Split rtp_receiver for readability.

Bug: webrtc:10304
Change-Id: I85e421060d8560cf36cdb5970ae190efc77e7709
Reviewed-on: https://webrtc-review.googlesource.com/c/122085
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26634}
This commit is contained in:
Ruslan Burakov 2019-02-11 10:29:19 +01:00 committed by Commit Bot
parent b66003ca79
commit 501bfba0cb
10 changed files with 755 additions and 611 deletions

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@ -128,6 +128,8 @@ rtc_static_library("peerconnection") {
visibility = [ "*" ]
cflags = []
sources = [
"audio_rtp_receiver.cc",
"audio_rtp_receiver.h",
"audio_track.cc",
"audio_track.h",
"data_channel.cc",
@ -175,6 +177,8 @@ rtc_static_library("peerconnection") {
"stream_collection.h",
"track_media_info_map.cc",
"track_media_info_map.h",
"video_rtp_receiver.cc",
"video_rtp_receiver.h",
"video_track.cc",
"video_track.h",
"video_track_source.cc",

248
pc/audio_rtp_receiver.cc Normal file
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@ -0,0 +1,248 @@
/*
* 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/audio_rtp_receiver.h"
#include <stddef.h>
#include <utility>
#include <vector>
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "pc/audio_track.h"
#include "pc/media_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
: AudioRtpReceiver(worker_thread,
receiver_id,
CreateStreamsFromIds(std::move(stream_ids))) {}
AudioRtpReceiver::AudioRtpReceiver(
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 rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
track_(AudioTrackProxy::Create(rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
attachment_id_(GenerateUniqueId()) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
SetStreams(streams);
}
AudioRtpReceiver::~AudioRtpReceiver() {
track_->GetSource()->UnregisterAudioObserver(this);
track_->UnregisterObserver(this);
Stop();
}
void AudioRtpReceiver::OnChanged() {
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
}
}
bool AudioRtpReceiver::SetOutputVolume(double volume) {
RTC_DCHECK_GE(volume, 0.0);
RTC_DCHECK_LE(volume, 10.0);
RTC_DCHECK(media_channel_);
RTC_DCHECK(ssrc_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetOutputVolume(*ssrc_, volume);
});
}
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
return;
}
// When the track is disabled, the volume of the source, which is the
// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
// setting the volume to the source when the track is disabled.
if (!stopped_ && track_->enabled()) {
if (!SetOutputVolume(cached_volume_)) {
RTC_NOTREACHED();
}
}
}
std::vector<std::string> AudioRtpReceiver::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;
}
RtpParameters AudioRtpReceiver::GetParameters() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
});
}
void AudioRtpReceiver::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>
AudioRtpReceiver::GetFrameDecryptor() const {
return frame_decryptor_;
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (media_channel_ && ssrc_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
SetOutputVolume(0.0);
}
stopped_ = true;
}
void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
return;
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
source_->Stop(media_channel_, *ssrc_);
}
ssrc_ = ssrc;
source_->Start(media_channel_, *ssrc_);
Reconfigure();
}
void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void AudioRtpReceiver::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;
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return {};
}
return worker_thread_->Invoke<std::vector<RtpSource>>(
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
}
void AudioRtpReceiver::Reconfigure() {
RTC_DCHECK(!stopped_);
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
// Reattach the frame decryptor if we were reconfigured.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
}
void AudioRtpReceiver::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 AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
}
void AudioRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
} // namespace webrtc

126
pc/audio_rtp_receiver.h Normal file
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@ -0,0 +1,126 @@
/*
* 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.
*/
#ifndef PC_AUDIO_RTP_RECEIVER_H_
#define PC_AUDIO_RTP_RECEIVER_H_
#include <stdint.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "media/base/media_channel.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
namespace webrtc {
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public rtc::RefCountedObject<RtpReceiverInternal> {
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
void OnChanged() override;
// AudioSourceInterface::AudioObserver implementation
void OnSetVolume(double volume) override;
rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
}
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetMediaChannel(cricket::MediaChannel* media_channel) override;
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
private:
void Reconfigure();
bool SetOutputVolume(double volume);
rtc::Thread* const worker_thread_;
const std::string id_;
const rtc::scoped_refptr<RemoteAudioSource> source_;
const rtc::scoped_refptr<AudioTrackInterface> track_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool cached_track_enabled_;
double cached_volume_ = 1;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
} // namespace webrtc
#endif // PC_AUDIO_RTP_RECEIVER_H_

