Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
This commit is contained in:
ossu
242
webrtc/pc/rtpreceiver.cc
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242
webrtc/pc/rtpreceiver.cc
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/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/pc/rtpreceiver.h"
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#include "webrtc/api/mediastreamtrackproxy.h"
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#include "webrtc/api/videosourceproxy.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/pc/audiotrack.h"
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#include "webrtc/pc/videotrack.h"
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namespace webrtc {
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AudioRtpReceiver::AudioRtpReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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uint32_t ssrc,
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cricket::VoiceChannel* channel)
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: id_(track_id),
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ssrc_(ssrc),
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channel_(channel),
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track_(AudioTrackProxy::Create(
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rtc::Thread::Current(),
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AudioTrack::Create(track_id,
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RemoteAudioSource::Create(ssrc, channel)))),
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cached_track_enabled_(track_->enabled()) {
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RTC_DCHECK(track_->GetSource()->remote());
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track_->RegisterObserver(this);
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track_->GetSource()->RegisterAudioObserver(this);
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Reconfigure();
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stream->AddTrack(track_);
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if (channel_) {
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channel_->SignalFirstPacketReceived.connect(
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this, &AudioRtpReceiver::OnFirstPacketReceived);
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}
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}
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AudioRtpReceiver::~AudioRtpReceiver() {
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track_->GetSource()->UnregisterAudioObserver(this);
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track_->UnregisterObserver(this);
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Stop();
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}
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void AudioRtpReceiver::OnChanged() {
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if (cached_track_enabled_ != track_->enabled()) {
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cached_track_enabled_ = track_->enabled();
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Reconfigure();
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}
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}
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void AudioRtpReceiver::OnSetVolume(double volume) {
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RTC_DCHECK(volume >= 0 && volume <= 10);
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cached_volume_ = volume;
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if (!channel_) {
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LOG(LS_ERROR) << "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
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return;
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}
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// When the track is disabled, the volume of the source, which is the
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// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
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// setting the volume to the source when the track is disabled.
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if (!stopped_ && track_->enabled()) {
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if (!channel_->SetOutputVolume(ssrc_, cached_volume_)) {
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RTC_NOTREACHED();
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}
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}
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}
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RtpParameters AudioRtpReceiver::GetParameters() const {
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if (!channel_ || stopped_) {
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return RtpParameters();
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}
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return channel_->GetRtpReceiveParameters(ssrc_);
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}
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bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
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TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
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if (!channel_ || stopped_) {
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return false;
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}
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return channel_->SetRtpReceiveParameters(ssrc_, parameters);
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}
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void AudioRtpReceiver::Stop() {
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// TODO(deadbeef): Need to do more here to fully stop receiving packets.
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if (stopped_) {
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return;
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}
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if (channel_) {
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// Allow that SetOutputVolume fail. This is the normal case when the
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// underlying media channel has already been deleted.
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channel_->SetOutputVolume(ssrc_, 0);
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}
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stopped_ = true;
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}
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void AudioRtpReceiver::Reconfigure() {
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RTC_DCHECK(!stopped_);
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if (!channel_) {
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LOG(LS_ERROR) << "AudioRtpReceiver::Reconfigure: No audio channel exists.";
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return;
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}
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if (!channel_->SetOutputVolume(ssrc_,
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track_->enabled() ? cached_volume_ : 0)) {
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RTC_NOTREACHED();
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}
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}
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void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
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observer_ = observer;
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// Deliver any notifications the observer may have missed by being set late.
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if (received_first_packet_ && observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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}
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void AudioRtpReceiver::SetChannel(cricket::VoiceChannel* channel) {
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if (channel_) {
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channel_->SignalFirstPacketReceived.disconnect(this);
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}
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channel_ = channel;
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if (channel_) {
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channel_->SignalFirstPacketReceived.connect(
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this, &AudioRtpReceiver::OnFirstPacketReceived);
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}
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}
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void AudioRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
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if (observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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received_first_packet_ = true;
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}
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VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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rtc::Thread* worker_thread,
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uint32_t ssrc,
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cricket::VideoChannel* channel)
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: id_(track_id),
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ssrc_(ssrc),
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channel_(channel),
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source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
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true /* remote */)),
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track_(VideoTrackProxy::Create(
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rtc::Thread::Current(),
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worker_thread,
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VideoTrack::Create(
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track_id,
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VideoTrackSourceProxy::Create(rtc::Thread::Current(),
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worker_thread,
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source_)))) {
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source_->SetState(MediaSourceInterface::kLive);
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if (!channel_) {
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LOG(LS_ERROR)
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<< "VideoRtpReceiver::VideoRtpReceiver: No video channel exists.";
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} else {
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if (!channel_->SetSink(ssrc_, &broadcaster_)) {
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RTC_NOTREACHED();
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}
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}
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stream->AddTrack(track_);
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if (channel_) {
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channel_->SignalFirstPacketReceived.connect(
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this, &VideoRtpReceiver::OnFirstPacketReceived);
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}
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}
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VideoRtpReceiver::~VideoRtpReceiver() {
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// Since cricket::VideoRenderer is not reference counted,
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// we need to remove it from the channel before we are deleted.
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Stop();
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}
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RtpParameters VideoRtpReceiver::GetParameters() const {
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if (!channel_ || stopped_) {
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return RtpParameters();
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}
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return channel_->GetRtpReceiveParameters(ssrc_);
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}
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bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
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TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
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if (!channel_ || stopped_) {
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return false;
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}
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return channel_->SetRtpReceiveParameters(ssrc_, parameters);
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}
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void VideoRtpReceiver::Stop() {
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// TODO(deadbeef): Need to do more here to fully stop receiving packets.
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if (stopped_) {
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return;
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}
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source_->SetState(MediaSourceInterface::kEnded);
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source_->OnSourceDestroyed();
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if (!channel_) {
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LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
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} else {
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// Allow that SetSink fail. This is the normal case when the underlying
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// media channel has already been deleted.
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channel_->SetSink(ssrc_, nullptr);
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}
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stopped_ = true;
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}
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void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
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observer_ = observer;
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// Deliver any notifications the observer may have missed by being set late.
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if (received_first_packet_ && observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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}
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void VideoRtpReceiver::SetChannel(cricket::VideoChannel* channel) {
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if (channel_) {
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channel_->SignalFirstPacketReceived.disconnect(this);
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channel_->SetSink(ssrc_, nullptr);
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}
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channel_ = channel;
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if (channel_) {
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if (!channel_->SetSink(ssrc_, &broadcaster_)) {
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RTC_NOTREACHED();
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}
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channel_->SignalFirstPacketReceived.connect(
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this, &VideoRtpReceiver::OnFirstPacketReceived);
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}
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}
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void VideoRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
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if (observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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received_first_packet_ = true;
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}
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} // namespace webrtc
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