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一分钟快速搭建 rtmpd 服务器: https://blog.csdn.net/freeabc/article/details/102880984
软件下载地址: http://www.qiyicc.com/download/rtmpd.rar
github 地址：https://github.com/superconvert/smart_rtmpd
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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。
//-----------------------------------------------------------------------------
//
// 视频编码器的创建以及与源的关联
//
//-----------------------------------------------------------------------------
1. Java
PeerConnectionClient::createPeerConnectionInternal
peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels);
1.1. createVideoTrack 的建立
PeerConnectionClient::createVideoTrack(VideoCapturer capturer)
localVideoTrack = factory.createVideoTrack(VIDEO_TRACK_ID, videoSource);
1.2
PeerConnectionFactory::createVideoTrack(String id, VideoSource source)
checkPeerConnectionFactoryExists();
return new VideoTrack(nativeCreateVideoTrack(nativeFactory, id, source.getNativeVideoTrackSource()));
1.3
nativeCreateVideoTrack
1.4
Java_org_webrtc_PeerConnectionFactory_nativeCreateVideoTrack(JNIEnv* env, jclass jcaller, jlong factory, jstring id, jlong nativeVideoSource) {
return JNI_PeerConnectionFactory_CreateVideoTrack(env, factory, base::android::JavaParamRef<jstring>(env, id), nativeVideoSource);
}
1.5 ./sdk/android/src/jni/pc/peer_connection_factory.cc
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 jlongFromPointer(track.release());
}
native_source 就是 AndroidVideoTrackSource
1.6 ./pc/peer_connection_factory.cc
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/video_track.cc
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 = nativeAddTrack(track.getNativeMediaStreamTrack(), streamIds);
3.
Java_org_webrtc_PeerConnection_nativeAddTrack
JNI_PeerConnection_AddTrack
4. ./sdk/android/src/jni/pc/peer_connection.cc
JNI_PeerConnection_AddTrack
ExtractNativePC(jni, j_pc)->AddTrack(reinterpret_cast<MediaStreamTrackInterface*>(native_track),
JavaListToNativeVector<std::string, jstring>(jni, j_stream_labels, &JavaToNativeString));
5.
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));
6.
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);
7.
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);
8.
VideoRtpSender::SetTrack(MediaStreamTrackInterface* track)
RtpSenderBase::SetTrack(MediaStreamTrackInterface* track)
SetSend();
9.
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;
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();
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
这段语句是把源与编码器对象关联到一起!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
source_ = source;
if (source && stream_) {
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());
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)
source_proxy_->SetSource(source, degradation_preference);
而 source_proxy_ 是 VideoStreamEncoder::VideoStreamEncoder 构造时
source_proxy_(new VideoSourceProxy(this)),
对应的构造函数 video_stream_encoder_ 就是 VideoStreamEncoder
explicit VideoSourceProxy(VideoStreamEncoder* video_stream_encoder)
: video_stream_encoder_(video_stream_encoder),
degradation_preference_(DegradationPreference: ISABLED),
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 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) {
RTC_DCHECK(worker_thread_->IsCurrent());
VideoSourceBase::AddOrUpdateSink(sink, wants);
rtc::VideoSinkWants modified_wants = wants;
modified_wants.black_frames = !enabled();
video_source_->AddOrUpdateSink(sink, modified_wants);
}
video_source_ 就是 AndroidVideoTrackSource 对象
11.5
(void AndroidVideoTrackSource::AddOrUpdateSink)
void AdaptedVideoTrackSource::AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink, const rtc::VideoSinkWants& wants) {
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
就这个地方，把视频源对象 AndroidVideoTrackSource 与 视频编码对象 VideoStreamEncoder 关联一起的
broadcaster_.AddOrUpdateSink(sink, wants);
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
OnSinkWantsChanged(broadcaster_.wants());
}
12.
WebRtcVideoChannel::WebRtcVideoSendStream::ReconfigureEncoder()
webrtc::VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(codec_settings.codec);
stream_->ReconfigureVideoEncoder(encoder_config.Copy());
而 stream_ 是由下面流程产生的
12.1
WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream()
stream_ = call_->CreateVideoSendStream(std::move(config), parameters_.encoder_config.Copy());
12.2
webrtc::VideoSendStream* Call::CreateVideoSendStream(webrtc::VideoSendStream::Config config, VideoEncoderConfig encoder_config)
std::unique_ptr<FecController> fec_controller = config_.fec_controller_factory ?
config_.fec_controller_factory->CreateFecController() : std::make_unique<FecControllerDefault>(clock_);
return CreateVideoSendStream(std::move(config), std::move(encoder_config), std::move(fec_controller));
12.3
webrtc::VideoSendStream* Call::CreateVideoSendStream(webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config, std::unique_ptr<FecController> fec_controller)
VideoSendStream* send_stream = new VideoSendStream(clock_, num_cpu_cores_, module_process_thread_.get(),
task_queue_factory_, call_stats_.get(), transport_send_ptr_, bitrate_allocator_.get(),
video_send_delay_stats_.get(), event_log_, std::move(config), std::move(encoder_config),
suspended_video_send_ssrcs_, suspended_video_payload_states_, std::move(fec_controller));
13.
VideoSendStream::ReconfigureVideoEncoder
video_stream_encoder_->ConfigureEncoder(std::move(config), config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp)); --->
video_stream_encoder_ 是由就是 VideoStreamEncoder
VideoSendStream::VideoSendStream(Clock* clock, int num_cpu_cores, ProcessThread* module_process_thread, TaskQueueFactory* task_queue_factory,
CallStats* call_stats, RtpTransportControllerSendInterface* transport, BitrateAllocatorInterface* bitrate_allocator, SendDelayStats* send_delay_stats,
RtcEventLog* event_log, VideoSendStream::Config config, VideoEncoderConfig encoder_config, const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states, std::unique_ptr<FecController> fec_controller)
video_stream_encoder_ = CreateVideoStreamEncoder(clock, task_queue_factory, num_cpu_cores, &stats_proxy_, config_.encoder_settings);
14.
VideoStreamEncoder::ConfigureEncoder(VideoEncoderConfig config, size_t max_data_payload_length)
ReconfigureEncoder(); --->
15.
VideoStreamEncoder::ReconfigureEncoder()
std::vector<VideoStream> streams =
encoder_config_.video_stream_factory->CreateEncoderStreams(last_frame_info_->width, last_frame_info_->height, encoder_config_);
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
这个地方就是创建了编码器对象
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
encoder_ = settings_.encoder_factory->CreateVideoEncoder(encoder_config_.video_format); --->
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
这个函数还有对 encoder_ 的初始化工作 参见下面的视频编码器的初始化
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
if (encoder_->InitEncode(&send_codec_, VideoEncoder::Settings(settings_.capabilities, number_of_cores_, max_data_payload_length)) != 0)
16.
VideoEncoderFactoryWrapper::CreateVideoEncoder(const SdpVideoFormat& format)
ScopedJavaLocalRef<jobject> encoder = Java_VideoEncoderFactory_createEncoder(jni, encoder_factory_, j_codec_info); --->
return JavaToNativeVideoEncoder(jni, encoder);
17.
17.1
static base::android::ScopedJavaLocalRef<jobject> Java_VideoEncoderFactory_createEncoder(JNIEnv*
env, const base::android::JavaRef<jobject>& obj, const base::android::JavaRef<jobject>& info) {
jclass clazz = org_webrtc_VideoEncoderFactory_clazz(env);
CHECK_CLAZZ(env, obj.obj(), org_webrtc_VideoEncoderFactory_clazz(env), NULL);
jni_generator::JniJavaCallContextChecked call_context;
call_context.Init<base::android::MethodID::TYPE_INSTANCE>(env, clazz,
"createEncoder",
"(Lorg/webrtc/VideoCodecInfo;)Lorg/webrtc/VideoEncoder;",
&g_org_webrtc_VideoEncoderFactory_createEncoder);
jobject ret = env->CallObjectMethod(obj.obj(), call_context.base.method_id, info.obj());
return base::android::ScopedJavaLocalRef<jobject>(env, ret);
}
17.2
JavaToNativeVideoEncoder(JNIEnv* jni, const JavaRef<jobject>& j_encoder) {
const jlong native_encoder = Java_VideoEncoder_createNativeVideoEncoder(jni, j_encoder);
VideoEncoder* encoder;
if (native_encoder == 0) {
encoder = new VideoEncoderWrapper(jni, j_encoder);
} else {
encoder = reinterpret_cast<VideoEncoder*>(native_encoder);
}
return std::unique_ptr<VideoEncoder>(encoder);
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
VideoEncoderWrapper 被创建
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
0. PeerConnectionFactory
包含 rtc::Thread* network_thread_;
包含 rtc::Thread* worker_thread_;
包含 rtc::Thread* signaling_thread_;
包含 std::unique_ptr<rtc::Thread> owned_network_thread_;
包含 std::unique_ptr<rtc::Thread> owned_worker_thread_;
包含 const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
包含 Options options_;
包含 std::unique_ptr<cricket::ChannelManager> channel_manager_; 通道管理
包含 std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
包含 std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
包含 std::unique_ptr<cricket::MediaEngineInterface> media_engine_; 媒体引擎
包含 std::unique_ptr<webrtc::CallFactoryInterface> call_factory_; call 工厂
包含 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory_;
包含 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
包含 std::unique_ptr<NetworkStatePredictorFactoryInterface>
包含 network_state_predictor_factory_;
包含 std::unique_ptr<NetworkControllerFactoryInterface>
包含 injected_network_controller_factory_;
//------------------------------------------------------------------------------------------------------
包含 std::unique_ptr<MediaTransportFactory> media_transport_factory_;
这个传到 PeerConnection 里的 JsepTransportController
JsepTransportController::MaybeCreateDatagramTransport
auto datagram_transport_result = config_.media_transport_factory->CreateDatagramTransport(
network_thread_, settings);
//------------------------------------------------------------------------------------------------------
包含 std::unique_ptr<NetEqFactory> neteq_factory_;
包含 const std::unique_ptr<WebRtcKeyValueConfig> trials_;
1. internal::Call
PeerConnectionFactory::CreatePeerConnection 会产生一个 call 对象，并把 call 对象赋值给 PeerConnection
包含 TaskQueueFactory* const task_queue_factory_ , 这个就是 PeerConnectionFactory 的 task_queue_factory_
包含 const std::unique_ptr< rocessThread> module_process_thread_;
包含 const std::unique_ptr<CallStats> call_stats_;
包含 const std::unique_ptr<BitrateAllocator> bitrate_allocator_, bitrate_allocator_(new BitrateAllocator(this))
包含 Call::Config config_;
包含 SequenceChecker configuration_sequence_checker_;
包含 SequenceChecker worker_sequence_checker_;
包含 NetworkState audio_network_state_;
包含 NetworkState video_network_state_;
包含 bool aggregate_network_up_ RTC_GUARDED_BY(configuration_sequence_checker_);
包含 std::unique_ptr<RWLockWrapper> receive_crit_;
// Audio, Video, and FlexFEC receive streams are owned by the client that
// creates them.
包含 std::set<AudioReceiveStream*> audio_receive_streams_ RTC_GUARDED_BY(receive_crit_);
包含 std::set<VideoReceiveStream*> video_receive_streams_ RTC_GUARDED_BY(receive_crit_);
包含 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ RTC_GUARDED_BY(receive_crit_);
// TODO(nisse): Should eventually be injected at creation,
// with a single object in the bundled case.
包含 RtpStreamReceiverController audio_receiver_controller_;
包含 RtpStreamReceiverController video_receiver_controller_;
包含 std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_ RTC_GUARDED_BY(receive_crit_);
包含 std::unique_ptr<RWLockWrapper> send_crit_;
// Audio and Video send streams are owned by the client that creates them.
