按照 WebRTC 的設計，AudioSource 是一個音頻源，它可以向外提供 PCM 音頻幀數據，比如麥克風錄制可以提供 PCM 音頻幀數據，它應該是一個 AudioSource。然而在 WebRTC 實際的實現中，AudioSource 的作用相對于設計，有一些不一樣的地方。

這里以 WebRTC 的示例應用 peerconnection_client 的代碼為例，來看 AudioSource/AudioTrack 的角色和作用。Conductor::InitializePeerConnection() 函數中初始化 PeerConnection 時會創建并添加 AudioSource/AudioTrack (webrtc/src/examples/peerconnection/client/conductor.cc)：

void Conductor::AddTracks() {if (!peer_connection_->GetSenders().empty()) {return; // Already added tracks.}rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(peer_connection_factory_->CreateAudioTrack(kAudioLabel, peer_connection_factory_->CreateAudioSource(cricket::AudioOptions())));auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId});if (!result_or_error.ok()) {RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: "<< result_or_error.error().message();}rtc::scoped_refptr<CapturerTrackSource> video_device =CapturerTrackSource::Create();if (video_device) {rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_(peer_connection_factory_->CreateVideoTrack(kVideoLabel, video_device));main_wnd_->StartLocalRenderer(video_track_);result_or_error = peer_connection_->AddTrack(video_track_, {kStreamId});if (!result_or_error.ok()) {RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: "<< result_or_error.error().message();}} else {RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed";}main_wnd_->SwitchToStreamingUI(); }

實際創建 LocalAudioSource 的調用棧如下：

#0 webrtc::LocalAudioSource::Create(cricket::AudioOptions const*) (audio_options=0x0) at ../../pc/local_audio_source.cc:20 #1 0x0000555556155b79 in webrtc::PeerConnectionFactory::CreateAudioSource(cricket::AudioOptions const&) (this=0x7fffe0004060, options=...)at ../../pc/peer_connection_factory.cc:182 #2 0x000055555617a478 in webrtc::ReturnType<rtc::scoped_refptr<webrtc::AudioSourceInterface> >::Invoke<webrtc::PeerConnectionFactoryInterface, rtc::scoped_refptr<webrtc::AudioSourceInterface> (webrtc::PeerConnectionFactoryInterface::*)(cricket::AudioOptions const&), cricket::AudioOptions const>(webrtc::PeerConnectionFactoryInterface*, rtc::scoped_refptr<webrtc::AudioSourceInterface> (webrtc::PeerConnectionFactoryInterface::*)(cricket::AudioOptions const&), cricket::AudioOptions const&&) (this=0x7fffffffba30, c=0x7fffe0004060, m=&virtual table offset 88) at ../../pc/proxy.h:105 #3 0x0000555556175131 in webrtc::MethodCall<webrtc::PeerConnectionFactoryInterface, rtc::scoped_refptr<webrtc::AudioSourceInterface>, cricket::AudioOptions const&>::Invoke<0ul>(std::integer_sequence<unsigned long, 0ul>) (this=0x7fffffffba10) at ../../pc/proxy.h:153 #4 0x0000555556183b9c in webrtc::MethodCall<webrtc::PeerConnectionFactoryInterface, rtc::scoped_refptr<webrtc::AudioSourceInterface>, cricket::AudioOptions const&>::Run() (this=0x7fffffffba10) at ../../pc/proxy.h:146 #5 0x00005555560575f4 in rtc::Thread::QueuedTaskHandler::OnMessage(rtc::Message*) (this=0x5555589fc7a8, msg=0x7fffef7fdab0)at ../../rtc_base/thread.cc:1042

