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