webrtc_m130/webrtc/pc/rtpreceiver.cc
hbos 8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 14:39:05 +00:00

243 lines
7.3 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/pc/rtpreceiver.h"
#include "webrtc/api/mediastreamtrackproxy.h"
#include "webrtc/api/videosourceproxy.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/videotrack.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(const std::string& track_id,
uint32_t ssrc,
cricket::VoiceChannel* channel)
: id_(track_id),
ssrc_(ssrc),
channel_(channel),
track_(AudioTrackProxy::Create(
rtc::Thread::Current(),
AudioTrack::Create(track_id,
RemoteAudioSource::Create(ssrc, channel)))),
cached_track_enabled_(track_->enabled()) {
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
Reconfigure();
if (channel_) {
channel_->SignalFirstPacketReceived.connect(
this, &AudioRtpReceiver::OnFirstPacketReceived);
}
}
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) {
RTC_DCHECK(volume >= 0 && volume <= 10);
cached_volume_ = volume;
if (!channel_) {
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 (!channel_->SetOutputVolume(ssrc_, cached_volume_)) {
RTC_NOTREACHED();
}
}
}
RtpParameters AudioRtpReceiver::GetParameters() const {
if (!channel_ || stopped_) {
return RtpParameters();
}
return channel_->GetRtpReceiveParameters(ssrc_);
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
if (!channel_ || stopped_) {
return false;
}
return channel_->SetRtpReceiveParameters(ssrc_, parameters);
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
channel_->SetOutputVolume(ssrc_, 0);
}
stopped_ = true;
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
return channel_->GetSources(ssrc_);
}
void AudioRtpReceiver::Reconfigure() {
RTC_DCHECK(!stopped_);
if (!channel_) {
LOG(LS_ERROR) << "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!channel_->SetOutputVolume(ssrc_,
track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
}
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::SetChannel(cricket::VoiceChannel* channel) {
if (channel_) {
channel_->SignalFirstPacketReceived.disconnect(this);
}
channel_ = channel;
if (channel_) {
channel_->SignalFirstPacketReceived.connect(
this, &AudioRtpReceiver::OnFirstPacketReceived);
}
}
void AudioRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
VideoRtpReceiver::VideoRtpReceiver(const std::string& track_id,
rtc::Thread* worker_thread,
uint32_t ssrc,
cricket::VideoChannel* channel)
: id_(track_id),
ssrc_(ssrc),
channel_(channel),
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);
if (!channel_) {
LOG(LS_ERROR)
<< "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
} else {
if (!channel_->SetSink(ssrc_, &broadcaster_)) {
RTC_NOTREACHED();
}
}
if (channel_) {
channel_->SignalFirstPacketReceived.connect(
this, &VideoRtpReceiver::OnFirstPacketReceived);
}
}
VideoRtpReceiver::~VideoRtpReceiver() {
// Since cricket::VideoRenderer is not reference counted,
// we need to remove it from the channel before we are deleted.
Stop();
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!channel_ || stopped_) {
return RtpParameters();
}
return channel_->GetRtpReceiveParameters(ssrc_);
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
if (!channel_ || stopped_) {
return false;
}
return channel_->SetRtpReceiveParameters(ssrc_, parameters);
}
void VideoRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
source_->SetState(MediaSourceInterface::kEnded);
source_->OnSourceDestroyed();
if (!channel_) {
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.
channel_->SetSink(ssrc_, nullptr);
}
stopped_ = true;
}
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::SetChannel(cricket::VideoChannel* channel) {
if (channel_) {
channel_->SignalFirstPacketReceived.disconnect(this);
channel_->SetSink(ssrc_, nullptr);
}
channel_ = channel;
if (channel_) {
if (!channel_->SetSink(ssrc_, &broadcaster_)) {
RTC_NOTREACHED();
}
channel_->SignalFirstPacketReceived.connect(
this, &VideoRtpReceiver::OnFirstPacketReceived);
}
}
void VideoRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
} // namespace webrtc