LocalAudioSource 的頭文件 (webrtc/src/pc/local_audio_source.h) 如下：

namespace webrtc {class LocalAudioSource : public Notifier<AudioSourceInterface> {public:// Creates an instance of LocalAudioSource.static rtc::scoped_refptr<LocalAudioSource> Create(const cricket::AudioOptions* audio_options);SourceState state() const override { return kLive; }bool remote() const override { return false; }const cricket::AudioOptions options() const override { return options_; }void AddSink(AudioTrackSinkInterface* sink) override {}void RemoveSink(AudioTrackSinkInterface* sink) override {}protected:LocalAudioSource() {}~LocalAudioSource() override {}private:void Initialize(const cricket::AudioOptions* audio_options);cricket::AudioOptions options_; };} // namespace webrtc

LocalAudioSource 的源文件 (webrtc/src/pc/local_audio_source.cc) 如下：

using webrtc::MediaSourceInterface;namespace webrtc {rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(const cricket::AudioOptions* audio_options) {auto source = rtc::make_ref_counted<LocalAudioSource>();source->Initialize(audio_options);return source; }void LocalAudioSource::Initialize(const cricket::AudioOptions* audio_options) {if (!audio_options)return;options_ = *audio_options; }} // namespace webrtc

這里展示 LocalAudioSource 的頭文件和源文件是為了說明，LocalAudioSource 的功能真的是非常簡單，它基本上也就只能存一個 audio_options 而已。

接著來看下 LocalAudioSource 實現的相關接口 (webrtc/src/api/media_stream_interface.h)：

// Base class for sources. A MediaStreamTrack has an underlying source that // provides media. A source can be shared by multiple tracks. class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,public NotifierInterface {public:enum SourceState { kInitializing, kLive, kEnded, kMuted };virtual SourceState state() const = 0;virtual bool remote() const = 0;protected:~MediaSourceInterface() override = default; }; .......// Interface for receiving audio data from a AudioTrack. class AudioTrackSinkInterface {public:virtual void OnData(const void* audio_data,int bits_per_sample,int sample_rate,size_t number_of_channels,size_t number_of_frames) {RTC_NOTREACHED() << "This method must be overridden, or not used.";}// In this method, `absolute_capture_timestamp_ms`, when available, is// supposed to deliver the timestamp when this audio frame was originally// captured. This timestamp MUST be based on the same clock as// rtc::TimeMillis().virtual void OnData(const void* audio_data,int bits_per_sample,int sample_rate,size_t number_of_channels,size_t number_of_frames,absl::optional<int64_t> absolute_capture_timestamp_ms) {// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one// pure virtual.return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,number_of_frames);}// Returns the number of channels encoded by the sink. This can be less than// the number_of_channels if down-mixing occur. A value of -1 means an unknown// number.virtual int NumPreferredChannels() const { return -1; }protected:virtual ~AudioTrackSinkInterface() {} };class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {public:class AudioObserver {public:virtual void OnSetVolume(double volume) = 0;protected:virtual ~AudioObserver() {}};// TODO(deadbeef): Makes all the interfaces pure virtual after they're// implemented in chromium.// Sets the volume of the source. `volume` is in the range of [0, 10].// TODO(tommi): This method should be on the track and ideally volume should// be applied in the track in a way that does not affect clones of the track.virtual void SetVolume(double volume) {}// Registers/unregisters observers to the audio source.virtual void RegisterAudioObserver(AudioObserver* observer) {}virtual void UnregisterAudioObserver(AudioObserver* observer) {}// TODO(tommi): Make pure virtual.virtual void AddSink(AudioTrackSinkInterface* sink) {}virtual void RemoveSink(AudioTrackSinkInterface* sink) {}// Returns options for the AudioSource.// (for some of the settings this approach is broken, e.g. setting// audio network adaptation on the source is the wrong layer of abstraction).virtual const cricket::AudioOptions options() const; };

接口的設計反映了設計意圖，即 LocalAudioSource 應該是一個可以自己獲得或者生成音頻 PCM 數據，并吐出去的組件。

AudioTrack 用于將 AudioSource 接入 pipeline。如前面看到的， Conductor::AddTracks() 函數在創建了 AudioSource 之后，會立即創建 AudioTrack。創建 AudioTrack 的調用棧如下：

