Files
platform-external-webrtc/pc/rtpreceiver.cc
Steve Anton 9158ef6575 Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80.

Original change's description:
> Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
> 
> This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.
> 
> Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
> 
> Original change's description:
> > Add AddTransceiver and GetTransceivers to PeerConnection
> > 
> > WebRTC 1.0 has added the transceiver API to PeerConnection. This
> > is the first step towards exposing this to WebRTC consumers. For
> > now, transceivers can be added and fetched but there is not yet
> > support for creating offers/answers or setting local/remote
> > descriptions. That support ("Unified Plan") will be added in
> > follow-up CLs.
> > 
> > The transceiver API is currently only available if the application
> > opts in by specifying the kUnifiedPlan SDP semantics when creating
> > the PeerConnection.
> > 
> > Bug: webrtc:7600
> > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> > Reviewed-on: https://webrtc-review.googlesource.com/23880
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20896}
> 
> TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
> 
> Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7600
> Reviewed-on: https://webrtc-review.googlesource.com/26400
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20897}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org

Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 22:27:49 +00:00

256 lines
7.5 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 "pc/rtpreceiver.h"
#include <utility>
#include <vector>
#include "api/mediastreamtrackproxy.h"
#include "api/videosourceproxy.h"
#include "pc/audiotrack.h"
#include "pc/videotrack.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(
const std::string& receiver_id,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
uint32_t ssrc,
cricket::VoiceChannel* channel)
: id_(receiver_id),
ssrc_(ssrc),
channel_(channel),
track_(AudioTrackProxy::Create(
rtc::Thread::Current(),
AudioTrack::Create(receiver_id,
RemoteAudioSource::Create(ssrc, channel)))),
streams_(std::move(streams)),
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_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!channel_) {
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 (!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_) {
RTC_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,
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams,
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_),
worker_thread))),
streams_(std::move(streams)) {
source_->SetState(MediaSourceInterface::kLive);
if (!channel_) {
RTC_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_) {
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.
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