webrtc视频的采集，编码，发送流程详细分析（希望对底层深入了解的朋友有所帮助）

webrtc视频的采集，编码，发送流程详细分析（希望对底层深

⼊了解的朋友有所帮助）

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webrtc 视频的采集，编码，发送流程详细分析

我写⽂章⼀般是两个思路：

1. 下⼀步要调⽤什么对象的⽅法

2. 这⼀步的对象，怎么关联到下⼀步的对象的流程分析

这⼀步的流程主要阐述怎么关联下⼀步的对象的流程分析，当然这⼀步做了什么具体的⼯作，不能

详细展⽰，否则，太庞⼤了，需要各位朋友针对重点的部分，⾃⼰揣摩了。系统温度监控

//-----------------------------------------------------------------------------

// 视频编码器的创建以及与源的关联

标本缸

//-----------------------------------------------------------------------------

1. Java

PeerConnectionClient::createPeerConnectionInternal

peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels);

1.1. createVideoTrack 的建⽴

PeerConnectionClient::createVideoTrack(VideoCapturer capturer)

//---------------------------------------------------------

水力冲洗门

// 这个⾥⾯会在 Jni 层创建⼀个 AndroidVideoTrackSource 对象

//---------------------------------------------------------

videoSource = ateVideoSource(capturer.isScreencast());

//--------------------------------------------------------

// 这个会为 source 绑定⼀个 track 对象

/---------------------------------------------------------

localVideoTrack = ateVideoTrack(VIDEO_TRACK_ID, videoSource);

1.2

PeerConnectionFactory::createVideoTrack(String id, VideoSource source)

checkPeerConnectionFactoryExists();

return new VideoTrack(nativeCreateVideoTrack(nativeFactory, id, NativeVideoTrackSource()));

1.3

nativeCreateVideoTrack

1.4

Java_org_webrtc_PeerConnectionFactory_nativeCreateVideoTrack(JNIEnv* env, jclass jcaller, jlong factory, jstring id, jlong nativeVideoSource) { return JNI_PeerConnectionFactory_CreateVideoTr

ack(env, factory, base::android::JavaParamRef<jstring>(env, id), nativeVideoSource);

}

1.5 ./sdk/android/src/jni/pc/peer_

JNI_PeerConnectionFactory_CreateVideoTrack(JNIEnv* jni, jlong native_factory, const JavaParamRef<jstring>& id, jlong native_source) {

rtc::scoped_refptr<VideoTrackInterface> track = PeerConnectionFactoryFromJava(native_factory)->CreateVideoTrack(

JavaToStdString(jni, id), reinterpret_cast<VideoTrackSourceInterface*>(native_source));

return lease());

}

native_source 就是 AndroidVideoTrackSource

1.6 ./pc/peer_

rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(const std::string& id, VideoTrackSourceInterface* source) { RTC_DCHECK(signaling_thread_->IsCurrent());

rtc::scoped_refptr<VideoTrackInterface> track(VideoTrack::Create(id, source, worker_thread_));

return VideoTrackProxy::Create(signaling_thread_, worker_thread_, track);

}

1.7 ./pc/

rtc::scoped_refptr<VideoTrack> VideoTrack::Create(const std::string& id, VideoTrackSourceInterface* source, rtc::Thread* worker_thread) {

rtc::RefCountedObject<VideoTrack>* track = new rtc::RefCountedObject<VideoTrack>(id, source, worker_thread);

return track;

}

从上⾯流程看出 VideoTrack 包含了 AndroidVideoTrackSource

2. Java

PeerConnection::addTrack

RtpSender newSender = NativeMediaStreamTrack(), streamIds);

Java_org_webrtc_PeerConnection_nativeAddTrack

JNI_PeerConnection_AddTrack

4. ./sdk/android/src/jni/pc/

JNI_PeerConnection_AddTrack

ExtractNativePC(jni, j_pc)->AddTrack(reinterpret_cast<MediaStreamTrackInterface*>(native_track),

JavaListToNativeVector<std::string, jstring>(jni, j_stream_labels, &JavaToNativeString));

PeerConnection::AddTrack(rtc::scoped_refptr<MediaStreamTrackInterface> track, const std::vector<std::string>& stream_ids)

auto sender_or_error = (IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids) : AddTrackPlanB(track, stream_ids));

PeerConnection::AddTrackPlanB(rtc::scoped_refptr<MediaStreamTrackInterface> track, const std::vector<std::string>& stream_ids)

auto new_sender = CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});

if (track->kind() == MediaStreamTrackInterface::kAudioKind) {

new_sender->internal()->SetMediaChannel(voice_media_channel());

GetAudioTransceiver()->internal()->AddSender(new_sender);

const RtpSenderInfo* sender_info = FindSenderInfo(local_audio_sender_infos_, new_sender->internal()->stream_ids()[0], track->id()); if (sender_info) {

new_sender->internal()->SetSsrc(sender_info->first_ssrc);

}

} else {

RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());

new_sender->internal()->SetMediaChannel(video_media_channel());

GetVideoTransceiver()->internal()->AddSender(new_sender);

const RtpSenderInfo* sender_info = FindSenderInfo(local_video_sender_infos_, new_sender->internal()->stream_ids()[0], track->id()); if (sender_info) {

new_sender->internal()->SetSsrc(sender_info->first_ssrc);

}

return rtc::scoped_refptr<RtpSenderInterface>(new_sender);