#0 webrtc::AudioTrack::Create(std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&, rtc::scoped_refptr<webrtc::AudioSourceInterface> const&)(id="Dp\033VUU\000\000jp\033VUU\000\000\270p\033VUU\000\000\376p\033VUU\000\000\362j\033VUU\000\000\006k\033VUU\000\000tB\002VUU\000\000\210B\002VUU\000\000\234B\002VUU\000\000Lk\033VUU\000\000`k\033VUU\000\000\032k\033VUU\000\000\070q\033VUU\000\000\320q\033VUU\000\000\226r\033VUU\000\000\370\377\377\377\377\377\377\377\070\067&XUU\000\000\303q\033VUU\000\000\211r\033VUU\000\000\364p\033VUU\000\000-q\033VUU\000\000\372\277qWUU\000\000\373\277qWUU\000\000\000\300qWUU\000\000\b\300qWUU\000\000"..., source=...) at ../../pc/audio_track.cc:21 #1 0x0000555556156c6f in webrtc::PeerConnectionFactory::CreateAudioTrack(std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&, webrtc::AudioSourceInterface*) (this=0x7fffe0004060, id="audio_label", source=0x7fffe000d4f0) at ../../pc/peer_connection_factory.cc:282 #2 0x000055555617a76c in webrtc::ReturnType<rtc::scoped_refptr<webrtc::AudioTrackInterface> >::Invoke<webrtc::PeerConnectionFactoryInterface, rtc::scoped_refptr<webrtc::AudioTrackInterface> (webrtc::PeerConnectionFactoryInterface::*)(std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&, webrtc::AudioSourceInterface*), std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const, webrtc::AudioSourceInterface*>(webrtc::PeerConnectionFactoryInterface*, rtc::scoped_refptr<webrtc::AudioTrackInterface> (webrtc::PeerConnectionFactoryInterface::*)(std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&, webrtc::AudioSourceInterface*), std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&&, webrtc::AudioSourceInterface*&&) (this=0x7fffffffc240, c=0x7fffe0004060, m=&virtual table offset 104) at ../../pc/proxy.h:105

AudioTrack 的完整代碼 (webrtc/src/pc/audio_track.cc) 如下：

namespace webrtc {// static rtc::scoped_refptr<AudioTrack> AudioTrack::Create(const std::string& id,const rtc::scoped_refptr<AudioSourceInterface>& source) {return rtc::make_ref_counted<AudioTrack>(id, source); }AudioTrack::AudioTrack(const std::string& label,const rtc::scoped_refptr<AudioSourceInterface>& source): MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {if (audio_source_) {audio_source_->RegisterObserver(this);OnChanged();} }AudioTrack::~AudioTrack() {RTC_DCHECK_RUN_ON(&thread_checker_);set_state(MediaStreamTrackInterface::kEnded);if (audio_source_)audio_source_->UnregisterObserver(this); }std::string AudioTrack::kind() const {return kAudioKind; }AudioSourceInterface* AudioTrack::GetSource() const {// Callable from any thread.return audio_source_.get(); }void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {RTC_DCHECK_RUN_ON(&thread_checker_);if (audio_source_)audio_source_->AddSink(sink); }void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {RTC_DCHECK_RUN_ON(&thread_checker_);if (audio_source_)audio_source_->RemoveSink(sink); }void AudioTrack::OnChanged() {RTC_DCHECK_RUN_ON(&thread_checker_);if (audio_source_->state() == MediaSourceInterface::kEnded) {set_state(kEnded);} else {set_state(kLive);} }} // namespace webrtc

