release-request-2950741e-0b24-4d70-b748-67bb2b5435b5-for-git_oc-mr1-release-43... (9467e2c5) · Commits · e / os / android_hardware_qcom_audio

hal/audio_hw.c

+31 −17

Original line number	Diff line number	Diff line
		@@ -1164,6 +1164,8 @@ int select_devices(struct audio_device *adev,
		audio_usecase_t hfp_ucid;
		struct listnode *node;
		int status = 0;
		struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
		USECASE_AUDIO_PLAYBACK_VOIP);

		usecase = get_usecase_from_list(adev, uc_id);
		if (usecase == NULL) {
		@@ -1207,8 +1209,6 @@ int select_devices(struct audio_device *adev,
		in_snd_device = SND_DEVICE_NONE;
		if (out_snd_device == SND_DEVICE_NONE) {
		struct stream_out *voip_out = adev->primary_output;
		struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
		USECASE_AUDIO_PLAYBACK_VOIP);

		out_snd_device = platform_get_output_snd_device(adev->platform,
		usecase->stream.out->devices);
		@@ -1360,6 +1360,12 @@ int select_devices(struct audio_device *adev,
		voice_set_sidetone(adev, out_snd_device, true);
		}

		if (usecase == voip_usecase) {
		struct stream_out *voip_out = voip_usecase->stream.out;
		audio_extn_utils_send_app_type_gain(adev,
		voip_out->app_type_cfg.app_type,
		&voip_out->app_type_cfg.gain[0]);
		}
		return status;
		}

		@@ -1946,9 +1952,6 @@ int start_output_stream(struct stream_out *out)
		register_out_stream(out);
		audio_extn_perf_lock_release();
		audio_extn_tfa_98xx_enable_speaker();
		audio_extn_utils_send_app_type_gain(out->dev,
		out->app_type_cfg.app_type,
		&out->app_type_cfg.gain[0]);

		// consider a scenario where on pause lower layers are tear down.
		// so on resume, swap mixer control need to be sent only when
		@@ -2672,20 +2675,35 @@ static ssize_t out_write(struct audio_stream_out stream, const void buffer,
		} else {
		error_code = ERROR_CODE_WRITE;
		if (out->pcm) {
		size_t bytes_to_write = bytes;

		if (out->muted)
		memset((void *)buffer, 0, bytes);
		// FIXME: this can be removed once audio flinger mixer supports mono output
		if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP) {
		size_t channel_count = audio_channel_count_from_out_mask(out->channel_mask);
		int16_t src = (int16_t )buffer;
		int16_t dst = (int16_t )buffer;

		LOG_ALWAYS_FATAL_IF(out->config.channels != 1 \|\| channel_count != 2 \|\|
		out->format != AUDIO_FORMAT_PCM_16_BIT,
		"out_write called for VOIP use case with wrong properties");

		ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);
		for (size_t i = 0; i < frames ; i++, dst++, src += 2) {
		*dst = (int16_t)(((int32_t)src[0] + (int32_t)src[1]) >> 1);
		}
		bytes_to_write /= 2;
		}
		ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes_to_write);

		long ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
		out->config.rate;
		long ns = (frames * NANOS_PER_SECOND) / out->config.rate;
		request_out_focus(out, ns);

		bool use_mmap = is_mmap_usecase(out->usecase) \|\| out->realtime;
		if (use_mmap)
		ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
		ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes_to_write);
		else
		ret = pcm_write(out->pcm, (void *)buffer, bytes);
		ret = pcm_write(out->pcm, (void *)buffer, bytes_to_write);

		release_out_focus(out, ns);
		} else {
		@@ -3803,17 +3821,13 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
		out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
		out->config = pcm_config_afe_proxy_playback;
		adev->voice_tx_output = out;
		} else if (out->flags == (AUDIO_OUTPUT_FLAG_DIRECT \|
		AUDIO_OUTPUT_FLAG_VOIP_RX)) {
		} else if (out->flags == AUDIO_OUTPUT_FLAG_VOIP_RX) {
		//FIXME: add support for MONO stream configuration when audioflinger mixer supports it
		uint32_t buffer_size, frame_size;
		out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_MONO;
		out->channel_mask = AUDIO_CHANNEL_OUT_MONO;
		out->usecase = USECASE_AUDIO_PLAYBACK_VOIP;
		out->config = pcm_config_voip;
		out->config.format = pcm_format_from_audio_format(config->format);
		out->config.rate = config->sample_rate;
		out->config.channels =
		audio_channel_count_from_out_mask(config->channel_mask);
		buffer_size = get_stream_buffer_size(VOIP_PLAYBACK_PERIOD_DURATION_MSEC,
		config->sample_rate,
		config->format,