The RtpReceiverObserverInterface is created. The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession. WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface. The application can listen to the Signal by implementing and registering a RtpReceiverObserver. Review-Url: https://codereview.webrtc.org/1999853002 Cr-Commit-Position: refs/heads/master@{#13139}
172 lines
5.5 KiB
C++
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
|