包含 std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ RTC_GUARDED_BY(send_crit_);
包含 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ RTC_GUARDED_BY(send_crit_);
-------------------------------------------------------------------------------------------------------
包含 std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(send_crit_);
这个 Call::CreateVideoSendStream 接口内
VideoSendStream* send_stream = new VideoSendStream
video_send_streams_.insert(send_stream);
-------------------------------------------------------------------------------------------------------
包含 RtpStateMap suspended_audio_send_ssrcs_ RTC_GUARDED_BY(configuration_sequence_checker_);
包含 RtpStateMap suspended_video_send_ssrcs_ RTC_GUARDED_BY(configuration_sequence_checker_);
包含 RtpPayloadStateMap suspended_video_payload_states_ RTC_GUARDED_BY(configuration_sequence_checker_);
包含 webrtc::RtcEventLog* event_log_;
// The following members are only accessed (exclusively) from one thread and
// from the destructor, and therefore doesn't need any explicit
// synchronization.
包含 RateCounter received_bytes_per_second_counter_;
包含 RateCounter received_audio_bytes_per_second_counter_;
包含 RateCounter received_video_bytes_per_second_counter_;
包含 RateCounter received_rtcp_bytes_per_second_counter_;
包含 absl: ptional<int64_t> first_received_rtp_audio_ms_;
包含 absl:ptional<int64_t> last_received_rtp_audio_ms_;
包含 absl:ptional<int64_t> first_received_rtp_video_ms_;
包含 absl:ptional<int64_t> last_received_rtp_video_ms_;
包含 rtc::CriticalSection last_bandwidth_bps_crit_;
包含 uint32_t last_bandwidth_bps_ RTC_GUARDED_BY(&last_bandwidth_bps_crit_);
// TODO(holmer): Remove this lock once BitrateController no longer calls
// OnNetworkChanged from multiple threads.
包含 rtc::CriticalSection bitrate_crit_;
包含 uint32_t min_allocated_send_bitrate_bps_ RTC_GUARDED_BY(&worker_sequence_checker_);
包含 uint32_t configured_max_padding_bitrate_bps_ RTC_GUARDED_BY(&bitrate_crit_);
包含 AvgCounter estimated_send_bitrate_kbps_counter_ RTC_GUARDED_BY(&bitrate_crit_);
包含 AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(&bitrate_crit_);
包含 ReceiveSideCongestionController receive_side_cc_;
包含 const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_;
包含 const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
包含 const int64_t start_ms_;
// Caches transport_send_.get(), to avoid racing with destructor.
// Note that this is declared before transport_send_ to ensure that it is not
// invalidated until no more tasks can be running on the transport_send_ task
// queue.
------------------------------------------------------------------------
包含 RtpTransportControllerSendInterface* const transport_send_ptr_;
// Declared last since it will issue callbacks from a task queue. Declaring it
// last ensures that it is destroyed first and any running tasks are finished.
包含 std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
这个就是 RtpTransportControllerSend 对象
----------------------------------------------------------------------
包含 bool is_target_rate_observer_registered_ RTC_GUARDED_BY(&configuration_sequence_checker_) = false;
2. WebRtcVideoChannel 这个对象最终会 PeerConnection::RtpTransceiver transceivers_ 设置到这个里面，这个对象时负责收发的管理
PeerConnection::ApplyLocalDescription --->
PeerConnection::CreateChannels --->
PeerConnection::CreateVideoChannel --->
ChannelManager::CreateVideoChannel --->
WebRtcVideoEngine::CreateMediaChannel 这个里面分配的 WebRtcVideoChannel 对象
包含 WebRtcVideoChannel::WebRtcVideoSendStream 对象 send_streams_
包含 WebRtcVideoChannel::WebRtcVideoReceiveStream 对象 receive_streams_
包含 VideoEncoderFactory * 对象 encoder_factory_
包含 VideoDecoderFactory * 对象 decoder_factory_
包含 VideoBitrateAllocatorFactory * 对象 bitrate_allocator_factory_
包含 std::set<uint32_t> 发送源 send_ssrcs_
包含 std::set<uint32_t> 接收源 receive_ssrcs_
包含 absl:ptional<VideoCodecSettings> 发送编码 send_codec_
包含 std::vector<VideoCodecSettings> 接收编码 recv_codecs_
包含 std::vector<VideoCodecSettings> 协商编码 negotiated_codecs_
包含 VideoSendParameters 视频发送参数 send_params_
包含 VideoRecvParameters 视频接收参数 recv_params_
包含 webrtc::Call* const 对象指针 call_
3. WebRtcVideoChannel::WebRtcVideoSendStream 包含 VideoSendStream 对象 stream_
包含 rtc::VideoSourceInterface<webrtc::VideoFrame>* 对象 source_，其实就是一个 VideoTrack,
VideoRtpSender::SetSend
WebRtcVideoChannel::SetVideoSend
WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend 设置过来的, 就是 VideoRtpSender 的 track_
这个过程绑定了 stream_ 与 track_ 的 pipeline, stream_ 内部通过 VideoStreamEncoder 实现的
这样采集的视频就可以传递到编码器中了
VideoRtpSender 的 track_ 是通过
JNI_PeerConnection_AddTrack
ExtractNativePC(jni, j_pc)->AddTrack(reinterpret_cast<MediaStreamTrackInterface*>(native_track),
PeerConnection::AddTrack
PeerConnection::AddTrackPlanB
PeerConnection::CreateSender
RtpSenderBase::SetTrack 设置过来
VideoTrack 就是视频源
包含 webrtc::VideoSendStream* 对象 stream_, 就是 VideoSendStream 对象
包含 rtc::VideoSinkInterface<webrtc::VideoFrame>* 对象 encoder_sink_，其实就是一个 VideoStreamEncoder 对象
包含 VideoSendStreamParameters 对象 parameters_
包含 webrtc::RtpParameters 对象 rtp_parameters_
4. VideoSendStream 包含 VideoStreamEncoder 对象 video_stream_encoder_，
包含 std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_, 就是 VideoStreamEncoder 对象
VideoSendStreamImpl 的初始化过程中会绑定编码器与发送流的 pipeline SetSink
这样的编码器的视频就能传递到 VideoSendStreamImpl 对象
包含 std::unique_ptr<VideoSendStreamImpl> send_stream_，就是 VideoSendStreamImpl 对象
5. VideoSendStreamImpl
包含 VideoStreamEncoderInterface* const video_stream_encoder_， 就是 VideoStreamEncoder 对象
包含 EncoderRtcpFeedback encoder_feedback_，
包含 RtcpBandwidthObserver* const bandwidth_observer_
包含 RtpTransportControllerSendInterface* const transport_, 就是 RtpTransportControllerSend
包含 RtpVideoSenderInterface* const rtp_video_sender_， 就是 RtpVideoSender 对象
rtp_video_sender_(transport_->CreateRtpVideoSender(
suspended_ssrcs,
suspended_payload_states,
config_->rtp,
onfig_->rtcp_report_interval_ms,
config_->send_transport,
CreateObservers(call_stats, &encoder_feedback_, stats_proxy_, send_delay_stats),
event_log,
std::move(fec_controller),
CreateFrameEncryptionConfig(config_)))
RtpVideoSender 就是具体的发送数据对象
包含 CallStats* const call_stats_;
包含 BitrateAllocatorInterface* const bitrate_allocator_
包含 bool disable_padding_;
包含 int max_padding_bitrate_;
包含 int encoder_min_bitrate_bps_;
包含 uint32_t encoder_max_bitrate_bps_;
包含 uint32_t encoder_target_rate_bps_;
包含 double encoder_bitrate_priority_;
包含 bool has_packet_feedback_;
5. RtpVideoSender
包含 const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender> rtp_streams_;
在 RtpVideoSender::RtpVideoSender 时，
rtp_streams_(CreateRtpStreamSenders(clock,
rtp_config,
observers,
rtcp_report_interval_ms,
send_transport,
transport->GetBandwidthObserver(),
transport,
flexfec_sender_.get(),
event_log,
retransmission_limiter,
this,
frame_encryptor,
crypto_options)),
rtp_streams_ 就是 RTPSenderVideo 的一个列表
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
//-----------------------------------------------------------------------------
//
// 视频编码器的初始化
//
//-----------------------------------------------------------------------------
1.
这个的流程参见上面的 流程 15
VideoStreamEncoder::ReconfigureEncoder()
if (encoder_->InitEncode(&send_codec_, VideoEncoder::Settings(settings_.capabilities, number_of_cores_, max_data_payload_length)) != 0)
2.
VideoEncoderWrapper::InitEncode(const VideoCodec* codec_settings, const Settings& settings)
JNIEnv* jni = AttachCurrentThreadIfNeeded();
codec_settings_ = *codec_settings;
capabilities_ = settings.capabilities;
number_of_cores_ = settings.number_of_cores;
num_resets_ = 0;
{
rtc::CritScope lock(&encoder_queue_crit_);
encoder_queue_ = TaskQueueBase::Current();
}
return InitEncodeInternal(jni);
3.
VideoEncoderWrapper::InitEncodeInternal(JNIEnv* jni)
ScopedJavaLocalRef<jobject> capabilities = Java_Capabilities_Constructor(jni, capabilities_->loss_notification);
ScopedJavaLocalRef<jobject> settings = Java_Settings_Constructor(
jni, number_of_cores_, codec_settings_.width, codec_settings_.height,
static_cast<int>(codec_settings_.startBitrate),
static_cast<int>(codec_settings_.maxFramerate),
static_cast<int>(codec_settings_.numberOfSimulcastStreams),
automatic_resize_on, capabilities);
ScopedJavaLocalRef<jobject> callback = Java_VideoEncoderWrapper_createEncoderCallback(jni, jlongFromPointer(this));
int32_t status = JavaToNativeVideoCodecStatus(jni, Java_VideoEncoder_initEncode(jni, encoder_, settings, callback));
RTC_LOG(LS_INFO) << "initEncode: " << status;
encoder_info_.supports_native_handle = true;
encoder_info_.implementation_name = GetImplementationName(jni);
encoder_info_.scaling_settings = GetScalingSettingsInternal(jni);
encoder_info_.is_hardware_accelerated = IsHardwareVideoEncoder(jni, encoder_);
encoder_info_.has_internal_source = false;
if (status == WEBRTC_VIDEO_CODEC_OK) {
initialized_ = true;
}
return status;
4.
