webrtc_m130/pc/video_rtp_receiver.cc
Henrik Boström a8ad11de82 [Rollback] Don't end tracks when transceiver is still in use.
Prior to this CL, calling RtpTransceiver::SetChannel() with null
arguments would cause the receiver's track to end. This is wrong,
because the channel can be nulled for other reasons than the transceiver
being stopped/removed - such as when the transceiver is rolled back but
still in use. Also, stopping a transceiver will end the track, so we
should simply ensure to always stop the transceiver when that is needed.

This CL makes sure that the transceiver is stopped or stopping in all
appropriate places, allowing us to remove the ability to end the source
for any other reason. A side-effect of this is that:
- The track never ends prematurely, fixing https://crbug.com/1315611.
- Removed transceivers are always stopped, fixing
  https://crbug.com/webrtc/14005.

This CL fixes the issue of track being ended in the ontrack event when
running https://jsfiddle.net/henbos/nxebusjm/.
- We don't have WPT test coverage for this, so I'll add that separately.

With SetSourceEnded() removed, some stopping/stop in response to
rejecting locally SDP munged content had to be added in order not to
regress the existing test coverage for this:
*PeerConnectionInterfaceTest.RejectMediaContent/1

Bug: chromium:1315611, webrtc:14005.
Change-Id: I21f30a1259e51324066dc84f72a72485b9e0fadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36669}
2022-04-27 11:57:52 +00:00

370 lines
11 KiB
C++

/*
* 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/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.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();
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_);
source_->SetState(MediaSourceInterface::kEnded);
track_->internal()->set_ended();
}
// RTC_RUN_ON(&signaling_thread_checker_)
void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
MediaSourceInterface::SourceState state = source_->state();
// TODO(tommi): Can we restart the media channel without blocking?
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
RestartMediaChannel_w(std::move(ssrc), state);
});
source_->SetState(MediaSourceInterface::kLive);
}
// RTC_RUN_ON(worker_thread_)
void VideoRtpReceiver::RestartMediaChannel_w(
absl::optional<uint32_t> ssrc,
MediaSourceInterface::SourceState state) {
if (!media_channel_) {
return; // Can't restart.
}
const bool encoded_sink_enabled = saved_encoded_sink_enabled_;
if (state != MediaSourceInterface::kInitializing) {
if (ssrc == ssrc_)
return;
// Disconnect from a previous ssrc.
SetSink(nullptr);
if (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());
}
}
// 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(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_ && ssrc_)
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
SetMediaChannel_w(media_channel);
}
// RTC_RUN_ON(worker_thread_)
void VideoRtpReceiver::SetMediaChannel_w(cricket::MediaChannel* media_channel) {
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);
}
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_);
}
}
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_);
if (!ssrc_ || !media_channel_)
return std::vector<RtpSource>();
return media_channel_->GetSources(*ssrc_);
}
void VideoRtpReceiver::SetupMediaChannel(absl::optional<uint32_t> ssrc,
cricket::MediaChannel* media_channel) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(media_channel);
MediaSourceInterface::SourceState state = source_->state();
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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_->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