Conductor::AddTracks() 還會將 AudioTrack 添加進 PeerConnection：

#0 webrtc::AudioRtpSender::AttachTrack() (this=0x0) at ../../pc/rtp_sender.cc:501 #1 0x0000555556d7bc40 in webrtc::RtpSenderBase::SetTrack(webrtc::MediaStreamTrackInterface*) (this=0x7fffe0014568, track=0x7fffe000ab70)at ../../pc/rtp_sender.cc:254 #2 0x0000555556203da4 in webrtc::ReturnType<bool>::Invoke<webrtc::RtpSenderInterface, bool (webrtc::RtpSenderInterface::*)(webrtc::MediaStreamTrackInterface*), webrtc::MediaStreamTrackInterface*>(webrtc::RtpSenderInterface*, bool (webrtc::RtpSenderInterface::*)(webrtc::MediaStreamTrackInterface*), webrtc::MediaStreamTrackInterface*&&) (this=0x7fffef7fca20, c=0x7fffe0014568, m=&virtual table offset 32) at ../../pc/proxy.h:105 #3 0x0000555556202fa7 in webrtc::MethodCall<webrtc::RtpSenderInterface, bool, webrtc::MediaStreamTrackInterface*>::Invoke<0ul>(std::integer_sequence<unsigned long, 0ul>) (this=0x7fffef7fca00) at ../../pc/proxy.h:153 #4 0x0000555556201061 in webrtc::MethodCall<webrtc::RtpSenderInterface, bool, webrtc::MediaStreamTrackInterface*>::Marshal(rtc::Location const&, rtc::Thread*) (this=0x7fffef7fca00, posted_from=..., t=0x55555852ef30) at ../../pc/proxy.h:136 #5 0x00005555561fe79a in webrtc::RtpSenderProxyWithInternal<webrtc::RtpSenderInternal>::SetTrack(webrtc::MediaStreamTrackInterface*)(this=0x7fffe0007e40, a1=0x7fffe000ab70) at ../../pc/rtp_sender_proxy.h:27 #6 0x0000555556d988ca in webrtc::RtpTransmissionManager::CreateSender(cricket::MediaType, std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&, rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, std::vector<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> >, std::allocator<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > > > const&, std::vector<webrtc::RtpEncodingParameters, std::allocator<webrtc::RtpEncodingParameters> > const&)(this=0x7fffe0014150, media_type=cricket::MEDIA_TYPE_AUDIO, id="audio_label", track=..., stream_ids=std::vector of length 1, capacity 1 = {...}, send_encodings=std::vector of length 0, capacity 0) at ../../pc/rtp_transmission_manager.cc:229 #7 0x0000555556d97fc5 in webrtc::RtpTransmissionManager::AddTrackUnifiedPlan(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, std::vector<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> >, std::allocator<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > > > const&) (this=0x7fffe0014150, track=..., stream_ids=std::vector of length 1, capacity 1 = {...}) at ../../pc/rtp_transmission_manager.cc:196 #8 0x0000555556d96232 in webrtc::RtpTransmissionManager::AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, std::vector<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> >, std::allocator<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > > > const&) (this=0x7fffe0014150, track=..., stream_ids=std::vector of length 1, capacity 1 = {...}) at ../../pc/rtp_transmission_manager.cc:112 #9 0x00005555561c0386 in webrtc::PeerConnection::AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, std::vector<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> >, std::allocator<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > > > const&)(this=0x7fffe0004590, track=..., stream_ids=std::vector of length 1, capacity 1 = {...}) at ../../pc/peer_connection.cc:834

RtpSenderBase::SetTrack() 在執行時，如果可以開始發送，會執行 SetSend()，否則不執行：

bool RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) {TRACE_EVENT0("webrtc", "RtpSenderBase::SetTrack");if (stopped_) {RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";return false;}if (track && track->kind() != track_kind()) {RTC_LOG(LS_ERROR) << "SetTrack with " << track->kind()<< " called on RtpSender with " << track_kind()<< " track.";return false;}// Detach from old track.if (track_) {DetachTrack();track_->UnregisterObserver(this);RemoveTrackFromStats();}// Attach to new track.bool prev_can_send_track = can_send_track();// Keep a reference to the old track to keep it alive until we call SetSend.rtc::scoped_refptr<MediaStreamTrackInterface> old_track = track_;track_ = track;if (track_) {track_->RegisterObserver(this);AttachTrack();}// Update channel.if (can_send_track()) {SetSend();AddTrackToStats();} else if (prev_can_send_track) {ClearSend();}attachment_id_ = (track_ ? GenerateUniqueId() : 0);return true; }

