This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
239 lines
7.2 KiB
C++
239 lines
7.2 KiB
C++
/*
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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(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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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(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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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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