Java_VideoEncoder_initEncode(JNIEnv* env, const base::android::JavaRef<jobject>& obj, const base::android::JavaRef<jobject>& settings,
const base::android::JavaRef<jobject>& encodeCallback)
jclass clazz = org_webrtc_VideoEncoder_clazz(env);
CHECK_CLAZZ(env, obj.obj(), org_webrtc_VideoEncoder_clazz(env), NULL);
jni_generator::JniJavaCallContextChecked call_context;
call_context.Init<base::android::MethodID::TYPE_INSTANCE>(env, clazz, "initEncode",
"(Lorg/webrtc/VideoEncoder$Settings;Lorg/webrtc/VideoEncoder$Callback;)Lorg/webrtc/VideoCodecStatus;",
&g_org_webrtc_VideoEncoder_initEncode);
jobject ret = env->CallObjectMethod(obj.obj(), call_context.base.method_id, settings.obj(), encodeCallback.obj());
5. Java
HardwareVideoEncoder::initEncode(Settings settings, Callback callback)
initEncodeInternal
6. Java
HardwareVideoEncoder::initEncodeInternal
// 创建编码器，并设置参数
codec = mediaCodecWrapperFactory.createByCodecName(codecName);
// 开启编码数据发送线程
outputThread = createOutputThread();
outputThread.start();
//-----------------------------------------------------------------------------------------
// 编码流程分析
//-----------------------------------------------------------------------------------------
1. Java
ScreenCapturerAndroid:nFrame(VideoFrame frame)
capturerObserver.onFrameCaptured(frame);
2. Java
VideoSource::CapturerObserver:nFrameCaptured(VideoFrame frame)
final VideoProcessor.FrameAdaptationParameters parameters = nativeAndroidVideoTrackSource.adaptFrame(frame);
VideoFrame adaptedFrame = VideoProcessor.applyFrameAdaptationParameters(frame, parameters);
if (adaptedFrame != null) {
nativeAndroidVideoTrackSource.onFrameCaptured(adaptedFrame);
adaptedFrame.release();
}
这里的 nativeAndroidVideoTrackSource 就是
2.1
nativeCreateVideoSource(nativeFactory, isScreencast, alignTimestamps)
2.2
Java_org_webrtc_PeerConnectionFactory_nativeCreateVideoSource
2.3 ./sdk/android/src/jni/pc/peer_connection_factory.cc
JNI_PeerConnectionFactory_CreateVideoSource
return jlongFromPointer(CreateVideoSource(jni, factory->signaling_thread(), factory->worker_thread(), is_screencast, align_timestamps));
2.4 ./jni/pc/video.cc
void* CreateVideoSource(JNIEnv* env, rtc::Thread* signaling_thread, rtc::Thread* worker_thread, jboolean is_screencast, jboolean align_timestamps)
rtc::scoped_refptr<AndroidVideoTrackSource> source(new rtc::RefCountedObject<AndroidVideoTrackSource>(signaling_thread, env, is_screencast,
align_timestamps));
3.
NativeAndroidVideoTrackSource:nFrameCaptured(VideoFrame frame)
nativeOnFrameCaptured(nativeAndroidVideoTrackSource, frame.getRotation(), frame.getTimestampNs(), frame.getBuffer());
4.
Java_org_webrtc_NativeAndroidVideoTrackSource_nativeOnFrameCaptured
AndroidVideoTrackSource* native = reinterpret_cast<AndroidVideoTrackSource*>(nativeAndroidVideoTrackSource);
CHECK_NATIVE_PTR(env, jcaller, native, "OnFrameCaptured");
return native->OnFrameCaptured(env, rotation, timestampNs, base::android::JavaParamRef<jobject>(env, buffer));
5.
./sdk/android/src/jni/android_video_track_source.cc
AndroidVideoTrackSource::OnFrameCaptured
OnFrame(VideoFrame::Builder()
.set_video_frame_buffer(buffer)
.set_rotation(rotation)
.set_timestamp_us(j_timestamp_ns / rtc::kNumNanosecsPerMicrosec)
.build());
6.
调用基类的
./media/base/adapted_video_track_source.cc
AdaptedVideoTrackSource::OnFrame
broadcaster_.OnFrame(frame);
而 broadcaster_ 里的 sink 由接口 AddOrUpdateSink 添加 VideoStreamEncoder，具体流程参见《视频编码器的创建以及与源的关联》 11.5
7.
VideoBroadcaster::OnFrame(const webrtc::VideoFrame& frame)
if (sink_pair.wants.black_frames) {
webrtc::VideoFrame black_frame = webrtc::VideoFrame::Builder()
.set_video_frame_buffer(GetBlackFrameBuffer(frame.width(), frame.height()))
.set_rotation(frame.rotation())
.set_timestamp_us(frame.timestamp_us())
.set_id(frame.id())
.build();
sink_pair.sink->OnFrame(black_frame);
} else if (!previous_frame_sent_to_all_sinks_ && frame.has_update_rect()) {
// Since last frame was not sent to some sinks, no reliable update
// information is available, so we need to clear the update rect.
webrtc::VideoFrame copy = frame;
copy.clear_update_rect();
sink_pair.sink->OnFrame(copy);
} else {
sink_pair.sink->OnFrame(frame);
}
}
7.1
sink_pairs() 中的 sinks_ 是由 AddOrUpdateSink 添加的 VideoStreamEncoder，具体流程参见《视频编码器的创建以及与源的关联》 11.5
7.2
AdaptedVideoTrackSource::AddOrUpdateSink
broadcaster_.AddOrUpdateSink(sink, wants);
所以是由 AndroidVideoTrackSource (AdaptedVideoTrackSource) 添加的，下面我们分析这个 AndroidVideoTrackSource
8. ./video/video_stream_encoder.cc
void VideoStreamEncoder::OnFrame(const VideoFrame& video_frame)
MaybeEncodeVideoFrame(incoming_frame, post_time_us);
9. ./video/video_stream_encoder.cc
void VideoStreamEncoder::MaybeEncodeVideoFrame(const VideoFrame& video_frame, int64_t time_when_posted_us)
EncodeVideoFrame(video_frame, time_when_posted_us);
10. ./video/video_stream_encoder.cc
void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame, int64_t time_when_posted_us)
const int32_t encode_status = encoder_->Encode(out_frame, &next_frame_types_);
11.
int32_t VideoEncoderWrapper::Encode(const VideoFrame& frame, const std::vector<VideoFrameType>* frame_types)
JNIEnv* jni = AttachCurrentThreadIfNeeded();
// Construct encode info.
ScopedJavaLocalRef<jobjectArray> j_frame_types = NativeToJavaFrameTypeArray(jni, *frame_types);
ScopedJavaLocalRef<jobject> encode_info = Java_EncodeInfo_Constructor(jni, j_frame_types);
FrameExtraInfo info;
info.capture_time_ns = frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec;
info.timestamp_rtp = frame.timestamp();
frame_extra_infos_.push_back(info);
ScopedJavaLocalRef<jobject> j_frame = NativeToJavaVideoFrame(jni, frame);
ScopedJavaLocalRef<jobject> ret = Java_VideoEncoder_encode(jni, encoder_, j_frame, encode_info);
ReleaseJavaVideoFrame(jni, j_frame);
return HandleReturnCode(jni, ret, "encode");
12.
static base::android::ScopedJavaLocalRef<jobject> Java_VideoEncoder_encode(JNIEnv* env, const
base::android::JavaRef<jobject>& obj, const base::android::JavaRef<jobject>& frame,
const base::android::JavaRef<jobject>& info) {
jclass clazz = org_webrtc_VideoEncoder_clazz(env);
CHECK_CLAZZ(env, obj.obj(), org_webrtc_VideoEncoder_clazz(env), NULL);
jni_generator::JniJavaCallContextChecked call_context;
call_context.Init<base::android::MethodID::TYPE_INSTANCE>(env, clazz, "encode",
"(Lorg/webrtc/VideoFrame;Lorg/webrtc/VideoEncoder$EncodeInfo;)Lorg/webrtc/VideoCodecStatus;",
&g_org_webrtc_VideoEncoder_encode);
jobject ret = env->CallObjectMethod(obj.obj(), call_context.base.method_id, frame.obj(), info.obj());
return base::android::ScopedJavaLocalRef<jobject>(env, ret);
}
13. Java
public VideoCodecStatus HardwareVideoEncoder::encode(VideoFrame videoFrame, EncodeInfo encodeInfo)
这个里面对视频进行完成了编码
//--------------------------------------------------------------------------------------------
//
// 视频的发送
//
//--------------------------------------------------------------------------------------------
1. Java
视频编码器初始化流程 6， outputThread.start(); 开启了视频发送线程
void HardwareVideoEncoder::deliverEncodedImage()
callback.onEncodedFrame(encodedImage, new CodecSpecificInfo());
这个 callback 是初始化编码器时，传递过来的。参见编码器初始化流程 3
ScopedJavaLocalRef<jobject> callback = Java_VideoEncoderWrapper_createEncoderCallback(jni, jlongFromPointer(this));
static base::android::ScopedJavaLocalRef<jobject> Java_VideoEncoderWrapper_createEncoderCallback(JNIEnv* env, jlong nativeEncoder) {
jclass clazz = org_webrtc_VideoEncoderWrapper_clazz(env);
CHECK_CLAZZ(env, clazz,
org_webrtc_VideoEncoderWrapper_clazz(env), NULL);
jni_generator::JniJavaCallContextChecked call_context;
call_context.Init<
base::android::MethodID::TYPE_STATIC>(
env,
clazz,
"createEncoderCallback",
"(J)Lorg/webrtc/VideoEncoder$Callback;",
&g_org_webrtc_VideoEncoderWrapper_createEncoderCallback);
jobject ret = env->CallStaticObjectMethod(clazz, call_context.base.method_id, nativeEncoder);
return base::android::ScopedJavaLocalRef<jobject>(env, ret);
}
这个就是 Java 层的一个 lambda 函数
static VideoEncoder.Callback VideoEncoderWrapper::createEncoderCallback(final long nativeEncoder) {
return (EncodedImage frame,
VideoEncoder.CodecSpecificInfo info) -> nativeOnEncodedFrame(nativeEncoder, frame);
}
2.
nativeOnEncodedFrame
3.
JNI_GENERATOR_EXPORT void Java_org_webrtc_VideoEncoderWrapper_nativeOnEncodedFrame(
JNIEnv* env,
jclass jcaller,
jlong nativeVideoEncoderWrapper,
jobject frame) {
VideoEncoderWrapper* native = reinterpret_cast<VideoEncoderWrapper*>(nativeVideoEncoderWrapper);
CHECK_NATIVE_PTR(env, jcaller, native, "OnEncodedFrame");
return native->OnEncodedFrame(env, base::android::JavaParamRef<jobject>(env, frame));
}
4.
void VideoEncoderWrapper::OnEncodedFrame(JNIEnv* jni, const JavaRef<jobject>& j_encoded_image)
callback_->OnEncodedImage(frame_copy, &info, &header);
而 callback_ 是通过函数 RegisterEncodeCompleteCallback 注册过来的。
在函数 VideoStreamEncoder::ReconfigureEncoder() 里 encoder_(VideoEncoderWrapper)
encoder_->RegisterEncodeCompleteCallback(this);
上述流程分析后我们得出下一步要进入 VideoStreamEncoder::OnEncodedImage 函数了
5.
EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation)
EncodedImageCallback::Result result = sink_->OnEncodedImage(
image_copy, codec_info_copy ? codec_info_copy.get() : codec_specific_info,
fragmentation_copy ? fragmentation_copy.get() : fragmentation);
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
而 sink_ 是 VideoStreamEncoder::SetSink 设置过来的 VideoSendStreamImpl ，具体流程如下
下面是编码器与发送对象的关联过程！！！！
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
5.1
WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream
webrtc::VideoSendStream::Config config = parameters_.config.Copy();
stream_ = call_->CreateVideoSendStream(std::move(config), parameters_.encoder_config.Copy());
这个产生一个 VideoSendStream 对象， WebRtcVideoSendStream 包含一个 VideoSendStream 对象
5.2 ./call/call.cc
webrtc::VideoSendStream* Call::CreateVideoSendStream
VideoSendStream* send_stream = new VideoSendStream(
clock_, num_cpu_cores_, module_process_thread_.get(), task_queue_factory_,
call_stats_.get(), transport_send_ptr_, bitrate_allocator_.get(),
video_send_delay_stats_.get(), event_log_, std::move(config),
std::move(encoder_config), suspended_video_send_ssrcs_,
suspended_video_payload_states_, std::move(fec_controller));
transport_send_ptr_ 就是 Call::Create 里的 RtpTransportControllerSend 参见下面的
5.3 ./video/video_send_stream.cc
VideoSendStream::VideoSendStream(
Clock* clock,
int num_cpu_cores,
ProcessThread* module_process_thread,
TaskQueueFactory* task_queue_factory,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
std::unique_ptr<FecController> fec_controller): worker_queue_(transport->GetWorkerQueue()),
stats_proxy_(clock, config, encoder_config.content_type),
config_(std::move(config)),
content_type_(encoder_config.content_type)
video_stream_encoder_ = CreateVideoStreamEncoder(clock, task_queue_factory, num_cpu_cores,
&stats_proxy_, config_.encoder_settings);
CreateVideoStreamEncoder 函数产生一个 VideoStreamEncoder 对象 video_stream_encoder_，
std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(Clock* clock, TaskQueueFactory* task_queue_factory,
uint32_t number_of_cores, VideoStreamEncoderObserver* encoder_stats_observer, const VideoStreamEncoderSettings& settings) {
return std::make_unique<VideoStreamEncoder>(clock, number_of_cores, encoder_stats_observer, settings,
std::make_unique<OveruseFrameDetector>(encoder_stats_observer),task_queue_factory);
}
worker_queue_->ostTask(ToQueuedTask([this, clock, call_stats, transport, bitrate_allocator,
send_delay_stats, event_log, &suspended_ssrcs, &encoder_config, &suspended_payload_states,
&fec_controller]() { send_stream_.reset(new VideoSendStreamImpl(clock, &stats_proxy_,
worker_queue_, call_stats, transport, bitrate_allocator, send_delay_stats,
video_stream_encoder_.get(), event_log, &config_, encoder_config.max_bitrate_bps,
encoder_config.bitrate_priority, suspended_ssrcs, suspended_payload_states,
encoder_config.content_type, std::move(fec_controller)));}, [this]() { thread_sync_event_.Set(); }));
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
5.4 这个地方关联编码器 VideoStreamEncoder 对象与 VideoSendStreamImpl 对象
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
VideoSendStreamImpl::VideoSendStreamImpl(
Clock* clock,
SendStatisticsProxy* stats_proxy,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
VideoStreamEncoderInterface* video_stream_encoder,
RtcEventLog* event_log,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
double initial_encoder_bitrate_priority,
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type,
std::unique_ptr<FecController> fec_controller)
这个地方调用 VideoStreamEncoder 的 SetSink , 就是关联 VideoStreamEncoder 与 VideoSendStreamImpl
video_stream_encoder_->SetSink(this, rotation_applied);
上述流程分析后我们得出下一步要进入 VideoSendStreamImpl::OnEncodedImage 函数了
6.
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation)
EncodedImageCallback::Result result(EncodedImageCallback::Result::OK);
// 我们需要分析 rtp_video_sender_ 到底是哪个类的对象
result = rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info, fragmentation);
rtp_video_sender_ 是初始化过程中
rtp_video_sender_(transport_->CreateRtpVideoSender(
suspended_ssrcs,
suspended_payload_states,
config_->rtp,
config_->rtcp_report_interval_ms,
config_->send_transport,
CreateObservers(call_stats,
&encoder_feedback_,
stats_proxy_,
send_delay_stats),
event_log,
std::move(fec_controller),
CreateFrameEncryptionConfig(config_)))
这里的 transport 就是 RtpTransportControllerSend 这里产生一个 RtpVideoSender 对象
RtpTransportControllerSend::CreateRtpVideoSender
video_rtp_senders_.push_back(std::make_unique<RtpVideoSender>(clock_, suspended_ssrcs, states,
rtp_config, rtcp_report_interval_ms, send_transport, observers,
// TODO(holmer): Remove this circular dependency by injecting
// the parts of RtpTransportControllerSendInterface that are really used.
this, event_log, &retransmission_rate_limiter_, std::move(fec_controller),
frame_encryption_config.frame_encryptor, frame_encryption_config.crypto_options));
上述流程分析后我们得出下一步要进入 RtpVideoSender::OnEncodedImage 函数了
7.
EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation)
// 我们需要分析 rtp_streams_[stream_index].sender_video 到底是哪个类的对象
bool send_result = rtp_streams_[stream_index].sender_video->SendVideo(
rtp_config_.payload_type, codec_type_, rtp_timestamp,
encoded_image.capture_time_ms_, encoded_image, fragmentation,
params_[stream_index].GetRtpVideoHeader(encoded_image, codec_specific_info, shared_frame_id_),
expected_retransmission_time_ms);
7.1 rtp_streams_ 的创建
RtpVideoSender::RtpVideoSender(
Clock* clock,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
RateLimiter* retransmission_limiter,
std::unique_ptr<FecController> fec_controller,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options)
rtp_streams_(CreateRtpStreamSenders(clock, rtp_config, observers, rtcp_report_interval_ms, send_transport, transport->GetBandwidthObserver(),
transport, flexfec_sender_.get(), event_log, retransmission_limiter, this, frame_encryptor, crypto_options))
7.2 CreateRtpStreamSenders 函数
std::vector<RtpStreamSender> CreateRtpStreamSenders(
Clock* clock,
const RtpConfig& rtp_config,
const RtpSenderObservers& observers,
int rtcp_report_interval_ms,
Transport* send_transport,
RtcpBandwidthObserver* bandwidth_callback,
RtpTransportControllerSendInterface* transport,
FlexfecSender* flexfec_sender,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcp::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
configuration.receiver_only = false;
configuration.outgoing_transport = send_transport;
configuration.intra_frame_callback = observers.intra_frame_callback;
configuration.rtcp_loss_notification_observer = observers.rtcp_loss_notification_observer;
configuration.bandwidth_callback = bandwidth_callback;
configuration.network_state_estimate_observer = transport->network_state_estimate_observer();
configuration.transport_feedback_callback = transport->transport_feedback_observer();
configuration.rtt_stats = observers.rtcp_rtt_stats;
configuration.rtcp_packet_type_counter_observer = observers.rtcp_type_observer;
// ---------------------------------------------------
// RtpPacketSender* RtpTransportControllerSend::packet_sender() 应该是这个吧 ???????
configuration.paced_sender = transport->packet_sender();
//----------------------------------------------------
configuration.send_bitrate_observer = observers.bitrate_observer;
configuration.send_side_delay_observer = observers.send_delay_observer;
configuration.send_packet_observer = observers.send_packet_observer;
configuration.event_log = event_log;
configuration.retransmission_rate_limiter = retransmission_rate_limiter;
configuration.overhead_observer = overhead_observer;
configuration.rtp_stats_callback = observers.rtp_stats;
configuration.frame_encryptor = frame_encryptor;
configuration.require_frame_encryption = crypto_options.sframe.require_frame_encryption;
configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
std::vector<RtpStreamSender> rtp_streams;
const std::vector<uint32_t>& flexfec_protected_ssrcs = rtp_config.flexfec.protected_media_ssrcs;
RTC_DCHECK(rtp_config.rtx.ssrcs.empty() || rtp_config.rtx.ssrcs.size() == rtp_config.rtx.ssrcs.size());
// 根据源 ssrcs 创建多个发送对象
for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
configuration.local_media_ssrc = rtp_config.ssrcs;
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(), flexfec_protected_ssrcs.end(),
configuration.local_media_ssrc) != flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
auto playout_delay_oracle = std::make_unique<layoutDelayOracle>();
configuration.ack_observer = playout_delay_oracle.get();
if (rtp_config.rtx.ssrcs.size() > i) {
configuration.rtx_send_ssrc = rtp_config.rtx.ssrcs;
}
// 这个产生一个 ModuleRtpRtcpImpl 对象
auto rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
FieldTrialBasedConfig field_trial_config;
RTPSenderVideo::Config video_config;
video_config.clock = configuration.clock;
video_config.rtp_sender = rtp_rtcp->RtpSender();
video_config.flexfec_sender = configuration.flexfec_sender;
video_config.playout_delay_oracle = playout_delay_oracle.get();
video_config.frame_encryptor = frame_encryptor;
video_config.require_frame_encryption =
crypto_options.sframe.require_frame_encryption;
video_config.need_rtp_packet_infos = rtp_config.lntf.enabled;
video_config.enable_retransmit_all_layers = false;
video_config.field_trials = &field_trial_config;
const bool should_disable_red_and_ulpfec =
ShouldDisableRedAndUlpfec(enable_flexfec, rtp_config);
if (rtp_config.ulpfec.red_payload_type != -1 &&
!should_disable_red_and_ulpfec) {
video_config.red_payload_type = rtp_config.ulpfec.red_payload_type;
}
if (rtp_config.ulpfec.ulpfec_payload_type != -1 &&
!should_disable_red_and_ulpfec) {
video_config.ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;
}
// 创建对象 RTPSenderVideo
auto sender_video = std::make_unique<RTPSenderVideo>(video_config);
rtp_streams.emplace_back(std::move(playout_delay_oracle),
std::move(rtp_rtcp), std::move(sender_video));
}
return rtp_streams;
}
上述流程分析后我们得出下一步要进入 RTPSenderVideo::SendVideo 函数了
8. 这个里面对包做了大量处理
bool RTPSenderVideo::SendVideo(int payload_type, absl:ptional<VideoCodecType> codec_type, uint32_t rtp_timestamp, int64_t capture_time_ms,
rtc::ArrayView<const uint8_t> payload, const RTPFragmentationHeader* fragmentation, RTPVideoHeader video_header,
absl:ptional<int64_t> expected_retransmission_time_ms)
LogAndSendToNetwork(std::move(rtp_packets), unpacketized_payload_size);
9.
void RTPSenderVideo: ogAndSendToNetwork(std::vector<std::unique_ptr<RtpPacketToSend>> packets, size_t unpacketized_payload_size)
rtp_sender_->EnqueuePackets(std::move(packets));
rtp_sender_ 是构造函数传递过来的参数 RTPSenderVideo::RTPSenderVideo(const Config& config) : rtp_sender_(config.rtp_sender)
过程如下：
./call/rtp_video_sender.cc
RtpRtcp::Configuration configuration;
configuration.paced_sender = transport->packet_sender();
auto rtp_rtcp = RtpRtcp::Create(configuration);
// 就是这个 rtp_sender
video_config.rtp_sender = rtp_rtcp->RtpSender();
auto sender_video = std::make_unique<RTPSenderVideo>(video_config);
我们分析 RtpRtcp::Create 函数以及 rtp_rtcp->RtpSender() 函数
./modules/rtp_rtcp/source/rtp_rtcp_impl.cc
std::unique_ptr<RtpRtcp> RtpRtcp::Create(const Configuration& configuration) {
RTC_DCHECK(configuration.clock);
return std::make_unique<ModuleRtpRtcpImpl>(configuration);
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtcp_sender_(configuration),
rtcp_receiver_(configuration, this),
clock_(configuration.clock),
last_bitrate_process_time_(clock_->TimeInMilliseconds()),
last_rtt_process_time_(clock_->TimeInMilliseconds()),
next_process_time_(clock_->TimeInMilliseconds() + kRtpRtcpMaxIdleTimeProcessMs),
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
ack_observer_(configuration.ack_observer),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
// 这是 rtp_sender_ 产生的地方
rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(
rtp_sender_->packet_generator.TimestampOffset());
}
// Set default packet size limit.
// TODO(nisse): Kind-of duplicates
// webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
const size_t kTcpOverIpv4HeaderSize = 40;
SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
}
是 ModuleRtpRtcpImpl 的 RtpSender () 过来的 就是 ModuleRtpRtcpImpl 的 rtp_sender_ 对象的
RTPSender* ModuleRtpRtcpImpl::RtpSender() {
return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
}
而 packet_generator 的创建见下面流程
ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext(
const RtpRtcp::Configuration& config)
: packet_history(config.clock),
packet_sender(config, &packet_history),
non_paced_sender(&packet_sender),
packet_generator(
config,
&packet_history,
config.paced_sender ? config.paced_sender : &non_paced_sender) {}
而 packet_generator 就是一个 RTPSender 对象，结构定义如下：
struct RtpSenderContext {
explicit RtpSenderContext(const RtpRtcp::Configuration& config);
// Storage of packets, for retransmissions and padding, if applicable.