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@ -30,6 +30,7 @@
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctp_transport.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
#include "pc/channel.h"
#include "pc/channel_manager.h"
@ -43,6 +44,7 @@
#include "pc/sctp_utils.h"
#include "pc/sdp_utils.h"
#include "pc/stream_collection.h"
#include "pc/video_rtp_receiver.h"
#include "pc/video_track.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"

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@ -16,10 +16,7 @@
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/video_track_source_proxy.h"
#include "pc/audio_track.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"
@ -27,17 +24,15 @@
namespace webrtc {
namespace {
// This function is only expected to be called on the signalling thread.
int GenerateUniqueId() {
int RtpReceiverInternal::GenerateUniqueId() {
static int g_unique_id = 0;
return ++g_unique_id;
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds(
std::vector<std::string> stream_ids) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>>
RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
stream_ids.size());
for (size_t i = 0; i < stream_ids.size(); ++i) {
@ -50,7 +45,7 @@ std::vector<rtc::scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds(
// Attempt to attach the frame decryptor to the current media channel on the
// correct worker thread only if both the media channel exists and a ssrc has
// been allocated to the stream.
void MaybeAttachFrameDecryptorToMediaChannel(
void RtpReceiverInternal::MaybeAttachFrameDecryptorToMediaChannel(
const absl::optional<uint32_t>& ssrc,
rtc::Thread* worker_thread,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
@ -63,410 +58,4 @@ void MaybeAttachFrameDecryptorToMediaChannel(
}
}
} // namespace
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
: AudioRtpReceiver(worker_thread,
receiver_id,
CreateStreamsFromIds(std::move(stream_ids))) {}
AudioRtpReceiver::AudioRtpReceiver(
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 rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
track_(AudioTrackProxy::Create(rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
attachment_id_(GenerateUniqueId()) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
SetStreams(streams);
}
AudioRtpReceiver::~AudioRtpReceiver() {
track_->GetSource()->UnregisterAudioObserver(this);
track_->UnregisterObserver(this);
Stop();
}
void AudioRtpReceiver::OnChanged() {
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
}
}
bool AudioRtpReceiver::SetOutputVolume(double volume) {
RTC_DCHECK_GE(volume, 0.0);
RTC_DCHECK_LE(volume, 10.0);
RTC_DCHECK(media_channel_);
RTC_DCHECK(ssrc_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetOutputVolume(*ssrc_, volume);
});
}
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
return;
}
// When the track is disabled, the volume of the source, which is the
// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
// setting the volume to the source when the track is disabled.
if (!stopped_ && track_->enabled()) {
if (!SetOutputVolume(cached_volume_)) {
RTC_NOTREACHED();
}
}
}
std::vector<std::string> AudioRtpReceiver::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;
}
RtpParameters AudioRtpReceiver::GetParameters() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
});
}
void AudioRtpReceiver::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>
AudioRtpReceiver::GetFrameDecryptor() const {
return frame_decryptor_;
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (media_channel_ && ssrc_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
SetOutputVolume(0.0);
}
stopped_ = true;
}
void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
return;
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
source_->Stop(media_channel_, *ssrc_);
}
ssrc_ = ssrc;
source_->Start(media_channel_, *ssrc_);
Reconfigure();
}
void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void AudioRtpReceiver::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;
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return {};
}
return worker_thread_->Invoke<std::vector<RtpSource>>(
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
}
void AudioRtpReceiver::Reconfigure() {
RTC_DCHECK(!stopped_);
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
// Reattach the frame decryptor if we were reconfigured.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
}
void AudioRtpReceiver::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 AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
}
void AudioRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
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()) {
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(ssrc_);
return worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&] { return media_channel_->SetSink(*ssrc_, sink); });
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, 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_ || !ssrc_) {
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);
}
stopped_ = true;
}
void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "VideoRtpReceiver::SetupMediaChannel: No video channel exists.";
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
SetSink(nullptr);
}
ssrc_ = ssrc;
SetSink(source_->sink());
// Attach any existing frame decryptor to the media channel.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
}
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::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

