webrtc initialize workflow

(conductor.cc:131): InitializePeerConnection
(create_peerconnection_factory.cc:46): CreatePeerConnectionFactory
(audio_processing_impl.cc:340): Create
(audio_processing_impl.cc:397): AudioProcessingImpl
(audio_processing_impl.cc:429): Capture analyzer activated: 0
(audio_processing_impl.cc:448): Initialize
(webrtc_video_engine.cc:480): WebRtcVideoEngine
(webrtc_voice_engine.cc:193): WebRtcVoiceEngine
(peer_connection_factory.cc:88): CreateModularPeerConnectionFactory
(peer_connection_factory.cc:119): PeerConnectionFactory
(peer_connection_factory.cc:160): Initialize
(channel_manager.cc:34): ChannelManager
(channel_manager.cc:119): Init
(webrtc_voice_engine.cc:218): Init

void WebRtcVoiceEngine::Init() {
	RTC_LOG(LS_INFO)<< __FUNCTION__;
	RTC_DCHECK(worker_thread_checker_.IsCurrent());


	// TaskQueue expects to be created/destroyed on the same thread.
	low_priority_worker_queue_.reset(
			new rtc::TaskQueue(
					task_queue_factory_->CreateTaskQueue("rtc-low-prio",
							webrtc::TaskQueueFactory::Priority::LOW)));

	// Load our audio codec lists.
	RTC_LOG(LS_INFO)<< "Supported send codecs in order of preference:";
	send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
	for (const AudioCodec& codec : send_codecs_) {
		RTC_LOG(LS_INFO)<< ToString(codec);
	}

	RTC_LOG(LS_INFO)<< "Supported recv codecs in order of preference:";
	recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
	for (const AudioCodec& codec : recv_codecs_) {
		RTC_LOG(LS_INFO)<< ToString(codec);
	}

#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
	// No ADM supplied? Create a default one.
	if (!adm_) {
		adm_ = webrtc::AudioDeviceModule::Create(
				webrtc::AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory_);
	}
#endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
	RTC_CHECK(adm());
	webrtc::adm_helpers::Init(adm());
	webrtc::apm_helpers::Init(apm());

	// Set up AudioState.
	{
		webrtc::AudioState::Config config;
		if (audio_mixer_) {
			config.audio_mixer = audio_mixer_;
		} else {
			config.audio_mixer = webrtc::AudioMixerImpl::Create();
		}
		config.audio_processing = apm_;
		config.audio_device_module = adm_;
		audio_state_ = webrtc::AudioState::Create(config);
	}

	// Connect the ADM to our audio path.
	adm()->RegisterAudioCallback(audio_state()->audio_transport());

	// Set default engine options.
	{
		AudioOptions options;
		options.echo_cancellation = true;
		options.auto_gain_control = true;
		options.noise_suppression = true;
		options.highpass_filter = true;
		options.stereo_swapping = false;
		options.audio_jitter_buffer_max_packets = 200;
		options.audio_jitter_buffer_fast_accelerate = false;
		options.audio_jitter_buffer_min_delay_ms = 0;
		options.audio_jitter_buffer_enable_rtx_handling = false;
		options.typing_detection = true;
		options.experimental_agc = false;
		options.extended_filter_aec = false;
		options.delay_agnostic_aec = false;
		options.experimental_ns = false;
		options.residual_echo_detector = true;
		bool error = ApplyOptions(options);
		RTC_DCHECK(error);
	}

	initialized_ = true;
}

create AudioDeviceModule，ADM负责采集audio

adm_ = webrtc::AudioDeviceModule::Create(
                webrtc::AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory_);

create AudioMixer，AMD负责做混音，接受的audio需要做混音

config.audio_mixer = webrtc::AudioMixerImpl::Create();

create AudioProcessing，APM负责语音处理

apm在create_peerconnection_factory.cc已经创建了

audio_processing = AudioProcessingBuilder().Create();

 

总结：CreatePeerConnectionFactory这个函数总共创建和初始化了PeerConnectionFactory，ChannelManager，MediaEngine，AudioState，以及AudioState中的AudioMixer，AudioProcessing，AudioDeviceModule