RtpPacketHistory packet_history;
// Handles final time timestamping/stats/etc and handover to Transport.
RtpSenderEgress packet_sender;
// If no paced sender configured, this class will be used to pass packets
// from |packet_generator_| to |packet_sender_|.
RtpSenderEgress::NonPacedPacketSender non_paced_sender;
// Handles creation of RTP packets to be sent.
RTPSender packet_generator;
};
上述流程分析后我们得出下一步要进入 RTPSender::EnqueuePackets 函数了
10.
./modules/rtp_rtcp/source/rtp_sender.cc
上述步骤把发送的数据放到 RTPSender 的队列里了
void RTPSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
RTC_DCHECK(!packets.empty());
int64_t now_ms = clock_->TimeInMilliseconds();
for (auto& packet : packets) {
RTC_DCHECK(packet);
RTC_CHECK(packet->packet_type().has_value())
<< "acket type must be set before sending.";
if (packet->capture_time_ms() <= 0) {
packet->set_capture_time_ms(now_ms);
}
}
paced_sender_->EnqueuePackets(std::move(packets));
}
而 paced_sender_ 是构造时传递过来的
RTPSender::RTPSender(const RtpRtcp::Configuration& config,
RtpPacketHistory* packet_history,
RtpPacketSender* packet_sender)
: clock_(config.clock),
random_(clock_->TimeInMicroseconds()),
audio_configured_(config.audio),
ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.flexfec_sender
? absl::make_optional(config.flexfec_sender->ssrc())
: absl::nullopt),
packet_history_(packet_history),
paced_sender_(packet_sender),
sending_media_(true), // Default to sending media.
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(config.extmap_allow_mixed),
// RTP variables
sequence_number_forced_(false),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
supports_bwe_extension_(false),
retransmission_rate_limiter_(config.retransmission_rate_limiter)
而这里的 paced_sender_ 就是步骤 9 里的 configuration.paced_sender = transport->packet_sender(); 也就是
RtpPacketSender* RtpTransportControllerSend::packet_sender();
./call/rtp_transport_controller_send.cc
RtpPacketSender* RtpTransportControllerSend::packet_sender() {
if (use_task_queue_pacer_) {
return task_queue_pacer_.get();
}
return process_thread_pacer_.get();
}
其实就是 process_thread_pacer_ 或 task_queue_pacer_
RtpTransportControllerSend::RtpTransportControllerSend(
Clock* clock,
webrtc::RtcEventLog* event_log,
NetworkStatePredictorFactoryInterface* predictor_factory,
NetworkControllerFactoryInterface* controller_factory,
const BitrateConstraints& bitrate_config,
std::unique_ptr<rocessThread> process_thread,
TaskQueueFactory* task_queue_factory,
const WebRtcKeyValueConfig* trials)
: clock_(clock),
event_log_(event_log),
bitrate_configurator_(bitrate_config),
process_thread_(std::move(process_thread)),
use_task_queue_pacer_(IsEnabled(trials, "WebRTC-TaskQueuePacer")),
process_thread_pacer_(use_task_queue_pacer_
? nullptr
: new PacedSender(clock,
&packet_router_,
event_log,
trials,
process_thread_.get())),
task_queue_pacer_(use_task_queue_pacer_
? new TaskQueuePacedSender(clock,
&packet_router_,
event_log,
trials,
task_queue_factory)
其实就是一个 PacedSender 对象
11.
./modules/pacing/paced_sender.cc
void PacedSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
{
rtc::CritScope cs(&critsect_);
for (auto& packet : packets) {
pacing_controller_.EnqueuePacket(std::move(packet));
}
}
MaybeWakupProcessThread();
}
12.
void PacedSender::MaybeWakupProcessThread() {
// Tell the process thread to call our TimeUntilNextProcess() method to get
// a new time for when to call Process().
if (process_thread_ &&
process_mode_ == PacingController: rocessMode::kDynamic) {
process_thread_->WakeUp(&module_proxy_);
}
}
13.
void PacedSender:rocess() {
rtc::CritScope cs(&critsect_);
pacing_controller_.ProcessPackets();
}
pacing_controller_ 就是对象
PacingController pacing_controller_ RTC_GUARDED_BY(critsect_);
./modules/pacing/paced_sender.cc
PacedSender:acedSender(Clock* clock,
PacketRouter* packet_router,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials,
ProcessThread* process_thread)
: process_mode_((field_trials != nullptr &&
field_trials->Lookup("WebRTC-Pacer-DynamicProcess")
.find("Enabled") == 0)
? PacingController:rocessMode::kDynamic
: PacingController:rocessMode::kPeriodic),
pacing_controller_(clock,
static_cast<acingController:acketSender*>(this),
event_log,
field_trials,
process_mode_),
clock_(clock),
packet_router_(packet_router),
process_thread_(process_thread) {
if (process_thread_)
process_thread_->RegisterModule(&module_proxy_, RTC_FROM_HERE);
}
14.
./modules/pacing/pacing_controller.cc
void PacingController:rocessPackets()
packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
packet_sender_ 是初始化过程中传递过来的
PacingController:acingController(Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials,
ProcessMode mode)
: mode_(mode),
clock_(clock),
packet_sender_(packet_sender)
15.
./modules/pacing/paced_sender.cc
void PacedSender::SendRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info) {
critsect_.Leave();
packet_router_->SendPacket(std::move(packet), cluster_info);
critsect_.Enter();
}
packet_router_ 是初始化传递过来的
PacedSender:acedSender(Clock* clock,
PacketRouter* packet_router,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials,
ProcessThread* process_thread)
: process_mode_((field_trials != nullptr &&
field_trials->Lookup("WebRTC-Pacer-DynamicProcess")
.find("Enabled") == 0)
? PacingController:rocessMode::kDynamic
: PacingController::ProcessMode::kPeriodic),
pacing_controller_(clock,
static_cast<acingController::PacketSender*>(this),
event_log,
field_trials,
process_mode_),
clock_(clock),
packet_router_(packet_router)
就是 RtpTransportControllerSend 的 packet_router_
PacketRouter packet_router_;
16.
./modules/pacing/packet_router.cc
void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info) {
rtc::CritScope cs(&modules_crit_);
// With the new pacer code path, transport sequence numbers are only set here,
// on the pacer thread. Therefore we don't need atomics/synchronization.
if (packet->IsExtensionReserved<TransportSequenceNumber>()) {
packet->SetExtension<TransportSequenceNumber>((++transport_seq_) & 0xFFFF);
}
uint32_t ssrc = packet->Ssrc();
auto kv = send_modules_map_.find(ssrc);
if (kv == send_modules_map_.end()) {
RTC_LOG(LS_WARNING)
<< "Failed to send packet, matching RTP module not found "
"or transport error. SSRC = "
<< packet->Ssrc() << ", sequence number " << packet->SequenceNumber();
return;
}
RtpRtcp* rtp_module = kv->second.first;
if (!rtp_module->TrySendPacket(packet.get(), cluster_info)) {
RTC_LOG(LS_WARNING) << "Failed to send packet, rejected by RTP module.";
return;
}
if (rtp_module->SupportsRtxPayloadPadding()) {
// This is now the last module to send media, and has the desired
// properties needed for payload based padding. Cache it for later use.
last_send_module_ = rtp_module;
}
}
send_modules_map_ 则是这么设置过来的
./call/rtp_video_sender.cc
RtpVideoSender::RtpVideoSender(
Clock* clock,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
RateLimiter* retransmission_limiter,
std::unique_ptr<FecController> fec_controller,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options)
: send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
account_for_packetization_overhead_(!webrtc::field_trial::IsDisabled(
"WebRTC-SubtractPacketizationOverhead")),
use_early_loss_detection_(
!webrtc::field_trial::IsDisabled("WebRTC-UseEarlyLossDetection")),
active_(false),
module_process_thread_(nullptr),
suspended_ssrcs_(std::move(suspended_ssrcs)),
flexfec_sender_(
MaybeCreateFlexfecSender(clock, rtp_config, suspended_ssrcs_)),
fec_controller_(std::move(fec_controller)),
fec_allowed_(true),
rtp_streams_(CreateRtpStreamSenders(clock,
rtp_config,
observers,
rtcp_report_interval_ms,
send_transport,
transport->GetBandwidthObserver(),
transport,
flexfec_sender_.get(),
event_log,
retransmission_limiter,
this,
frame_encryptor,
crypto_options))
for (const RtpStreamSender& stream : rtp_streams_) {
constexpr bool remb_candidate = true;
transport->packet_router()->AddSendRtpModule(stream.rtp_rtcp.get(),
remb_candidate);
}
std::vector<RtpStreamSender> CreateRtpStreamSenders(
Clock* clock,
const RtpConfig& rtp_config,
const RtpSenderObservers& observers,
int rtcp_report_interval_ms,
Transport* send_transport,
RtcpBandwidthObserver* bandwidth_callback,
RtpTransportControllerSendInterface* transport,
FlexfecSender* flexfec_sender,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options)
rtp_streams.emplace_back(std::move(playout_delay_oracle),
std::move(rtp_rtcp), std::move(sender_video));
return rtp_streams;
其实就是 RtpRtcp::Create 的对象 ModuleRtpRtcpImpl
17.
./modules/rtp_rtcp/source/rtp_rtcp_impl.cc
bool ModuleRtpRtcpImpl::TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(rtp_sender_);
// TODO(sprang): Consider if we can remove this check.
if (!rtp_sender_->packet_generator.SendingMedia()) {
return false;
}
rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
return true;
}
就是 struct RtpSenderContext 里 RtpSenderEgress packet_sender;
18.
./modules/rtp_rtcp/source/rtp_sender_egress.cc
void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info)
const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
19.
bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
UpdateRtpOverhead(packet);
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
而 transport_ 是构造时传递过来的
RtpSenderEgress::RtpSenderEgress(const RtpRtcp::Configuration& config,
RtpPacketHistory* packet_history)
: ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.flexfec_sender
? absl::make_optional(config.flexfec_sender->ssrc())
: absl::nullopt),
populate_network2_timestamp_(config.populate_network2_timestamp),
send_side_bwe_with_overhead_(
IsEnabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)),
clock_(config.clock),
packet_history_(packet_history),
transport_(config.outgoing_transport),
其实就是函数 std::vector<RtpStreamSender> CreateRtpStreamSenders
RtpRtcp::Configuration configuration;
configuration.outgoing_transport = send_transport;
继续追踪 就是这个 VideoSendStreamImpl 里的 config_->send_transport
rtp_video_sender_(transport_->CreateRtpVideoSender(
suspended_ssrcs,
suspended_payload_states,
config_->rtp,
onfig_->rtcp_report_interval_ms,
config_->send_transport,
VideoSendStream::VideoSendStream
webrtc::VideoSendStream* Call::CreateVideoSendStream
WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream
webrtc::VideoSendStream::Config config = parameters_.config.Copy();
WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
parameters_(std::move(config), options, max_bitrate_bps, codec_settings)
VideoSendStreamParameters parameters_
bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
if (!ValidateStreamParams(sp))
return false;
if (!ValidateSendSsrcAvailability(sp))
return false;
for (uint32_t used_ssrc : sp.ssrcs)
send_ssrcs_.insert(used_ssrc);
webrtc::VideoSendStream::Config config(this);
VideoSendStream::Config::Config(Transport* send_transport)
其实 send_transport 就是 WebRtcVideoChannel 对象
20.
./media/engine/webrtc_video_engine.cc
bool WebRtcVideoChannel::SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
rtc::PacketOptions rtc_options;
rtc_options.packet_id = options.packet_id;
if (DscpEnabled()) {
rtc_options.dscp = PreferredDscp();
}
rtc_options.info_signaled_after_sent.included_in_feedback =
options.included_in_feedback;
rtc_options.info_signaled_after_sent.included_in_allocation =
options.included_in_allocation;
return MediaChannel::SendPacket(&packet, rtc_options);
}
21.