在真正需要開始啟動發送時，AudioRtpSender::SetSend() 會被調到，如 SDP offer 成功獲得了應答時：

#0 webrtc::AudioRtpSender::SetSend() (this=0x3000000020) at ../../pc/rtp_sender.cc:523 #1 0x0000555556d7c3c8 in webrtc::RtpSenderBase::SetSsrc(unsigned int) (this=0x7fffc8013ce8, ssrc=1129908154) at ../../pc/rtp_sender.cc:280 #2 0x000055555624c79a in webrtc::SdpOfferAnswerHandler::ApplyLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface, std::default_delete<webrtc::SessionDescriptionInterface> >, std::map<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> >, cricket::ContentGroup const*, std::less<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > >, std::allocator<std::pair<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const, cricket::ContentGroup const*> > > const&)(this=0x7fffc8004ae0, desc=std::unique_ptr<webrtc::SessionDescriptionInterface> = {...}, bundle_groups_by_mid=std::map with 2 elements = {...})at ../../pc/sdp_offer_answer.cc:1438 #3 0x000055555625112c in webrtc::SdpOfferAnswerHandler::DoSetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface, std::default_delete<webrtc::SessionDescriptionInterface> >, rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>)(this=0x7fffc8004ae0, desc=std::unique_ptr<webrtc::SessionDescriptionInterface> = {...}, observer=...) at ../../pc/sdp_offer_answer.cc:1934 #4 0x000055555624a0df in webrtc::SdpOfferAnswerHandler::<lambda(std::function<void()>)>::operator()(std::function<void()>)(__closure=0x7fffcfbf9f90, operations_chain_callback=...) at ../../pc/sdp_offer_answer.cc:1159 #5 0x00005555562816e9 in rtc::rtc_operations_chain_internal::OperationWithFunctor<webrtc::SdpOfferAnswerHandler::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*)::<lambda(std::function<void()>)> >::Run(void) (this=0x7fffc801ba30)at ../../rtc_base/operations_chain.h:71 #6 0x00005555562787e8 in rtc::OperationsChain::ChainOperation<webrtc::SdpOfferAnswerHandler::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*)::<lambda(std::function<void()>)> >(webrtc::SdpOfferAnswerHandler::<lambda(std::function<void()>)> &&)(this=0x7fffc8004e00, functor=...) at ../../rtc_base/operations_chain.h:154 #7 0x000055555624a369 in webrtc::SdpOfferAnswerHandler::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*)(this=0x7fffc8004ae0, observer=0x7fffc8018ae0, desc_ptr=0x7fffc801dc70) at ../../pc/sdp_offer_answer.cc:1143 #8 0x00005555561cb811 in webrtc::PeerConnection::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*)(this=0x7fffc8003d10, observer=0x7fffc8018ae0, desc_ptr=0x7fffc801dc70) at ../../pc/peer_connection.cc:1336 #9 0x0000555556178768 in webrtc::ReturnType<void>::Invoke<webrtc::PeerConnectionInterface, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>(webrtc::PeerConnectionInterface*, void (webrtc::PeerConnectionInterface::*)(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*), webrtc::SetSessionDescriptionObserver*&&, webrtc::SessionDescriptionInterface*&&) (this=0x7fffcfbfa3e0, c=0x7fffc8003d10, m=&virtual table offset 296) at ../../pc/proxy.h:119 #10 0x000055555617412f in webrtc::MethodCall<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::Invoke<0ul, 1ul>(std::integer_sequence<unsigned long, 0ul, 1ul>) (this=0x7fffcfbfa3c0) at ../../pc/proxy.h:153 #11 0x000055555616f59d in webrtc::MethodCall<webrtc::PeerConnectionInterface, void, webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*>::Marshal(rtc::Location const&, rtc::Thread*) (this=0x7fffcfbfa3c0, posted_from=..., t=0x555558758c00) at ../../pc/proxy.h:136 #12 0x0000555556167f64 in webrtc::PeerConnectionProxyWithInternal<webrtc::PeerConnectionInterface>::SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) (this=0x7fffc8004600, a1=0x7fffc8018ae0, a2=0x7fffc801dc70) at ../../pc/peer_connection_proxy.h:109 #13 0x00005555557a5692 in Conductor::OnSuccess(webrtc::SessionDescriptionInterface*) (this=0x5555589cf4e0, desc=0x7fffc801dc70)at ../../examples/peerconnection/client/conductor.cc:551 #14 0x00005555562848ef in webrtc::CreateSessionDescriptionObserverOperationWrapper::OnSuccess(webrtc::SessionDescriptionInterface*)(this=0x5555587591e0, desc=0x7fffc801dc70) at ../../pc/sdp_offer_answer.cc:845 #15 0x00005555562ced1f in webrtc::WebRtcSessionDescriptionFactory::OnMessage(rtc::Message*) (this=0x7fffc8005040, msg=0x7fffcfbfaab0)at ../../pc/webrtc_session_description_factory.cc:306 #16 0x0000555556055398 in rtc::Thread::Dispatch(rtc::Message*) (this=0x555558758c00, pmsg=0x7fffcfbfaab0) at ../../rtc_base/thread.cc:711

