| /* |
| * 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_factory.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_(rtc::make_ref_counted<VideoRtpTrackSource>(&source_callback_)), |
| track_(VideoTrackProxyWithInternal<VideoTrack>::Create( |
| rtc::Thread::Current(), |
| worker_thread, |
| VideoTrack::Create(receiver_id, |
| CreateVideoTrackSourceProxy(rtc::Thread::Current(), |
| worker_thread, |
| source_), |
| worker_thread))), |
| attachment_id_(GenerateUniqueId()) { |
| RTC_DCHECK(worker_thread_); |
| SetStreams(streams); |
| RTC_DCHECK_EQ(source_->state(), MediaSourceInterface::kLive); |
| } |
| |
| VideoRtpReceiver::~VideoRtpReceiver() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RTC_DCHECK(stopped_); |
| RTC_DCHECK(!media_channel_); |
| } |
| |
| std::vector<std::string> VideoRtpReceiver::stream_ids() const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| 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; |
| } |
| |
| rtc::scoped_refptr<DtlsTransportInterface> VideoRtpReceiver::dtls_transport() |
| const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| return dtls_transport_; |
| } |
| |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>> |
| VideoRtpReceiver::streams() const { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| return streams_; |
| } |
| |
| RtpParameters VideoRtpReceiver::GetParameters() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!media_channel_) |
| return RtpParameters(); |
| return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_) |
| : media_channel_->GetDefaultRtpReceiveParameters(); |
| } |
| |
| void VideoRtpReceiver::SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| frame_decryptor_ = std::move(frame_decryptor); |
| // Special Case: Set the frame decryptor to any value on any existing channel. |
| if (media_channel_ && ssrc_) { |
| media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); |
| } |
| } |
| |
| rtc::scoped_refptr<FrameDecryptorInterface> |
| VideoRtpReceiver::GetFrameDecryptor() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| return frame_decryptor_; |
| } |
| |
| void VideoRtpReceiver::SetDepacketizerToDecoderFrameTransformer( |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| frame_transformer_ = std::move(frame_transformer); |
| if (media_channel_) { |
| media_channel_->SetDepacketizerToDecoderFrameTransformer( |
| ssrc_.value_or(0), frame_transformer_); |
| } |
| } |
| |
| void VideoRtpReceiver::Stop() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| // TODO(deadbeef): Need to do more here to fully stop receiving packets. |
| |
| if (!stopped_) { |
| source_->SetState(MediaSourceInterface::kEnded); |
| stopped_ = true; |
| } |
| |
| worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (media_channel_) { |
| SetSink(nullptr); |
| SetMediaChannel_w(nullptr); |
| } |
| source_->ClearCallback(); |
| }); |
| } |
| |
| void VideoRtpReceiver::StopAndEndTrack() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| Stop(); |
| track_->internal()->set_ended(); |
| } |
| |
| void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| |
| // `stopped_` will be `true` on construction. RestartMediaChannel |
| // can in this case function like "ensure started" and flip `stopped_` |
| // to false. |
| |
| // TODO(tommi): Can we restart the media channel without blocking? |
| bool ok = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&, was_stopped = |
| stopped_] { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!media_channel_) { |
| // Ignore further negotiations if we've already been stopped and don't |
| // have an associated media channel. |
| RTC_DCHECK(was_stopped); |
| return false; // Can't restart. |
| } |
| |
| if (!was_stopped && ssrc_ == ssrc) { |
| // Already running with that ssrc. |
| return true; |
| } |
| |
| // Disconnect from the previous ssrc. |
| if (!was_stopped) { |
| SetSink(nullptr); |
| } |
| |
| bool encoded_sink_enabled = saved_encoded_sink_enabled_; |
| SetEncodedSinkEnabled(false); |
| |
| // Set up the new ssrc. |
| ssrc_ = std::move(ssrc); |
| SetSink(source_->sink()); |
| if (encoded_sink_enabled) { |
| SetEncodedSinkEnabled(true); |
| } |
| |
| if (frame_transformer_ && media_channel_) { |
| media_channel_->SetDepacketizerToDecoderFrameTransformer( |
| ssrc_.value_or(0), frame_transformer_); |
| } |
| |
| if (media_channel_ && ssrc_) { |
| if (frame_decryptor_) { |
| media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); |
| } |
| |
| media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs()); |
| } |
| |
| return true; |
| }); |
| |
| if (!ok) |
| return; |
| |
| stopped_ = false; |
| } |
| |
| // RTC_RUN_ON(worker_thread_) |
| void VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) { |
| if (ssrc_) { |
| media_channel_->SetSink(*ssrc_, sink); |
| } else { |
| media_channel_->SetDefaultSink(sink); |
| } |
| } |
| |
| void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RestartMediaChannel(ssrc); |
| } |
| |
| void VideoRtpReceiver::SetupUnsignaledMediaChannel() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RestartMediaChannel(absl::nullopt); |
| } |
| |
| uint32_t VideoRtpReceiver::ssrc() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| return ssrc_.value_or(0); |
| } |
| |
| void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| SetStreams(CreateStreamsFromIds(std::move(stream_ids))); |
| } |
| |
| void VideoRtpReceiver::set_transport( |
| rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| dtls_transport_ = std::move(dtls_transport); |
| } |
| |
| void VideoRtpReceiver::SetStreams( |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| // 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) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| 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) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| delay_.Set(delay_seconds); |
| if (media_channel_ && ssrc_) |
| media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs()); |
| } |
| |
| void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| RTC_DCHECK(media_channel == nullptr || |
| media_channel->media_type() == media_type()); |
| |
| if (stopped_ && !media_channel) |
| return; |
| |
| worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| SetMediaChannel_w(media_channel); |
| }); |
| } |
| |
| // RTC_RUN_ON(worker_thread_) |
| void VideoRtpReceiver::SetMediaChannel_w(cricket::MediaChannel* media_channel) { |
| if (media_channel == media_channel_) |
| return; |
| |
| 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() { |
| RTC_DCHECK_RUN_ON(&signaling_thread_checker_); |
| if (observer_) { |
| observer_->OnFirstPacketReceived(media_type()); |
| } |
| received_first_packet_ = true; |
| } |
| |
| std::vector<RtpSource> VideoRtpReceiver::GetSources() const { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (!ssrc_ || !media_channel_) |
| return std::vector<RtpSource>(); |
| 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; |
| } |
| |
| // RTC_RUN_ON(worker_thread_) |
| void VideoRtpReceiver::SetEncodedSinkEnabled(bool enable) { |
| if (!media_channel_) |
| return; |
| |
| // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC |
| const auto ssrc = ssrc_.value_or(0); |
| |
| if (enable) { |
| media_channel_->SetRecordableEncodedFrameCallback( |
| ssrc, [source = source_](const RecordableEncodedFrame& frame) { |
| source->BroadcastRecordableEncodedFrame(frame); |
| }); |
| } else { |
| media_channel_->ClearRecordableEncodedFrameCallback(ssrc); |
| } |
| } |
| |
| } // namespace webrtc |