./media/base/media_channel.h
bool SendPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options) {
return DoSendPacket(packet, false, options);
}
22.
./media/base/media_channel.h
bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
bool rtcp,
const rtc::PacketOptions& options) {
rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return false;
return (!rtcp) ? network_interface_->SendPacket(packet, options)
: network_interface_->SendRtcp(packet, options);
}
network_interface_ 是通过 void MediaChannel::SetInterface 设置过来的
./pc/channel.cc
void BaseChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
RTC_DCHECK_RUN_ON(worker_thread_);
media_transport_config_ = media_transport_config;
network_thread_->Invoke<void>(
RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this, media_transport_config);
}
23.
其实就是 VideoChannel
bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return SendPacket(false, packet, options);
}
bool BaseChannel::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options)
return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
: rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
rtp_transport_ 是通过接口
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport)
./pc/peer_connection.cc
cricket::VideoChannel* PeerConnection::CreateVideoChannel(const std::string& mid)
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
video_channel->SetRtpTransport(rtp_transport);
./pc/peer_connection.h
RtpTransportInternal* GetRtpTransport(const std::string& mid)
RTC_RUN_ON(signaling_thread()) {
auto rtp_transport = transport_controller_->GetRtpTransport(mid);
RTC_DCHECK(rtp_transport);
return rtp_transport;
}
bool PeerConnection::Initialize
transport_controller_.reset(new JsepTransportController(
signaling_thread(), network_thread(), port_allocator_.get(),
async_resolver_factory_.get(), config));
./pc/jsep_transport_controller.cc
RtpTransportInternal* JsepTransportController::GetRtpTransport(
const std::string& mid) const {
auto jsep_transport = GetJsepTransportForMid(mid);
if (!jsep_transport) {
return nullptr;
}
return jsep_transport->rtp_transport();
}
./pc/jsep_transport_controller.cc
const cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
const std::string& mid) const {
auto it = mid_to_transport_.find(mid);
return it == mid_to_transport_.end() ? nullptr : it->second;
}
mid_to_transport_
bool JsepTransportController::SetTransportForMid
RTCError JsepTransportController::MaybeCreateJsepTransport(
bool local,
const cricket::ContentInfo& content_info,
const cricket::SessionDescription& description)
rtc::scoped_refptr<webrtc::IceTransportInterface> ice =
CreateIceTransport(content_info.name, /*rtcp=*/false);
/------------------------------------------------------
rtc::scoped_refptr<webrtc::IceTransportInterface>
JsepTransportController::CreateIceTransport(const std::string& transport_name,
bool rtcp) {
int component = rtcp ? cricket::ICE_CANDIDATE_COMPONENT_RTCP
: cricket::ICE_CANDIDATE_COMPONENT_RTP;
IceTransportInit init;
init.set_port_allocator(port_allocator_);
init.set_async_resolver_factory(async_resolver_factory_);
init.set_event_log(config_.event_log);
return config_.ice_transport_factory->CreateIceTransport(
transport_name, component, std::move(init));
}
//-----------------------------------------------------------
./api/ice_transport_factory.cc
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
cricket::PortAllocator* port_allocator) {
IceTransportInit init;
init.set_port_allocator(port_allocator);
return CreateIceTransport(std::move(init));
}
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
IceTransportInit init) {
return new rtc::RefCountedObject<IceTransportWithTransportChannel>(
std::make_unique<cricket::P2PTransportChannel>(
"", 0, init.port_allocator(), init.async_resolver_factory(),
init.event_log()));
}
//-----------------------------------------------------------
/------------------------------------------------------
std::unique_ptr<cricket:tlsTransportInternal> rtp_dtls_transport =
CreateDtlsTransport(content_info, ice->internal(), nullptr);
// 其实就是 DtlsTransport 对象
std::unique_ptr<cricket:tlsTransportInternal>
JsepTransportController::CreateDtlsTransport(
const cricket::ContentInfo& content_info,
cricket::IceTransportInternal* ice,
DatagramTransportInterface* datagram_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
std::unique_ptr<cricket:tlsTransportInternal> dtls;
if (datagram_transport) {
RTC_DCHECK(config_.use_datagram_transport ||
config_.use_datagram_transport_for_data_channels);
} else if (config_.dtls_transport_factory) {
dtls = config_.dtls_transport_factory->CreateDtlsTransport(ice, config_.crypto_options);
} else {
// 执行的是这个 DtlsTransport
dtls = std::make_unique<cricket:tlsTransport>(ice, config_.crypto_options, config_.event_log);
}
/-------------------------------------------------------------------
ice 就是底层的传输对象
/-------------------------------------------------------------------
dtls_srtp_transport = CreateDtlsSrtpTransport(
content_info.name, rtp_dtls_transport.get(), rtcp_dtls_transport.get());
std::unique_ptr<webrtc:tlsSrtpTransport>
JsepTransportController::CreateDtlsSrtpTransport(
const std::string& transport_name,
cricket:tlsTransportInternal* rtp_dtls_transport,
cricket:tlsTransportInternal* rtcp_dtls_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
auto dtls_srtp_transport = std::make_unique<webrtc:tlsSrtpTransport>(rtcp_dtls_transport == nullptr);
if (config_.enable_external_auth) {
dtls_srtp_transport->EnableExternalAuth();
}
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport, rtcp_dtls_transport);
dtls_srtp_transport->SetActiveResetSrtpParams(config_.active_reset_srtp_params);
dtls_srtp_transport->SignalDtlsStateChange.connect(this, &JsepTransportController::UpdateAggregateStates_n);
return dtls_srtp_transport;
}
我们看看 JsepTransport 创建过程
std::unique_ptr<cricket::JsepTransport> jsep_transport =
std::make_unique<cricket::JsepTransport>(
content_info.name, certificate_, std::move(ice), std::move(rtcp_ice),
std::move(unencrypted_rtp_transport), std::move(sdes_transport),
std::move(dtls_srtp_transport), std::move(datagram_rtp_transport),
std::move(rtp_dtls_transport), std::move(rtcp_dtls_transport),
std::move(sctp_transport), std::move(datagram_transport),
data_channel_transport);
JsepTransport 初始化过程
./pc/jsep_transport.cc
JsepTransport::JsepTransport(
const std::string& mid,
const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport,
rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport,
std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
std::unique_ptr<webrtc:tlsSrtpTransport> dtls_srtp_transport,
std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport,
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
std::unique_ptr<SctpTransportInternal> sctp_transport,
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport,
webrtc::DataChannelTransportInterface* data_channel_transport)
jsep_transport->rtp_transport() 则是 JsepTransport
webrtc::RtpTransportInternal* JsepTransport::rtp_transport() const {
rtc::CritScope scope(&accessor_lock_);
if (composite_rtp_transport_) {
return composite_rtp_transport_.get();
} else if (datagram_rtp_transport_) {
return datagram_rtp_transport_.get();
} else {
// 就是这个
return default_rtp_transport();
}
}
// Returns the default (non-datagram) rtp transport, if any.
webrtc::RtpTransportInternal* JsepTransport::default_rtp_transport() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(accessor_lock_) {
if (dtls_srtp_transport_) {
// 就是这个
return dtls_srtp_transport_.get();
} else if (sdes_transport_) {
return sdes_transport_.get();
} else if (unencrypted_rtp_transport_) {
return unencrypted_rtp_transport_.get();
} else {
return nullptr;
}
}
24.
./pc/dtls_srtp_transport.cc
./pc/srtp_transport.cc
调用 DtlsSrtpTransport 的基类的函数
bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags)
return SendPacket(/*rtcp=*/false, packet, updated_options, flags);
25.
./pc/rtp_transport.cc
调用 SrtpTransport 的基类的函数
bool RtpTransport::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
int ret = transport->SendPacket(packet->cdata<char>(), packet->size(),
options, flags);
if (ret != static_cast<int>(packet->size())) {
if (transport->GetError() == ENOTCONN) {
RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport.";
SetReadyToSend(rtcp, false);
}
return false;
}
return true;
}
下面我们分析一下 rtp_packet_transport_ 是怎么过来的
上面我们在调用过程中，有这个函数
JsepTransportController::CreateDtlsSrtpTransport
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport, rtcp_dtls_transport);
./pc/dtls_srtp_transport.cc
void DtlsSrtpTransport::SetDtlsTransports(
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport)
SetRtpPacketTransport(rtp_dtls_transport);
void RtpTransport::SetRtpPacketTransport(
rtc::PacketTransportInternal* new_packet_transport)
rtp_packet_transport_ = new_packet_transport;
其实就是 DtlsTransport
26.
./p2p/base/dtls_transport.cc
int DtlsTransport::SendPacket(const char* data,
size_t size,
const rtc::PacketOptions& options,
int flags)
if (!dtls_active_) {
// Not doing DTLS.
return ice_transport_->SendPacket(data, size, options);
}
switch (dtls_state()) {
case DTLS_TRANSPORT_NEW:
// Can't send data until the connection is active.
// TODO(ekr@rtfm.com): assert here if dtls_ is NULL?
return -1;
case DTLS_TRANSPORT_CONNECTING:
// Can't send data until the connection is active.
return -1;
case DTLS_TRANSPORT_CONNECTED:
if (flags & PF_SRTP_BYPASS) {
RTC_DCHECK(!srtp_ciphers_.empty());
if (!IsRtpPacket(data, size)) {
return -1;
}
return ice_transport_->SendPacket(data, size, options);
} else {
return (dtls_->WriteAll(data, size, NULL, NULL) == rtc::SR_SUCCESS)
? static_cast<int>(size)
: -1;
}
case DTLS_TRANSPORT_FAILED:
case DTLS_TRANSPORT_CLOSED:
// Can't send anything when we're closed.
return -1;
default:
RTC_NOTREACHED();
return -1;
}
}
ice_transport_ 是初始化过来的对象
DtlsTransport::DtlsTransport(IceTransportInternal* ice_transport,
const webrtc::CryptoOptions& crypto_options,
webrtc::RtcEventLog* event_log)
: transport_name_(ice_transport->transport_name()),
component_(ice_transport->component()),
ice_transport_(ice_transport),
downward_(NULL),
srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()),
ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
crypto_options_(crypto_options),
event_log_(event_log) {
RTC_DCHECK(ice_transport_);
ConnectToIceTransport();
}
参看上面的流程 ice_transport_ 就是这个对象 P2PTransportChannel\
27.
./p2p/base/p2p_transport_channel.cc
int P2PTransportChannel::SendPacket(const char* data,
size_t len,
const rtc::PacketOptions& options,
int flags) {
RTC_DCHECK_RUN_ON(network_thread_);
if (flags != 0) {
error_ = EINVAL;
return -1;
}
// If we don't think the connection is working yet, return ENOTCONN
// instead of sending a packet that will probably be dropped.
if (!ReadyToSend(selected_connection_)) {
error_ = ENOTCONN;
return -1;
}
last_sent_packet_id_ = options.packet_id;
rtc::PacketOptions modified_options(options);
modified_options.info_signaled_after_sent.packet_type =
rtc::PacketType::kData;
int sent = selected_connection_->Send(data, len, modified_options);
if (sent <= 0) {
RTC_DCHECK(sent < 0);
error_ = selected_connection_->GetError();
}
return sent;
}
底层的 socket 数据会最终调用
P2PTransportChannel::OnReadPacket
MaybeSwitchSelectedConnection(connection, IceControllerEvent::DATA_RECEIVED);
bool P2PTransportChannel::MaybeSwitchSelectedConnection(IceControllerEvent reason,
IceControllerInterface::SwitchResult result)
SwitchSelectedConnection(const_cast<Connection*>(*result.connection), reason);
void P2PTransportChannel::SwitchSelectedConnection(Connection* conn,
IceControllerEvent reason)
selected_connection_ = conn;
// 其实就是这个地方产生的 Connection
bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
PortInterface* origin_port)
Connection* connection = port->CreateConnection(remote_candidate, origin);
./p2p/base/turn_port.cc
Connection* TurnPort::CreateConnection(const Candidate& remote_candidate, CandidateOrigin origin)
// 这个里面会创建 TurnEntry 下面的流程 29 会用到
if (CreateOrRefreshEntry(remote_candidate.address(), next_channel_number_, remote_candidate.username())) {
// An entry was created.
next_channel_number_++;
}
// 这个里面会创建 ProxyConnection 下面的流程 28 会用到
ProxyConnection* conn = new ProxyConnection(this, index, remote_candidate);
AddOrReplaceConnection(conn);
28.
