/* * Copyright 2019 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "pc/video_rtp_receiver.h" #include #include #include #include "api/media_stream_proxy.h" #include "api/video_track_source_proxy.h" #include "pc/jitter_buffer_delay.h" #include "pc/jitter_buffer_delay_proxy.h" #include "pc/media_stream.h" #include "pc/video_track.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" #include "rtc_base/logging.h" #include "rtc_base/trace_event.h" namespace webrtc { VideoRtpReceiver::VideoRtpReceiver(rtc::Thread* worker_thread, std::string receiver_id, std::vector stream_ids) : VideoRtpReceiver(worker_thread, receiver_id, CreateStreamsFromIds(std::move(stream_ids))) {} VideoRtpReceiver::VideoRtpReceiver( rtc::Thread* worker_thread, const std::string& receiver_id, const std::vector>& streams) : worker_thread_(worker_thread), id_(receiver_id), source_(new RefCountedObject(this)), track_(VideoTrackProxyWithInternal::Create( rtc::Thread::Current(), worker_thread, VideoTrack::Create( receiver_id, VideoTrackSourceProxy::Create(rtc::Thread::Current(), worker_thread, source_), worker_thread))), attachment_id_(GenerateUniqueId()), delay_(JitterBufferDelayProxy::Create( rtc::Thread::Current(), worker_thread, new rtc::RefCountedObject(worker_thread))) { RTC_DCHECK(worker_thread_); SetStreams(streams); source_->SetState(MediaSourceInterface::kLive); } VideoRtpReceiver::~VideoRtpReceiver() { // Since cricket::VideoRenderer is not reference counted, // we need to remove it from the channel before we are deleted. Stop(); // Make sure we can't be called by the |source_| anymore. worker_thread_->Invoke(RTC_FROM_HERE, [this] { source_->ClearCallback(); }); } std::vector VideoRtpReceiver::stream_ids() const { std::vector stream_ids(streams_.size()); for (size_t i = 0; i < streams_.size(); ++i) stream_ids[i] = streams_[i]->id(); return stream_ids; } RtpParameters VideoRtpReceiver::GetParameters() const { if (!media_channel_ || stopped_) { return RtpParameters(); } return worker_thread_->Invoke(RTC_FROM_HERE, [&] { return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_) : media_channel_->GetDefaultRtpReceiveParameters(); }); } void VideoRtpReceiver::SetFrameDecryptor( rtc::scoped_refptr frame_decryptor) { frame_decryptor_ = std::move(frame_decryptor); // Special Case: Set the frame decryptor to any value on any existing channel. if (media_channel_ && ssrc_.has_value() && !stopped_) { worker_thread_->Invoke(RTC_FROM_HERE, [&] { media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); }); } } rtc::scoped_refptr VideoRtpReceiver::GetFrameDecryptor() const { return frame_decryptor_; } void VideoRtpReceiver::SetDepacketizerToDecoderFrameTransformer( rtc::scoped_refptr frame_transformer) { worker_thread_->Invoke(RTC_FROM_HERE, [&] { RTC_DCHECK_RUN_ON(worker_thread_); frame_transformer_ = std::move(frame_transformer); if (media_channel_ && !stopped_) { media_channel_->SetDepacketizerToDecoderFrameTransformer( ssrc_.value_or(0), frame_transformer_); } }); } void VideoRtpReceiver::Stop() { // TODO(deadbeef): Need to do more here to fully stop receiving packets. if (stopped_) { return; } source_->SetState(MediaSourceInterface::kEnded); if (!media_channel_) { RTC_LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists."; } else { // Allow that SetSink fails. This is the normal case when the underlying // media channel has already been deleted. worker_thread_->Invoke(RTC_FROM_HERE, [&] { RTC_DCHECK_RUN_ON(worker_thread_); SetSink(nullptr); }); } delay_->OnStop(); stopped_ = true; } void VideoRtpReceiver::StopAndEndTrack() { Stop(); track_->internal()->set_ended(); } void VideoRtpReceiver::RestartMediaChannel(absl::optional ssrc) { RTC_DCHECK(media_channel_); if (!stopped_ && ssrc_ == ssrc) { return; } worker_thread_->Invoke(RTC_FROM_HERE, [&] { RTC_DCHECK_RUN_ON(worker_thread_); if (!stopped_) { SetSink(nullptr); } bool encoded_sink_enabled = saved_encoded_sink_enabled_; SetEncodedSinkEnabled(false); stopped_ = false; ssrc_ = ssrc; SetSink(source_->sink()); if (encoded_sink_enabled) { SetEncodedSinkEnabled(true); } if (frame_transformer_ && media_channel_) { media_channel_->SetDepacketizerToDecoderFrameTransformer( ssrc_.value_or(0), frame_transformer_); } }); // Attach any existing frame decryptor to the media channel. MaybeAttachFrameDecryptorToMediaChannel( ssrc, worker_thread_, frame_decryptor_, media_channel_, stopped_); // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC // value. delay_->OnStart(media_channel_, ssrc.value_or(0)); } void VideoRtpReceiver::SetSink(rtc::VideoSinkInterface* sink) { RTC_DCHECK(media_channel_); if (ssrc_) { media_channel_->SetSink(*ssrc_, sink); return; } media_channel_->SetDefaultSink(sink); } void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) { if (!media_channel_) { RTC_LOG(LS_ERROR) << "VideoRtpReceiver::SetupMediaChannel: No video channel exists."; } RestartMediaChannel(ssrc); } void VideoRtpReceiver::SetupUnsignaledMediaChannel() { if (!media_channel_) { RTC_LOG(LS_ERROR) << "VideoRtpReceiver::SetupUnsignaledMediaChannel: No " "video channel exists."; } RestartMediaChannel(absl::nullopt); } void VideoRtpReceiver::set_stream_ids(std::vector stream_ids) { SetStreams(CreateStreamsFromIds(std::move(stream_ids))); } void VideoRtpReceiver::SetStreams( const std::vector>& streams) { // Remove remote track from any streams that are going away. for (const auto& existing_stream : streams_) { bool removed = true; for (const auto& stream : streams) { if (existing_stream->id() == stream->id()) { RTC_DCHECK_EQ(existing_stream.get(), stream.get()); removed = false; break; } } if (removed) { existing_stream->RemoveTrack(track_); } } // Add remote track to any streams that are new. for (const auto& stream : streams) { bool added = true; for (const auto& existing_stream : streams_) { if (stream->id() == existing_stream->id()) { RTC_DCHECK_EQ(stream.get(), existing_stream.get()); added = false; break; } } if (added) { stream->AddTrack(track_); } } streams_ = streams; } void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) { observer_ = observer; // Deliver any notifications the observer may have missed by being set late. if (received_first_packet_ && observer_) { observer_->OnFirstPacketReceived(media_type()); } } void VideoRtpReceiver::SetJitterBufferMinimumDelay( absl::optional delay_seconds) { delay_->Set(delay_seconds); } void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) { RTC_DCHECK(media_channel == nullptr || media_channel->media_type() == media_type()); worker_thread_->Invoke(RTC_FROM_HERE, [&] { RTC_DCHECK_RUN_ON(worker_thread_); bool encoded_sink_enabled = saved_encoded_sink_enabled_; if (encoded_sink_enabled && media_channel_) { // Turn off the old sink, if any. SetEncodedSinkEnabled(false); } media_channel_ = static_cast(media_channel); if (media_channel_) { if (saved_generate_keyframe_) { // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC media_channel_->GenerateKeyFrame(ssrc_.value_or(0)); saved_generate_keyframe_ = false; } if (encoded_sink_enabled) { SetEncodedSinkEnabled(true); } if (frame_transformer_) { media_channel_->SetDepacketizerToDecoderFrameTransformer( ssrc_.value_or(0), frame_transformer_); } } }); } void VideoRtpReceiver::NotifyFirstPacketReceived() { if (observer_) { observer_->OnFirstPacketReceived(media_type()); } received_first_packet_ = true; } std::vector VideoRtpReceiver::GetSources() const { if (!media_channel_ || !ssrc_ || stopped_) { return {}; } return worker_thread_->Invoke>( RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); }); } void VideoRtpReceiver::OnGenerateKeyFrame() { RTC_DCHECK_RUN_ON(worker_thread_); if (!media_channel_) { RTC_LOG(LS_ERROR) << "VideoRtpReceiver::OnGenerateKeyFrame: No video channel exists."; return; } // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC media_channel_->GenerateKeyFrame(ssrc_.value_or(0)); // We need to remember to request generation of a new key frame if the media // channel changes, because there's no feedback whether the keyframe // generation has completed on the channel. saved_generate_keyframe_ = true; } void VideoRtpReceiver::OnEncodedSinkEnabled(bool enable) { RTC_DCHECK_RUN_ON(worker_thread_); SetEncodedSinkEnabled(enable); // Always save the latest state of the callback in case the media_channel_ // changes. saved_encoded_sink_enabled_ = enable; } void VideoRtpReceiver::SetEncodedSinkEnabled(bool enable) { if (media_channel_) { if (enable) { // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC auto source = source_; media_channel_->SetRecordableEncodedFrameCallback( ssrc_.value_or(0), [source = std::move(source)](const RecordableEncodedFrame& frame) { source->BroadcastRecordableEncodedFrame(frame); }); } else { // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC media_channel_->ClearRecordableEncodedFrameCallback(ssrc_.value_or(0)); } } } } // namespace webrtc