blob: e6de6c71b3d10aeeb7576036e1f888e04f974b8a [file] [log] [blame]
/*
* 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/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/video_track_source_proxy.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/jitter_buffer_delay_proxy.h"
#include "pc/media_stream.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.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>()),
track_(VideoTrackProxy::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();
}
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;
}
bool VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
RTC_DCHECK(media_channel_);
RTC_DCHECK(!stopped_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
return media_channel_->SetSink(ssrc_.value_or(0), sink);
});
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!media_channel_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
return media_channel_->GetRtpReceiveParameters(ssrc_.value_or(0));
});
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
if (!media_channel_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
return media_channel_->SetRtpReceiveParameters(ssrc_.value_or(0),
parameters);
});
}
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::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 fail. This is the normal case when the underlying
// media channel has already been deleted.
SetSink(nullptr);
}
delay_->OnStop();
stopped_ = true;
}
void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
RTC_DCHECK(media_channel_);
if (!stopped_ && ssrc_ == ssrc) {
return;
}
if (!stopped_) {
SetSink(nullptr);
}
stopped_ = false;
ssrc_ = ssrc;
SetSink(source_->sink());
// 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::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());
media_channel_ = static_cast<cricket::VideoMediaChannel*>(media_channel);
}
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_); });
}
} // namespace webrtc