struct audio_usecase usecas
		int i, num_devices = 1;
		bool is_incall_rec_usecase = false;
		snd_device_t incall_rec_device;
		int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

		if (voice_is_in_call(my_data->adev))
		is_incall_rec_usecase = voice_is_in_call_rec_stream(usecase->stream.in);
		@@ -3576,11 +3577,16 @@ int platform_send_audio_calibration(void platform, struct audio_usecase usecas
		}

		for (i = 0; i < num_devices; i++) {
		if (!is_incall_rec_usecase)
		if (!is_incall_rec_usecase) {
		acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(new_snd_device[i])];
		else
		sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
		new_snd_device[i]);
		} else {
		// Use in_call_rec snd_device to extract the ACDB device ID instead of split snd devices
		acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(snd_device)];
		sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
		snd_device);
		}

		// Do not use Rx path default app type for TX path
		if ((usecase->type == PCM_CAPTURE) && (app_type == DEFAULT_APP_TYPE_RX_PATH)) {

hal/msm8960/platform.c

+1 −1

Original line number	Diff line number	Diff line
		@@ -510,7 +510,7 @@ int platform_get_default_app_type(void *platform __unused)
		}

		int platform_send_audio_calibration(void platform, struct audio_usecase usecase,
		int app_type __unused, int sample_rate __unused)
		int app_type __unused)
		{
		struct platform_data my_data = (struct platform_data )platform;
		int acdb_dev_id, acdb_dev_type;

hal/msm8974/platform.c

+9 −3

Original line number	Diff line number	Diff line
		@@ -3497,7 +3497,7 @@ int platform_get_backend_index(snd_device_t snd_device)
		}

		int platform_send_audio_calibration(void platform, struct audio_usecase usecase,
		int app_type, int sample_rate)
		int app_type)
		{
		struct platform_data my_data = (struct platform_data )platform;
		int acdb_dev_id, acdb_dev_type;
		@@ -3506,6 +3506,7 @@ int platform_send_audio_calibration(void platform, struct audio_usecase usecas
		int i, num_devices = 1;
		bool is_incall_rec_usecase = false;
		snd_device_t incall_rec_device;
		int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

		if (voice_is_in_call(my_data->adev))
		is_incall_rec_usecase = voice_is_in_call_rec_stream(usecase->stream.in);
		@@ -3540,11 +3541,16 @@ int platform_send_audio_calibration(void platform, struct audio_usecase usecas
		}

		for (i = 0; i < num_devices; i++) {
		if (!is_incall_rec_usecase)
		if (!is_incall_rec_usecase) {
		acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(new_snd_device[i])];
		else
		sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
		new_snd_device[i]);
		} else {
		// Use in_call_rec snd_device to extract the ACDB device ID instead of split snd devices
		acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(snd_device)];
		sample_rate = audio_extn_utils_get_app_sample_rate_for_device(my_data->adev, usecase,
		snd_device);
		}

		// Do not use Rx path default app type for TX path
		if ((usecase->type == PCM_CAPTURE) && (app_type == DEFAULT_APP_TYPE_RX_PATH)) {