Merge "hal: add support for callscreening." (62c0b82f) · Commits · e / os / android_hardware_qcom_audio

hal/audio_extn/audio_extn.c

+1 −1

Original line number	Diff line number	Diff line
		@@ -3665,7 +3665,7 @@ int audio_extn_set_device_cfg_params(struct audio_device *adev,
		/* Create an out stream to get snd device from audio device */
		reassign_device_list(&out.device_list, device_cfg_params->device, "");
		out.sample_rate = device_cfg_params->sample_rate;
		snd_device = platform_get_output_snd_device(adev->platform, &out);
		snd_device = platform_get_output_snd_device(adev->platform, &out, USECASE_TYPE_MAX);
		backend_idx = platform_get_backend_index(snd_device);

		ALOGV("%s:: device %d sample_rate %d snd_device %d backend_idx %d",

hal/audio_extn/audio_extn.h

+2 −1

Original line number	Diff line number	Diff line
		@@ -613,7 +613,8 @@ enum {
		typedef bool (fp_platform_is_edid_supported_format_t)(void, int);
		typedef int (fp_platform_set_device_params_t)(struct stream_out, int, int);
		typedef int (fp_platform_edid_get_max_channels_t)(void);
		typedef snd_device_t (fp_platform_get_output_snd_device_t)(void, struct stream_out*);
		typedef snd_device_t (fp_platform_get_output_snd_device_t)(void, struct stream_out*,
		usecase_type_t);
		typedef int (fp_platform_get_codec_backend_cfg_t)(struct audio_device,
		snd_device_t, struct audio_backend_cfg*);
		typedef bool (fp_platform_is_edid_supported_sample_rate_t)(void, int);

hal/audio_extn/passthru.c

+1 −1

Original line number	Diff line number	Diff line
		@@ -592,7 +592,7 @@ bool passthru_is_supported_backend_edid_cfg(struct audio_device *adev,
		snd_device_t out_snd_device = SND_DEVICE_NONE;
		int max_edid_channels = fp_platform_edid_get_max_channels(out->dev->platform);

		out_snd_device = fp_platform_get_output_snd_device(adev->platform, out);
		out_snd_device = fp_platform_get_output_snd_device(adev->platform, out, USECASE_TYPE_MAX);

		if (fp_platform_get_codec_backend_cfg(adev, out_snd_device, &backend_cfg)) {
		ALOGE("%s: ERROR: Unable to get current backend config!!!", __func__);

hal/audio_hw.c

+36 −21

Original line number	Diff line number	Diff line
		@@ -1810,7 +1810,8 @@ static void check_usecases_codec_backend(struct audio_device *adev,
		usecase->out_snd_device,
		platform_get_input_snd_device(
		adev->platform, NULL,
		&uc_info->device_list));
		&uc_info->device_list,
		usecase->type));
		enable_audio_route(adev, usecase);
		if (usecase->stream.out && usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
		out_set_voip_volume(&usecase->stream.out->stream,
		@@ -1863,9 +1864,12 @@ static void check_usecases_capture_codec_backend(struct audio_device *adev,
		/*
		* TODO: Enhance below condition to handle BT sco/USB multi recording
		*/
		if (usecase->type != PCM_PLAYBACK &&
		usecase != uc_info &&
		(usecase->in_snd_device != snd_device \|\| force_routing) &&

		bool capture_uc_needs_routing = usecase->type != PCM_PLAYBACK && (usecase != uc_info &&
		(usecase->in_snd_device != snd_device \|\| force_routing));
		bool call_proxy_snd_device = platform_is_call_proxy_snd_device(snd_device) \|\|
		platform_is_call_proxy_snd_device(usecase->in_snd_device);
		if (capture_uc_needs_routing && !call_proxy_snd_device &&
		((backend_check_cond &&
		(is_codec_backend_in_device_type(&usecase->device_list) \|\|
		(usecase->type == VOIP_CALL))) \|\|
		@@ -1875,7 +1879,7 @@ static void check_usecases_capture_codec_backend(struct audio_device *adev,
		platform_check_backends_match(snd_device,\
		usecase->in_snd_device))) &&
		(usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
		ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
		ALOGD("%s: Usecase (%s) is active on (%s) - disabling ..",
		__func__, use_case_table[usecase->id],
		platform_get_snd_device_name(usecase->in_snd_device));
		disable_audio_route(adev, usecase);
		@@ -1912,10 +1916,15 @@ static void check_usecases_capture_codec_backend(struct audio_device *adev,
		usecase->in_snd_device = snd_device;
		if (usecase->type != VOICE_CALL) {
		/* Update voc calibration before enabling VoIP route */
		if (usecase->type == VOIP_CALL)
		if (usecase->type == VOIP_CALL) {
		snd_device_t voip_snd_device;
		voip_snd_device = platform_get_output_snd_device(adev->platform,
		uc_info->stream.out,
		usecase->type);
		status = platform_switch_voice_call_device_post(adev->platform,
		platform_get_output_snd_device(adev->platform, uc_info->stream.out),
		voip_snd_device,
		usecase->in_snd_device);
		}
		enable_audio_route(adev, usecase);
		}
		}
		@@ -2586,10 +2595,11 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		uc_id);
		} else {
		out_snd_device = platform_get_output_snd_device(adev->platform,
		usecase->stream.out);
		usecase->stream.out, usecase->type);
		in_snd_device = platform_get_input_snd_device(adev->platform,
		NULL,
		&usecase->stream.out->device_list);
		&usecase->stream.out->device_list,
		usecase->type);
		}
		assign_devices(&usecase->device_list, &usecase->stream.out->device_list);
		} else if (usecase->type == TRANSCODE_LOOPBACK_RX) {
		@@ -2601,8 +2611,7 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		stream_out.sample_rate = usecase->stream.inout->out_config.sample_rate;
		stream_out.format = usecase->stream.inout->out_config.format;
		stream_out.channel_mask = usecase->stream.inout->out_config.channel_mask;
		out_snd_device = platform_get_output_snd_device(adev->platform,
		&stream_out);
		out_snd_device = platform_get_output_snd_device(adev->platform, &stream_out, usecase->type);
		assign_devices(&usecase->device_list,
		&usecase->stream.inout->out_config.device_list);
		} else if (usecase->type == TRANSCODE_LOOPBACK_TX ) {
		@@ -2610,7 +2619,7 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		ALOGE("%s: stream.inout is NULL", __func__);
		return -EINVAL;
		}
		in_snd_device = platform_get_input_snd_device(adev->platform, NULL, NULL);
		in_snd_device = platform_get_input_snd_device(adev->platform, NULL, NULL, usecase->type);
		assign_devices(&usecase->device_list,
		&usecase->stream.inout->in_config.device_list);
		} else {
		@@ -2647,7 +2656,7 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		voip_usecase->out_snd_device,
		platform_get_output_snd_device(
		adev->platform,
		usecase->stream.out));
		usecase->stream.out, usecase->type));
		}
		if ((voip_usecase) && (is_codec_backend_out_device_type(&voip_usecase->device_list) &&
		(is_codec_backend_out_device_type(&usecase->device_list) \|\|
		@@ -2679,7 +2688,8 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		out_snd_device = audio_extn_auto_hal_get_output_snd_device(adev, uc_id);
		else
		out_snd_device = platform_get_output_snd_device(adev->platform,
		usecase->stream.out);
		usecase->stream.out,
		usecase->type);
		voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);