PeerConnection::CreateSender(cricket::MediaType media_type, const std::string& id, rtc::scoped_refptr<MediaStreamTrackInterface> track, const std::vector<std::string>& stream_ids, const std::vector<RtpEncodingParameters>& send_encodings)

sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this)); NoteUsageEvent(UsageEvent::VIDEO_ADDED);

bool set_track_succeeded = sender->SetTrack(track); --->

RTC_DCHECK(set_track_succeeded);

sender->internal()->set_stream_ids(stream_ids);

sender->internal()->set_init_send_encodings(send_encodings);

VideoRtpSender::SetTrack(MediaStreamTrackInterface* track)

RtpSenderBase::SetTrack(MediaStreamTrackInterface* track)

SetSend();

VideoRtpSender::SetSend()

cricket::VideoOptions options;

VideoTrackSourceInterface* source = video_track()->GetSource();

if (source) {

options.is_screencast = source->is_screencast();

options.video_noise_reduction = source->needs_denoising();

}

switch (cached_track_content_hint_) {

case VideoTrackInterface::ContentHint::kNone:

break;

case VideoTrackInterface::ContentHint::kFluid:

options.is_screencast = false;气泡云

break;

case VideoTrackInterface::ContentHint::kDetailed:

case VideoTrackInterface::ContentHint::kText:

options.is_screencast = true;

break;

}

bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {

return video_media_channel()->SetVideoSend(ssrc_, &options, video_track());

10.

WebRtcVideoChannel::SetVideoSend(uint32_t ssrc, const VideoOptions* options, rtc::VideoSourceInterface<webrtc::VideoFrame>* source) const auto& kv = send_streams_.find(ssrc);

if (kv == send_streams_.end()) {

// Allow unknown ssrc only if source is null.

RTC_CHECK(source == nullptr);

RTC_LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;

return false;

}

return kv->second->SetVideoSend(options, source);

⽽ send_streams_ 就是 WebRtcVideoChannel::WebRtcVideoSendStream 参见下⾯的

《WebRtcVideoChannel 对象 AddSendStream (WebRtcVideoSendStream send_streams_) 的过程》

WebRtcVideoChannel::AddSendStream(const StreamParams& sp)11.

WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(const VideoOptions* options,

rtc::VideoSourceInterface<webrtc::VideoFrame>* source)

ReconfigureEncoder(); ---> 流程 12

source_ = source;

if (source && stream_) {

这段语句是把源与编码器对象关联到⼀起

stream_->SetSource(this, GetDegradationPreference());

}

stream_ 就是 VideoSendStream 对象，具体创建过程参见

void WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream() 中的

stream_ = call_->CreateVideoSendStream(std::move(config), parameters_.encoder_config.Copy());

stream_->SetSource 的接⼝继续分析

11.

11.1

void VideoSendStream::SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,

const DegradationPreference& degradation_preference) {

RTC_DCHECK_RUN_ON(&thread_checker_);

video_stream_encoder_->SetSource(source, degradation_preference);

}

11.2

void VideoStreamEncoder::SetSource(rtc::VideoSourceInterface<VideoFrame>* source,

const DegradationPreference& degradation_preference)

// 这个⾥⾯做 source 与 encoder 的关联

source_proxy_->SetSource(source, degradation_preference);

⽽ source_proxy_ 是 VideoStreamEncoder::VideoStreamEncoder 构造时创建的

./video/video_

VideoStreamEncoder::VideoStreamEncoder(

Clock* clock,

uint32_t number_of_cores,切削工具

VideoStreamEncoderObserver* encoder_stats_observer,

const VideoStreamEncoderSettings& settings,

std::unique_ptr<OveruseFrameDetector> overuse_detector,

TaskQueueFactory* task_queue_factory)

: source_proxy_(new VideoSourceProxy(this)),

VideoSourceProxy 的 video_stream_encoder_ 就是 VideoStreamEncoder

./video/video_

explicit VideoSourceProxy(VideoStreamEncoder* video_stream_encoder)

: video_stream_encoder_(video_stream_encoder),

degradation_preference_(DegradationPreference::DISABLED),

source_(nullptr),

max_framerate_(std::numeric_limits<int>::max()) {}

11.3 这个地⽅把 VideoStreamEncoder 和 Source 相关联

void VideoSourceProxy::SetSource(rtc::VideoSourceInterface<VideoFrame>* source,

const DegradationPreference& degradation_preference) {

// Called on libjingle's worker thread.

RTC_DCHECK_RUN_ON(&main_checker_);

rtc::VideoSourceInterface<VideoFrame>* old_source = nullptr;

rtc::VideoSinkWants wants;

{

rtc::CritScope lock(&crit_);

degradation_preference_ = degradation_preference;

old_source = source_;

source_ = source;

wants = GetActiveSinkWantsInternal();

}

if (old_source != source && old_source != nullptr) {

old_source->RemoveSink(video_stream_encoder_);

}

if (!source) {

return;

}

// source 就是上⾯设置过来的 WebRtcVideoChannel::WebRtcVideoSendStream 对象

source->AddOrUpdateSink(video_stream_encoder_, wants);

}

11.4

void WebRtcVideoChannel::WebRtcVideoSendStream::AddOrUpdateSink(

rtc::VideoSinkInterface<webrtc::VideoFrame>* sink, const rtc::VideoSinkWants& wants)

encoder_sink_ = sink;

source_->AddOrUpdateSink(encoder_sink_, wants);

11.5

void VideoTrack::AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink, const rtc::VideoSinkWants& wants) {

>亿万像素

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