AudioRtpSender::SetSend() 的代碼如下：

void AudioRtpSender::SetSend() {RTC_DCHECK(!stopped_);RTC_DCHECK(can_send_track());if (!media_channel_) {RTC_LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";return;}cricket::AudioOptions options; #if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_WEBKIT_BUILD)// TODO(tommi): Remove this hack when we move CreateAudioSource out of// PeerConnection. This is a bit of a strange way to apply local audio// options since it is also applied to all streams/channels, local or remote.if (track_->enabled() && audio_track()->GetSource() &&!audio_track()->GetSource()->remote()) {options = audio_track()->GetSource()->options();} #endif// `track_->enabled()` hops to the signaling thread, so call it before we hop// to the worker thread or else it will deadlock.bool track_enabled = track_->enabled();bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {return voice_media_channel()->SetAudioSend(ssrc_, track_enabled, &options,sink_adapter_.get());});if (!success) {RTC_LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc_;} }

AudioRtpSender::SetSend() 獲得 AudioSource 的 audio options，檢查一下 AudioTrack 是否被 enable，然后把這些值連同 AudioSource 一起丟給 VoiceMediaChannel，如在 WebRTC 音頻發送和接收處理過程 中看到的，VoiceMediaChannel 實際為 WebRtcVoiceMediaChannel。WebRtcVoiceMediaChannel::SetAudioSend() 的代碼如下：

bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,bool enable,const AudioOptions* options,AudioSource* source) {RTC_DCHECK_RUN_ON(worker_thread_);// TODO(solenberg): The state change should be fully rolled back if any one of// these calls fail.if (!SetLocalSource(ssrc, source)) {return false;}if (!MuteStream(ssrc, !enable)) {return false;}if (enable && options) {return SetOptions(*options);}return true; }