		if (voip_usecase)
		@@ -2730,7 +2740,8 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)

		in_snd_device = platform_get_input_snd_device(adev->platform,
		priority_in,
		&out_devices);
		&out_devices,
		usecase->type);
		}
		}
		}
		@@ -2796,12 +2807,12 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		(usecase->type == VOIP_CALL)) &&
		(usecase->out_snd_device != SND_DEVICE_NONE)) {
		/* Disable sidetone only if voice/voip call already exists */
		if (voice_is_call_state_active(adev) \|\|
		if (voice_is_call_state_active_in_call(adev) \|\|
		voice_extn_compress_voip_is_started(adev))
		voice_set_sidetone(adev, usecase->out_snd_device, false);

		/* Disable aanc only if voice call exists */
		if (voice_is_call_state_active(adev))
		if (voice_is_call_state_active_in_call(adev))
		voice_check_and_update_aanc_path(adev, usecase->out_snd_device, false);
		}

		@@ -2899,11 +2910,11 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)

		if (usecase->type == VOICE_CALL \|\| usecase->type == VOIP_CALL) {
		/* Enable aanc only if voice call exists */
		if (voice_is_call_state_active(adev))
		if (voice_is_call_state_active_in_call(adev))
		voice_check_and_update_aanc_path(adev, out_snd_device, true);

		/* Enable sidetone only if other voice/voip call already exists */
		if (voice_is_call_state_active(adev) \|\|
		if (voice_is_call_state_active_in_call(adev) \|\|
		voice_extn_compress_voip_is_started(adev))
		voice_set_sidetone(adev, out_snd_device, true);
		}
		@@ -8816,9 +8827,13 @@ static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)

		pthread_mutex_lock(&adev->lock);
		if (adev->mode != mode) {
		ALOGD("%s: mode %d\n", __func__, mode);
		ALOGD("%s: mode %d , prev_mode %d \n", __func__, mode , adev->mode);
		adev->prev_mode = adev->mode; /* prev_mode is kept to handle voip concurrency*/
		adev->mode = mode;
		if (voice_is_in_call(adev) &&
		if( mode == AUDIO_MODE_CALL_SCREEN ){
		adev->current_call_output = adev->primary_output;
		voice_start_call(adev);
		} else if (voice_is_in_call_or_call_screen(adev) &&
		(mode == AUDIO_MODE_NORMAL \|\|
		(mode == AUDIO_MODE_IN_COMMUNICATION && !voice_is_call_state_active(adev)))) {
		list_for_each(node, &adev->usecase_list) {

hal/audio_hw.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -612,6 +612,7 @@ struct audio_device {
		pthread_mutex_t cal_lock;
		struct mixer *mixer;
		audio_mode_t mode;
		audio_mode_t prev_mode;
		audio_devices_t out_device;
		struct stream_out *primary_output;
		struct stream_out *voice_tx_output;

Original line number	Diff line number	Diff line
		@@ -613,7 +613,8 @@ enum {
		typedef bool (fp_platform_is_edid_supported_format_t)(void, int);
		typedef int (fp_platform_set_device_params_t)(struct stream_out, int, int);
		typedef int (fp_platform_edid_get_max_channels_t)(void);
		typedef snd_device_t (fp_platform_get_output_snd_device_t)(void, struct stream_out*);
		typedef snd_device_t (fp_platform_get_output_snd_device_t)(void, struct stream_out*,
		usecase_type_t);
		typedef int (fp_platform_get_codec_backend_cfg_t)(struct audio_device,
		snd_device_t, struct audio_backend_cfg*);
		typedef bool (fp_platform_is_edid_supported_sample_rate_t)(void, int);