| /* | 
 |  *  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 <stddef.h> | 
 |  | 
 | #include <utility> | 
 | #include <vector> | 
 |  | 
 | #include "api/video/recordable_encoded_frame.h" | 
 | #include "api/video_track_source_proxy.h" | 
 | #include "pc/jitter_buffer_delay.h" | 
 | #include "pc/jitter_buffer_delay_proxy.h" | 
 | #include "pc/video_track.h" | 
 | #include "rtc_base/checks.h" | 
 | #include "rtc_base/location.h" | 
 | #include "rtc_base/logging.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | VideoRtpReceiver::VideoRtpReceiver(rtc::Thread* worker_thread, | 
 |                                    std::string receiver_id, | 
 |                                    std::vector<std::string> 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<rtc::scoped_refptr<MediaStreamInterface>>& streams) | 
 |     : worker_thread_(worker_thread), | 
 |       id_(receiver_id), | 
 |       source_(new RefCountedObject<VideoRtpTrackSource>(this)), | 
 |       track_(VideoTrackProxyWithInternal<VideoTrack>::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<JitterBufferDelay>(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<void>(RTC_FROM_HERE, | 
 |                                [this] { source_->ClearCallback(); }); | 
 | } | 
 |  | 
 | std::vector<std::string> VideoRtpReceiver::stream_ids() const { | 
 |   std::vector<std::string> 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<RtpParameters>(RTC_FROM_HERE, [&] { | 
 |     return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_) | 
 |                  : media_channel_->GetDefaultRtpReceiveParameters(); | 
 |   }); | 
 | } | 
 |  | 
 | void VideoRtpReceiver::SetFrameDecryptor( | 
 |     rtc::scoped_refptr<FrameDecryptorInterface> 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<void>(RTC_FROM_HERE, [&] { | 
 |       media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); | 
 |     }); | 
 |   } | 
 | } | 
 |  | 
 | rtc::scoped_refptr<FrameDecryptorInterface> | 
 | VideoRtpReceiver::GetFrameDecryptor() const { | 
 |   return frame_decryptor_; | 
 | } | 
 |  | 
 | void VideoRtpReceiver::SetDepacketizerToDecoderFrameTransformer( | 
 |     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) { | 
 |   worker_thread_->Invoke<void>(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<void>(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<uint32_t> ssrc) { | 
 |   RTC_DCHECK(media_channel_); | 
 |   if (!stopped_ && ssrc_ == ssrc) { | 
 |     return; | 
 |   } | 
 |   worker_thread_->Invoke<void>(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<VideoFrame>* 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<std::string> stream_ids) { | 
 |   SetStreams(CreateStreamsFromIds(std::move(stream_ids))); | 
 | } | 
 |  | 
 | void VideoRtpReceiver::SetStreams( | 
 |     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& 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<double> 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<void>(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<cricket::VideoMediaChannel*>(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<RtpSource> VideoRtpReceiver::GetSources() const { | 
 |   if (!media_channel_ || !ssrc_ || stopped_) { | 
 |     return {}; | 
 |   } | 
 |   return worker_thread_->Invoke<std::vector<RtpSource>>( | 
 |       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 |