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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/video_rtp_receiver.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/video_rtp_receiver.cc')
-rw-r--r-- | third_party/libwebrtc/pc/video_rtp_receiver.cc | 383 |
1 files changed, 383 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/video_rtp_receiver.cc b/third_party/libwebrtc/pc/video_rtp_receiver.cc new file mode 100644 index 0000000000..8a2e65c162 --- /dev/null +++ b/third_party/libwebrtc/pc/video_rtp_receiver.cc @@ -0,0 +1,383 @@ +/* + * 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 <string> +#include <utility> +#include <vector> + +#include "api/video/recordable_encoded_frame.h" +#include "pc/video_track.h" +#include "rtc_base/checks.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, source_, worker_thread))), + attachment_id_(GenerateUniqueId()) { + RTC_DCHECK(worker_thread_); + SetStreams(streams); + RTC_DCHECK_EQ(source_->state(), MediaSourceInterface::kInitializing); +} + +VideoRtpReceiver::~VideoRtpReceiver() { + RTC_DCHECK_RUN_ON(&signaling_thread_checker_); + 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(); + auto current_ssrc = ssrc(); + return current_ssrc.has_value() + ? media_channel_->GetRtpReceiveParameters(current_ssrc.value()) + : 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_ && signaled_ssrc_) { + media_channel_->SetFrameDecryptor(*signaled_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( + signaled_ssrc_.value_or(0), frame_transformer_); + } +} + +void VideoRtpReceiver::Stop() { + RTC_DCHECK_RUN_ON(&signaling_thread_checker_); + source_->SetState(MediaSourceInterface::kEnded); + track_->internal()->set_ended(); +} + +void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) { + RTC_DCHECK_RUN_ON(&signaling_thread_checker_); + MediaSourceInterface::SourceState state = source_->state(); + // TODO(tommi): Can we restart the media channel without blocking? + worker_thread_->BlockingCall([&] { + RTC_DCHECK_RUN_ON(worker_thread_); + RestartMediaChannel_w(std::move(ssrc), state); + }); + source_->SetState(MediaSourceInterface::kLive); +} + +void VideoRtpReceiver::RestartMediaChannel_w( + absl::optional<uint32_t> ssrc, + MediaSourceInterface::SourceState state) { + RTC_DCHECK_RUN_ON(worker_thread_); + if (!media_channel_) { + return; // Can't restart. + } + + const bool encoded_sink_enabled = saved_encoded_sink_enabled_; + + if (state != MediaSourceInterface::kInitializing) { + if (ssrc == signaled_ssrc_) + return; + + // Disconnect from a previous ssrc. + SetSink(nullptr); + + if (encoded_sink_enabled) + SetEncodedSinkEnabled(false); + } + + // Set up the new ssrc. + signaled_ssrc_ = std::move(ssrc); + SetSink(source_->sink()); + if (encoded_sink_enabled) { + SetEncodedSinkEnabled(true); + } + + if (frame_transformer_ && media_channel_) { + media_channel_->SetDepacketizerToDecoderFrameTransformer( + signaled_ssrc_.value_or(0), frame_transformer_); + } + + if (media_channel_ && signaled_ssrc_) { + if (frame_decryptor_) { + media_channel_->SetFrameDecryptor(*signaled_ssrc_, frame_decryptor_); + } + + media_channel_->SetBaseMinimumPlayoutDelayMs(*signaled_ssrc_, + delay_.GetMs()); + } +} + +void VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) { + RTC_DCHECK_RUN_ON(worker_thread_); + if (signaled_ssrc_) { + media_channel_->SetSink(*signaled_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); +} + +absl::optional<uint32_t> VideoRtpReceiver::ssrc() const { + RTC_DCHECK_RUN_ON(worker_thread_); + if (!signaled_ssrc_.has_value() && media_channel_) { + return media_channel_->GetUnsignaledSsrc(); + } + return signaled_ssrc_; +} + +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(video_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(video_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_ && signaled_ssrc_) + media_channel_->SetBaseMinimumPlayoutDelayMs(*signaled_ssrc_, + delay_.GetMs()); +} + +void VideoRtpReceiver::SetMediaChannel( + cricket::MediaReceiveChannelInterface* media_channel) { + RTC_DCHECK_RUN_ON(worker_thread_); + RTC_DCHECK(media_channel == nullptr || + media_channel->media_type() == media_type()); + + SetMediaChannel_w(media_channel); +} + +void VideoRtpReceiver::SetMediaChannel_w( + cricket::MediaReceiveChannelInterface* media_channel) { + RTC_DCHECK_RUN_ON(worker_thread_); + if (media_channel == media_channel_) + return; + + if (!media_channel) { + SetSink(nullptr); + } + + bool encoded_sink_enabled = saved_encoded_sink_enabled_; + if (encoded_sink_enabled && media_channel_) { + // Turn off the old sink, if any. + SetEncodedSinkEnabled(false); + } + + if (media_channel) { + media_channel_ = media_channel->AsVideoReceiveChannel(); + } else { + media_channel_ = nullptr; + } + + if (media_channel_) { + if (saved_generate_keyframe_) { + // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC + media_channel_->RequestRecvKeyFrame(signaled_ssrc_.value_or(0)); + saved_generate_keyframe_ = false; + } + if (encoded_sink_enabled) { + SetEncodedSinkEnabled(true); + } + if (frame_transformer_) { + media_channel_->SetDepacketizerToDecoderFrameTransformer( + signaled_ssrc_.value_or(0), frame_transformer_); + } + } + + if (!media_channel) + source_->ClearCallback(); +} + +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_); + auto current_ssrc = ssrc(); + if (!media_channel_ || !current_ssrc.has_value()) { + return {}; + } + return media_channel_->GetSources(current_ssrc.value()); +} + +void VideoRtpReceiver::SetupMediaChannel( + absl::optional<uint32_t> ssrc, + cricket::MediaReceiveChannelInterface* media_channel) { + RTC_DCHECK_RUN_ON(&signaling_thread_checker_); + RTC_DCHECK(media_channel); + MediaSourceInterface::SourceState state = source_->state(); + worker_thread_->BlockingCall([&] { + RTC_DCHECK_RUN_ON(worker_thread_); + SetMediaChannel_w(media_channel); + RestartMediaChannel_w(std::move(ssrc), state); + }); + source_->SetState(MediaSourceInterface::kLive); +} + +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_->RequestRecvKeyFrame(signaled_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) { + RTC_DCHECK_RUN_ON(worker_thread_); + if (!media_channel_) + return; + + // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC + const auto ssrc = signaled_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 |