./p2p/base/connection.cc
int ProxyConnection::Send(const void* data,
size_t size, const rtc::PacketOptions& options)
port_->SendTo(data, size, remote_candidate_.address(), options, true);
port_ 的是构造函数时，传递过来的，具体参见上面的 TurnPort::CreateConnection
// ProxyConnection 构造函数
ProxyConnection::ProxyConnection(Port* port, size_t index, const Candidate& remote_candidate) : Connection(port, index, remote_candidate) {}
// Connection 构造函数
Connection::Connection(Port* port, size_t index, const Candidate& remote_candidate)
: id_(rtc::CreateRandomId()), port_(port)
29.
./p2p/base/turn_port.cc
int TurnPort::SendTo(const void* data,
size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload)
TurnEntry* entry = FindEntry(addr);
int sent = entry->Send(data, size, payload, modified_options);
其实就是 entries_ 里的其中一个值，这个的创建，参见上面的 TurnPort::CreateConnection 过程如下：
bool TurnPort::CreateOrRefreshEntry(const rtc::SocketAddress& addr,
int channel_number,
const std::string& remote_ufrag)
entry = new TurnEntry(this, channel_number, addr, remote_ufrag);
entries_.push_back(entry);
return true;
30.
./p2p/base/turn_port.cc
int TurnEntry::Send(const void* data,
size_t size,
bool payload,
const rtc::PacketOptions& options)
// 这个 port_ 其实就是 TurnPort
return port_->Send(buf.Data(), buf.Length(), modified_options);
31.
./p2p/base/turn_port.cc
int TurnPort::Send(const void* data,
size_t len,
const rtc::PacketOptions& options) {
return socket_->SendTo(data, len, server_address_.address, options);
}
socket_ 是怎么创建的，这个还是比较复杂的具体参考博文 https://blog.csdn.net/freeabc/article/details/105659223
<<WebRtc 的初始化部分流程分析，包括 PeerConnection 对象的创建，采集与编码器的关联，以及怎么发送 sdp 到对方>>
./p2p/client/basic_port_allocator.cc
void AllocationSequence::OnMessage(rtc::Message* msg)
CreateRelayPorts();
void AllocationSequence::CreateRelayPorts()
CreateTurnPort(relay);
void AllocationSequence::CreateTurnPort(const RelayServerConfig& config)
CreateRelayPortArgs args;
args.network_thread = session_->network_thread();
args.socket_factory = session_->socket_factory();
args.network = network_;
args.username = session_->username();
args.password = session_->password();
args.server_address = &(*relay_port);
args.config = &config;
args.origin = session_->allocator()->origin();
args.turn_customizer = session_->allocator()->turn_customizer();
// session_ 就是 PortAllocatorSession
// allocator 就是 BasicPortAllocator
// relay_port_factory 分析参见 31.1
port = session_->allocator()->relay_port_factory()->Create(
args, session_->allocator()->min_port(),
session_->allocator()->max_port());
// 31.1 allocator 的产生过程如下，其实就是 TurnPortFactory
31.1.1
./pc/peer_connection_factory.cpp
PeerConnectionFactory::CreatePeerConnection
dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(default_network_manager_.get(),
packet_socket_factory,configuration.turn_customizer);
31.1.2
./p2p/client/basic_port_allocator.cc
BasicPortAllocator::BasicPortAllocator(
rtc::NetworkManager* network_manager,
rtc::PacketSocketFactory* socket_factory,
webrtc::TurnCustomizer* customizer,
RelayPortFactoryInterface* relay_port_factory)
: network_manager_(network_manager), socket_factory_(socket_factory)
InitRelayPortFactory(relay_port_factory);
31.1.3
./p2p/client/basic_port_allocator.cc
void BasicPortAllocator::InitRelayPortFactory(
RelayPortFactoryInterface* relay_port_factory) {
if (relay_port_factory != nullptr) {
relay_port_factory_ = relay_port_factory;
} else {
default_relay_port_factory_.reset(new TurnPortFactory());
relay_port_factory_ = default_relay_port_factory_.get();
}
}
}
// 31.2 TurnPort 的产生过程
./p2p/client/turn_port_factory.cc
std::unique_ptr<ort> TurnPortFactory::Create(const CreateRelayPortArgs& args,
int min_port,
int max_port) {
auto port = TurnPort::CreateUnique(
args.network_thread, args.socket_factory, args.network, min_port,
max_port, args.username, args.password, *args.server_address,
args.config->credentials, args.config->priority, args.origin,
args.config->tls_alpn_protocols, args.config->tls_elliptic_curves,
args.turn_customizer, args.config->tls_cert_verifier);
port->SetTlsCertPolicy(args.config->tls_cert_policy);
port->SetTurnLoggingId(args.config->turn_logging_id);
return std::move(port);
}
./p2p/base/turn_port.h
static std::unique_ptr<TurnPort> Create(
rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
uint16_t min_port,
uint16_t max_port,
const std::string& username, // ice username.
const std::string& password, // ice password.
const ProtocolAddress& server_address,
const RelayCredentials& credentials,
int server_priority,
const std::string& origin,
const std::vector<std::string>& tls_alpn_protocols,
const std::vector<std::string>& tls_elliptic_curves,
webrtc::TurnCustomizer* customizer,
rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr) {
// Using `new` to access a non-public constructor.
return absl::WrapUnique(
new TurnPort(thread, factory, network, min_port, max_port, username,
password, server_address, credentials, server_priority,
origin, tls_alpn_protocols, tls_elliptic_curves,
customizer, tls_cert_verifier));
}
到这里 TurnPort 就构建完毕，在程序的运行过程中，下面接口会被调用，并创建 TurnPort 的 socket_
参见博客 https://blog.csdn.net/freeabc/article/details/106000923 流程 7 的过程，7.6 章节
void TurnPort::PrepareAddress()
if (!CreateTurnClientSocket()) {
RTC_LOG(LS_ERROR) << "Failed to create TURN client socket";
OnAllocateError(STUN_ERROR_GLOBAL_FAILURE,
"Failed to create TURN client socket.");
return;
}
bool TurnPort::CreateTurnClientSocket()
socket_ = socket_factory()->CreateUdpSocket(
rtc::SocketAddress(Network()->GetBestIP(), 0), min_port(), max_port());
其实就是一个 AsyncUDPSocket 参见博文 https://blog.csdn.net/freeabc/article/details/106000923
AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket
32.
./rtc_base/async_udp_socket.cc
int AsyncUDPSocket::Send(const void* pv,
size_t cb,
const rtc::PacketOptions& options) {
rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
options.info_signaled_after_sent);
CopySocketInformationToPacketInfo(cb, *this, false, &sent_packet.info);
int ret = socket_->Send(pv, cb);
SignalSentPacket(this, sent_packet);
return ret;
}
socket_ 这个是 AsyncUDPSocket 初始化过来的, 根据博客 https://blog.csdn.net/freeabc/article/details/106000923
AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket
AsyncSocket* socket = socket_factory()->CreateAsyncSocket(address.family(), SOCK_DGRAM);
就是 SocketDispatcher 对象
33.
./rtc_base/physical_socket_server.cc
调用 SocketDispatcher 的基类函数
int PhysicalSocket::Send(const void* pv, size_t cb)
int sent = DoSend(
s_, reinterpret_cast<const char*>(pv), static_cast<int>(cb),
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
// Suppress SIGPIPE. Without this, attempting to send on a socket whose
// other end is closed will result in a SIGPIPE signal being raised to
// our process, which by default will terminate the process, which we
// don't want. By specifying this flag, we'll just get the error EPIPE
// instead and can handle the error gracefully.
MSG_NOSIGNAL
#else
0
#endif
);
34. 到这里就彻底发送出去了，调用了系统的 send 函数。。。。。。
int PhysicalSocket::DoSend(SOCKET socket, const char* buf, int len, int flags) {
return ::send(socket, buf, len, flags);
}
//----------------------------------------------------------------------------------------------
//
// 视频编码器的速率调整
//
//----------------------------------------------------------------------------------------------
1. 把 VideoSendStreamImpl 注册到 Call 对象的 bitrate_allocator_ 里
bitrate_allocator_ 就是 Call::Call 里的 bitrate_allocator_(new BitrateAllocator(this))
VideoSendStreamImpl::StartupVideoSendStream()
bitrate_allocator_->AddObserver(this, GetAllocationConfig());
2. 这样 bitrate_allocator_ 有更新就会调用 下面的函数
这个函数内更改编码速率
VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update)
video_stream_encoder_->OnBitrateUpdated(
encoder_target_rate, encoder_stable_target_rate, link_allocation,
rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
update.round_trip_time.ms());
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_)
3. bitrate_allocator_ 的更改，来源与 下面流程
首先 Call 对象注册到 RtpTransportControllerSend
void Call::RegisterRateObserver()
transport_send_ptr_->RegisterTargetTransferRateObserver(this);
4. 如果 RtpTransportControllerSend
./call/rtp_transport_controller_send.cc
RtpTransportControllerSend::UpdateControlState() --->
有传输速率更新则会调用
void Call::OnTargetTransferRate(TargetTransferRate msg)
bitrate_allocator_->OnNetworkEstimateChanged(msg);
5.
BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg)
uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
这个地方就是改变编码器的速率的地方
6. 步骤 2 里的
来自 VideoSendStream::VideoSendStream 里的
video_stream_encoder_ = CreateVideoStreamEncoder(clock, task_queue_factory, num_cpu_cores, &stats_proxy_, config_.encoder_settings);
就是 VideoStreamEncoder 对象
./api/video/video_stream_encoder_create.cc
7. 步骤 2 的调用 VideoStreamEncoder::OnBitrateUpdated
if (encoder_) {
encoder_->OnPacketLossRateUpdate(static_cast<float>(fraction_lost) / 256.f);
encoder_->OnRttUpdate(round_trip_time_ms);
}
SetEncoderRates(UpdateBitrateAllocationAndNotifyObserver(new_rate_settings));
VideoStreamEncoder::ReconfigureEncoder() 里创建了 encoder_
这个 encoder_ 目前就是 VideoEncoderWrapper，这个是上层的一个硬编码代理，并没有实现
接口
// Inform the encoder when the packet loss rate changes.
//
// Input: - packet_loss_rate : The packet loss rate (0.0 to 1.0).
virtual void OnPacketLossRateUpdate(float packet_loss_rate);
// Inform the encoder when the round trip time changes.