上面看到的 SetLocalSource() 執行過程如下：

#0 webrtc::LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink*) (this=0x7fffc40ace80, sink=0x7fffcebf9700) at ../../pc/rtp_sender.cc:416 #1 0x00005555560f6140 in cricket::WebRtcVoiceMediaChannel::WebRtcAudioSendStream::SetSource(cricket::AudioSource*)(this=0x7fffc40ace30, source=0x7fffc8007068) at ../../media/engine/webrtc_voice_engine.cc:980 #2 0x00005555560e91fe in cricket::WebRtcVoiceMediaChannel::SetLocalSource(unsigned int, cricket::AudioSource*)(this=0x7fffc4091b90, ssrc=1129908154, source=0x7fffc8007068) at ../../media/engine/webrtc_voice_engine.cc:2084 #3 0x00005555560e6141 in cricket::WebRtcVoiceMediaChannel::SetAudioSend(unsigned int, bool, cricket::AudioOptions const*, cricket::AudioSource*)(this=0x7fffc4091b90, ssrc=1129908154, enable=true, options=0x7fffcfbf96d0, source=0x7fffc8007068) at ../../media/engine/webrtc_voice_engine.cc:1905 #4 0x0000555556d7e93f in webrtc::AudioRtpSender::<lambda()>::operator()(void) const (__closure=0x7fffcfbf96b0) at ../../pc/rtp_sender.cc:545

在這里搭建了完整的音頻數據處理管線。

這里再來看下把數據處理管線中兩個節點連接在一起的代碼。對于 AudioSource 和 LocalAudioSinkAdapter 的連接，代碼是：

void AudioRtpSender::AttachTrack() {RTC_DCHECK(track_);cached_track_enabled_ = track_->enabled();audio_track()->AddSink(sink_adapter_.get()); }

對于 LocalAudioSinkAdapter 和 WebRtcAudioSendStream 的連接，代碼是：

class WebRtcVoiceMediaChannel::WebRtcAudioSendStream: public AudioSource::Sink {public: . . . . . .void SetSource(AudioSource* source) {RTC_DCHECK_RUN_ON(&worker_thread_checker_);RTC_DCHECK(source);if (source_) {RTC_DCHECK(source_ == source);return;}source->SetSink(this);source_ = source;UpdateSendState();}

對于 WebRtcAudioSendStream 和 webrtc::AudioSendStream 的連接，代碼是：

class WebRtcVoiceMediaChannel::WebRtcAudioSendStream: public AudioSource::Sink {public: . . . . . .void OnData(const void* audio_data,int bits_per_sample,int sample_rate,size_t number_of_channels,size_t number_of_frames,absl::optional<int64_t> absolute_capture_timestamp_ms) override {RTC_DCHECK_EQ(16, bits_per_sample);RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);RTC_DCHECK(stream_);std::unique_ptr<webrtc::AudioFrame> audio_frame(new webrtc::AudioFrame());audio_frame->UpdateFrame(audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),number_of_frames, sample_rate, audio_frame->speech_type_,audio_frame->vad_activity_, number_of_channels);// TODO(bugs.webrtc.org/10739): add dcheck that// `absolute_capture_timestamp_ms` always receives a value.if (absolute_capture_timestamp_ms) {audio_frame->set_absolute_capture_timestamp_ms(*absolute_capture_timestamp_ms);}stream_->SendAudioData(std::move(audio_frame));}

不過我們前面看到 LocalAudioSource 明明不提供任何數據用于發送。這是怎么回事呢？

前面看到的數據處理管線是音頻的抽象數據處理管線，當我們需要自己定義一段音頻數據，通過 WebRTC 發送出去，比如從諸如 mp4 這樣的媒體文件解碼出來一段音頻數據，我們完全可以自己定義一個合適的 AudioSource 實現。但對于麥克風來說，音頻數據處理管線中，webrtc::AudioSendStream 之前的部分，是 AudioDeviceModule 和 AudioTransportImpl 這些組件，更詳細的數據處理管線形態，如 WebRTC 音頻發送和接收處理過程 。

回到 WebRtcVoiceMediaChannel::SetAudioSend() 的代碼，可見 AudioSource 確實有一個很重要的職責，就是傳遞 audio options，用戶通過 AudioSource 將 audio options 傳給 WebRtcVoiceEngine，來控制一些模塊的行為，如 APM 里面的回聲，降噪等。此外，還可以通過 AudioSource 控制音頻流發送的停止/重啟等。

總結

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