View File

@ -31,7 +31,6 @@
#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
#include "media/base/video_broadcaster.h"
#include "pc/remote_audio_source.h"
#include "pc/video_track_source.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
@ -78,203 +77,19 @@ class RtpReceiverInternal : public RtpReceiverInterface {
// otherwise remains constant. Used to generate IDs for stats.
// The special value zero means that no track is attached.
virtual int AttachmentId() const = 0;
};
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public rtc::RefCountedObject<RtpReceiverInternal> {
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
protected:
static int GenerateUniqueId();
static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
CreateStreamsFromIds(std::vector<std::string> stream_ids);
static void MaybeAttachFrameDecryptorToMediaChannel(
const absl::optional<uint32_t>& ssrc,
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
void OnChanged() override;
// AudioSourceInterface::AudioObserver implementation
void OnSetVolume(double volume) override;
rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
}
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetMediaChannel(cricket::MediaChannel* media_channel) override;
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
private:
void Reconfigure();
bool SetOutputVolume(double volume);
rtc::Thread* const worker_thread_;
const std::string id_;
const rtc::scoped_refptr<RemoteAudioSource> source_;
const rtc::scoped_refptr<AudioTrackInterface> track_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool cached_track_enabled_;
double cached_volume_ = 1;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> streams_ids);
// TODO(hbos): Remove this when streams() is removed.
// https://crbug.com/webrtc/9480
VideoRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
virtual ~VideoRtpReceiver();
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
}
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetMediaChannel(cricket::MediaChannel* media_channel) override;
int AttachmentId() const override { return attachment_id_; }
std::vector<RtpSource> GetSources() const override;
private:
class VideoRtpTrackSource : public VideoTrackSource {
public:
VideoRtpTrackSource() : VideoTrackSource(true /* remote */) {}
rtc::VideoSourceInterface<VideoFrame>* source() override {
return &broadcaster_;
}
rtc::VideoSinkInterface<VideoFrame>* sink() { return &broadcaster_; }
private:
// |broadcaster_| is needed since the decoder can only handle one sink.
// It might be better if the decoder can handle multiple sinks and consider
// the VideoSinkWants.
rtc::VideoBroadcaster broadcaster_;
};
bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
rtc::Thread* const worker_thread_;
const std::string id_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
// |source_| is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
rtc::scoped_refptr<VideoRtpTrackSource> source_;
rtc::scoped_refptr<VideoTrackInterface> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
cricket::MediaChannel* media_channel,
bool stopped);
};
} // namespace webrtc

View File

@ -41,6 +41,7 @@
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/p2p_constants.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
#include "pc/channel.h"
#include "pc/channel_manager.h"
@ -51,6 +52,7 @@
#include "pc/rtp_sender.h"
#include "pc/rtp_transport_internal.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/video_rtp_receiver.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"

View File

@ -15,6 +15,7 @@
#include <vector>
#include "api/rtp_transceiver_interface.h"
#include "pc/channel_interface.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"

213
pc/video_rtp_receiver.cc Normal file
View File

@ -0,0 +1,213 @@
/*
* 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/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()) {
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(ssrc_);
return worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&] { return media_channel_->SetSink(*ssrc_, sink); });
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, 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_ || !ssrc_) {
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);
}
stopped_ = true;
}
void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "VideoRtpReceiver::SetupMediaChannel: No video channel exists.";
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
SetSink(nullptr);
}
ssrc_ = ssrc;
SetSink(source_->sink());
// Attach any existing frame decryptor to the media channel.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
}
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::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

144
pc/video_rtp_receiver.h Normal file
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@ -0,0 +1,144 @@
/*
* 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.
*/
#ifndef PC_VIDEO_RTP_RECEIVER_H_
#define PC_VIDEO_RTP_RECEIVER_H_
#include <stdint.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
#include "media/base/video_broadcaster.h"
#include "pc/rtp_receiver.h"
#include "pc/video_track_source.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
namespace webrtc {
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> streams_ids);
// TODO(hbos): Remove this when streams() is removed.
// https://crbug.com/webrtc/9480
VideoRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
virtual ~VideoRtpReceiver();
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
}
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetMediaChannel(cricket::MediaChannel* media_channel) override;
int AttachmentId() const override { return attachment_id_; }
std::vector<RtpSource> GetSources() const override;
private:
class VideoRtpTrackSource : public VideoTrackSource {
public:
VideoRtpTrackSource() : VideoTrackSource(true /* remote */) {}
rtc::VideoSourceInterface<VideoFrame>* source() override {
return &broadcaster_;
}
rtc::VideoSinkInterface<VideoFrame>* sink() { return &broadcaster_; }
private:
// |broadcaster_| is needed since the decoder can only handle one sink.
// It might be better if the decoder can handle multiple sinks and consider
// the VideoSinkWants.
rtc::VideoBroadcaster broadcaster_;
};
bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
rtc::Thread* const worker_thread_;
const std::string id_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
// |source_| is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
rtc::scoped_refptr<VideoRtpTrackSource> source_;
rtc::scoped_refptr<VideoTrackInterface> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
};
} // namespace webrtc
#endif // PC_VIDEO_RTP_RECEIVER_H_