//
// Input: - rtt_ms : The new RTT, in milliseconds.
virtual void OnRttUpdate(int64_t rtt_ms);
8.
void VideoStreamEncoder::SetEncoderRates(
const EncoderRateSettings& rate_settings)
encoder_->SetRates(rate_settings.rate_control);
9.
void VideoEncoderWrapper::SetRates(const RateControlParameters& parameters)
ScopedJavaLocalRef<jobject> j_bitrate_allocation = ToJavaBitrateAllocation(jni, parameters.bitrate);
ScopedJavaLocalRef<jobject> ret = Java_VideoEncoder_setRateAllocation(jni, encoder_, j_bitrate_allocation,
(jint)(parameters.framerate_fps + 0.5));
HandleReturnCode(jni, ret, "setRateAllocation");
10. Java
// 到这里就进行了速率调整。。。。。
public VideoCodecStatus HardwareVideoEncoder::setRateAllocation(BitrateAllocation bitrateAllocation, int framerate) {
encodeThreadChecker.checkIsOnValidThread();
if (framerate > MAX_VIDEO_FRAMERATE) {
framerate = MAX_VIDEO_FRAMERATE;
}
bitrateAdjuster.setTargets(bitrateAllocation.getSum(), framerate);
return VideoCodecStatus.OK;
}
//----------------------------------------------------------------------------------------------
//
// WebRtcVideoChannel 对象 AddSendStream (WebRtcVideoSendStream send_streams_) 的过程
//
//----------------------------------------------------------------------------------------------
1. Java
CallActivity::onConnectedToRoomInternal
peerConnectionClient.createOffer()
2. Java
public void PeerConnectionClient::createOffer() {
executor.execute(() -> {
if (peerConnection != null && !isError) {
Log.d(TAG, "C Create OFFER");
isInitiator = true;
peerConnection.createOffer(sdpObserver, sdpMediaConstraints);
}
});
}
3. Java
public void PeerConnection::createOffer(SdpObserver observer, MediaConstraints constraints) {
nativeCreateOffer(observer, constraints);
}
4. C++
void Java_org_webrtc_PeerConnection_nativeCreateOffer(JNIEnv* env, jobject jcaller, jobject observer, jobject constraints) {
return JNI_PeerConnection_CreateOffer(env, base::android::JavaParamRef<jobject>(env, jcaller),
base::android::JavaParamRef<jobject>(env, observer), base::android::JavaParamRef<jobject>(env,
constraints));
}
5. ./sdk/android/src/jni/pc/peer_connection.cc
static void JNI_PeerConnection_CreateOffer(JNIEnv* jni, const JavaParamRef<jobject>& j_pc, const JavaParamRef<jobject>& j_observer,
const JavaParamRef<jobject>& j_constraints) {
std::unique_ptr<MediaConstraints> constraints = JavaToNativeMediaConstraints(jni, j_constraints);
rtc::scoped_refptr<CreateSdpObserverJni> observer(new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer, std::move(constraints)));
PeerConnectionInterface::RTCOfferAnswerOptions options;
CopyConstraintsIntoOfferAnswerOptions(observer->constraints(), &options);
ExtractNativePC(jni, j_pc)->CreateOffer(observer, options);
}
6. ./pc/peer_connection.cc
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer, const RTCOfferAnswerOptions& options)
this_weak_ptr->DoCreateOffer(options, observer_wrapper);
7. ./pc/peer_connection.cc
void PeerConnection::DoCreateOffer(const RTCOfferAnswerOptions& options, rtc::scoped_refptr<CreateSessionDescriptionObserver> observer)
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
参见 PeerConnection::Initialize
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory)......
8. ./pc/webrtc_session_description_factory.cc
void WebRtcSessionDescriptionFactory::CreateOffer(CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options, const cricket::MediaSessionOptions& session_options)
InternalCreateOffer(request);
9. ./pc/webrtc_session_description_factory.cc
void WebRtcSessionDescriptionFactory::InternalCreateOffer(CreateSessionDescriptionRequest request)
std::unique_ptr<cricket::SessionDescription> desc = session_desc_factory_.CreateOffer(
request.options, pc_->local_description() ? pc_->local_description()->description() : nullptr);
9.1. ./pc/media_session.cc
MediaSessionDescriptionFactory::CreateOffer
这里是异步调用，创建成功则，调用
auto offer = std::make_unique<JsepSessionDescription>(SdpType::kOffer, std::move(desc), session_id_, rtc::ToString(session_version_++));
CopyCandidatesFromSessionDescription(pc_->local_description(), options.mid, offer.get());
PostCreateSessionDescriptionSucceeded(request.observer, std::move(offer));
10. ./pc/webrtc_session_description_factory.cc
WebRtcSessionDescriptionFactory::OnMessage(rtc::Message* msg)
10.1. 失败
param->observer->OnFailure(std::move(param->error))
10.2. 成功
param->observer->OnSuccess(param->description.release());
11. Java
SDPObserver::onCreateSuccess(final SessionDescription origSdp)
String sdpDescription = origSdp.description;
if (preferIsac) {
sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);
}
if (isVideoCallEnabled()) {
sdpDescription = preferCodec(sdpDescription, getSdpVideoCodecName(peerConnectionParameters), false);
}
final SessionDescription sdp = new SessionDescription(origSdp.type, sdpDescription);
localSdp = sdp;
executor.execute(() -> {
if (peerConnection != null && !isError) {
Log.d(TAG, "Set local SDP from " + sdp.type);
peerConnection.setLocalDescription(sdpObserver, sdp);
}
});
参看流程 2 传过来的 SDPObserver
12. Java
PeerConnection::setLocalDescription(SdpObserver observer, SessionDescription sdp) {
nativeSetLocalDescription(observer, sdp);
}
13. C++
Java_org_webrtc_PeerConnection_nativeSetLocalDescription
14. ./sdk/android/src/jni/pc/peer_connection.cc
JNI_PeerConnection_SetLocalDescription
15. ./pc/peer_connection.cc
PeerConnection::SetLocalDescription(SetSessionDescriptionObserver* observer, SessionDescriptionInterface* desc_ptr)
this_weak_ptr->DoSetLocalDescription(std::move(desc), std::move(observer_refptr));
16.
void PeerConnection::DoSetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetSessionDescriptionObserver> observer)
error = ApplyLocalDescription(std::move(desc));
17.
PeerConnection::ApplyLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc)
error = UpdateSessionState(type, cricket::CS_LOCAL, local_description()->description());
18.
RTCError PeerConnection::UpdateSessionState(SdpType type, cricket::ContentSource source,
const cricket::SessionDescription* description)
error = PushdownMediaDescription(type, source);
19.
RTCError PeerConnection::PushdownMediaDescription(SdpType type, cricket::ContentSource source)
for (const auto& transceiver : transceivers_) {
const ContentInfo* content_info = FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
const MediaContentDescription* content_desc = content_info->media_description();
bool success = (source == cricket::CS_LOCAL)
? channel->SetLocalContent(content_desc, type, &error)
: channel->SetRemoteContent(content_desc, type, &error);
transceivers_ 在 PeerConnection::Initialize 的过程中，进行初始化，就是 RtpTransceiver !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
bool PeerConnection::Initialize(const PeerConnectionInterface::RTCConfiguration& configuration, PeerConnectionDependencies dependencies)
transceivers_.push_back(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
transceivers_.push_back(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
在函数 CreateChannels 中为 transceiver 设置 VideoChannel 对象
RTCError PeerConnection::CreateChannels(const SessionDescription& desc)
cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
if (!video_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, "Failed to create video channel.");
}
GetVideoTransceiver()->internal()->SetChannel(video_channel);
20.
这是 VideoChannel 的父方法
bool BaseChannel::SetLocalContent(const MediaContentDescription* content, SdpType type, std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&BaseChannel::SetLocalContent_w, this, content, type, error_desc));
}
21.
VideoChannel::SetLocalContent_w(const MediaContentDescription* content, SdpType type, std::string* error_desc)
const VideoContentDescription* video = content->as_video();
if (!UpdateLocalStreams_w(video->streams(), type, error_desc))
22.
这是 VideoChannel 的父方法
bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams, SdpType type, std::string* error_desc)
if (media_channel()->AddSendStream(new_stream))
media_channel 就是 WebRtcVideoChannel 对象，具体参看下面 VideoChannel 的产生流程
22.1 PeerConnection 产生 VideoChannel 对象
cricket::VideoChannel* PeerConnection::CreateVideoChannel(const std::string& mid) {
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
MediaTransportConfig media_transport_config = transport_controller_->GetMediaTransportConfig(mid);
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_ptr_, configuration_.media_config, rtp_transport,
media_transport_config, signaling_thread(), mid, SrtpRequired(),
GetCryptoOptions(), &ssrc_generator_, video_options_,
video_bitrate_allocator_factory_.get());
if (!video_channel) {
return nullptr;
}
video_channel->SignalDtlsSrtpSetupFailure.connect(this, &eerConnection::OnDtlsSrtpSetupFailure);
video_channel->SignalSentPacket.connect(this, &PeerConnection::OnSentPacket_w);
video_channel->SetRtpTransport(rtp_transport);
return video_channel;
}
22.2 ChannelManager 产生 VideoChannel 对象
VideoChannel* ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const VideoOptions& options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
return CreateVideoChannel(call, media_config, rtp_transport, media_transport_config,
signaling_thread, content_name, srtp_required, crypto_options, ssrc_generator, options, video_bitrate_allocator_factory);
});
}
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(initialized_);
RTC_DCHECK(call);
if (!media_engine_) {
return nullptr;
}
// 这个地方创建了 WebRtcVideoChannel 对象!!!!!!!!!!!!!!!
VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel(call, media_config, options, crypto_options, video_bitrate_allocator_factory);
if (!media_channel) {
return nullptr;
}
auto video_channel = std::make_unique<VideoChannel>(worker_thread_, network_thread_, signaling_thread, absl::WrapUnique(media_channel), content_name,
srtp_required, crypto_options, ssrc_generator);
video_channel->Init_w(rtp_transport, media_transport_config);
VideoChannel* video_channel_ptr = video_channel.get();
video_channels_.push_back(std::move(video_channel));
return video_channel_ptr;
}
22.3 media_engine_->video().CreateMediaChannel 函数实现 WebRtcVideoChannel!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel(webrtc::Call* call, const MediaConfig& config, const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString();
return new WebRtcVideoChannel(call, config, options, crypto_options, encoder_factory_.get(), decoder_factory_.get(), video_bitrate_allocator_factory);
}
23.
bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
if (!ValidateStreamParams(sp))
return false;
if (!ValidateSendSsrcAvailability(sp))
return false;
for (uint32_t used_ssrc : sp.ssrcs)
send_ssrcs_.insert(used_ssrc);
webrtc::VideoSendStream::Config config(this);
for (const RidDescription& rid : sp.rids()) {
config.rtp.rids.push_back(rid.rid);
}
config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate;
config.periodic_alr_bandwidth_probing = video_config_.periodic_alr_bandwidth_probing;
config.encoder_settings.experiment_cpu_load_estimator = video_config_.experiment_cpu_load_estimator;
config.encoder_settings.encoder_factory = encoder_factory_;
config.encoder_settings.bitrate_allocator_factory = bitrate_allocator_factory_;
config.encoder_settings.encoder_switch_request_callback = this;
config.crypto_options = crypto_options_;
config.rtp.extmap_allow_mixed = ExtmapAllowMixed();
config.rtcp_report_interval_ms = video_config_.rtcp_report_interval_ms;
WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
call_, sp, std::move(config), default_send_options_,
video_config_.enable_cpu_adaptation, bitrate_config_.max_bitrate_bps,
send_codec_, send_rtp_extensions_, send_params_);
send_streams_[ssrc] = stream;
至此 WebRtcVideoChannel 中的 send_streams_ 对象产生, 就是 WebRtcVideoSendStream 流
24.
void WebRtcVideoChannel::WebRtcVideoSendStream::SetSend(bool send) {
RTC_DCHECK_RUN_ON(&thread_checker_);
sending_ = send;
UpdateSendState();
}
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发表于 2020-5-23 20:42:53
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