rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/api/rtpreceiver.cc

172 lines
5.5 KiB
C++

/*
* Copyright 2015 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 "webrtc/api/rtpreceiver.h"
#include "webrtc/api/mediastreamtrackproxy.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/videosourceproxy.h"
#include "webrtc/api/videotrack.h"
#include "webrtc/base/trace_event.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc,
AudioProviderInterface* provider)
: id_(track_id),
ssrc_(ssrc),
provider_(provider),
track_(AudioTrackProxy::Create(
rtc::Thread::Current(),
AudioTrack::Create(track_id,
RemoteAudioSource::Create(ssrc, provider)))),
cached_track_enabled_(track_->enabled()) {
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
Reconfigure();
stream->AddTrack(track_);
provider_->SignalFirstAudioPacketReceived.connect(
this, &AudioRtpReceiver::OnFirstAudioPacketReceived);
}
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();
}
}
void AudioRtpReceiver::OnSetVolume(double volume) {
// 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 (provider_ && track_->enabled())
provider_->SetAudioPlayoutVolume(ssrc_, volume);
}
RtpParameters AudioRtpReceiver::GetParameters() const {
return provider_->GetAudioRtpReceiveParameters(ssrc_);
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
return provider_->SetAudioRtpReceiveParameters(ssrc_, parameters);
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (!provider_) {
return;
}
provider_->SetAudioPlayout(ssrc_, false);
provider_ = nullptr;
}
void AudioRtpReceiver::Reconfigure() {
if (!provider_) {
return;
}
provider_->SetAudioPlayout(ssrc_, track_->enabled());
}
void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
observer_ = observer;
// If received the first packet before setting the observer, call the
// observer.
if (received_first_packet_) {
observer_->OnFirstPacketReceived(media_type());
}
}
void AudioRtpReceiver::OnFirstAudioPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
const std::string& track_id,
rtc::Thread* worker_thread,
uint32_t ssrc,
VideoProviderInterface* provider)
: id_(track_id),
ssrc_(ssrc),
provider_(provider),
source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
true /* remote */)),
track_(VideoTrackProxy::Create(
rtc::Thread::Current(),
worker_thread,
VideoTrack::Create(
track_id,
VideoTrackSourceProxy::Create(rtc::Thread::Current(),
worker_thread,
source_)))) {
source_->SetState(MediaSourceInterface::kLive);
provider_->SetVideoPlayout(ssrc_, true, &broadcaster_);
stream->AddTrack(track_);
provider_->SignalFirstVideoPacketReceived.connect(
this, &VideoRtpReceiver::OnFirstVideoPacketReceived);
}
VideoRtpReceiver::~VideoRtpReceiver() {
// Since cricket::VideoRenderer is not reference counted,
// we need to remove it from the provider before we are deleted.
Stop();
}
RtpParameters VideoRtpReceiver::GetParameters() const {
return provider_->GetVideoRtpReceiveParameters(ssrc_);
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
return provider_->SetVideoRtpReceiveParameters(ssrc_, parameters);
}
void VideoRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (!provider_) {
return;
}
source_->SetState(MediaSourceInterface::kEnded);
source_->OnSourceDestroyed();
provider_->SetVideoPlayout(ssrc_, false, nullptr);
provider_ = nullptr;
}
void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
observer_ = observer;
// If received the first packet before setting the observer, call the
// observer.
if (received_first_packet_) {
observer_->OnFirstPacketReceived(media_type());
}
}
void VideoRtpReceiver::OnFirstVideoPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
} // namespace webrtc