Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsender.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/app/webrtc/rtpsender.h"
 
-// This file is currently stubbed so that Chromium's build files can be updated.
+#include "talk/app/webrtc/localaudiosource.h"
+#include "talk/app/webrtc/videosourceinterface.h"
+
+namespace webrtc {
+
+LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
+
+LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
+  rtc::CritScope lock(&lock_);
+  if (sink_)
+    sink_->OnClose();
+}
+
+void LocalAudioSinkAdapter::OnData(const void* audio_data,
+                                   int bits_per_sample,
+                                   int sample_rate,
+                                   int number_of_channels,
+                                   size_t number_of_frames) {
+  rtc::CritScope lock(&lock_);
+  if (sink_) {
+    sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
+                  number_of_frames);
+  }
+}
+
+void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
+  rtc::CritScope lock(&lock_);
+  ASSERT(!sink || !sink_);
+  sink_ = sink;
+}
+
+AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
+                               uint32 ssrc,
+                               AudioProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()),
+      sink_adapter_(new LocalAudioSinkAdapter()) {
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+}
+
+AudioRtpSender::~AudioRtpSender() {
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void AudioRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "audio") {
+    LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = audio_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
+  return true;
+}
+
+void AudioRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  provider_->SetAudioSend(ssrc_, false, options, nullptr);
+  provider_ = nullptr;
+}
+
+void AudioRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  cricket::AudioOptions options;
+  if (track_->enabled() && track_->GetSource()) {
+    // TODO(xians): Remove this static_cast since we should be able to connect
+    // a remote audio track to peer connection.
+    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
+  }
+
+  // Use the renderer if the audio track has one, otherwise use the sink
+  // adapter owned by this class.
+  cricket::AudioRenderer* renderer =
+      track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
+  ASSERT(renderer != nullptr);
+  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
+}
+
+VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
+                               uint32 ssrc,
+                               VideoProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()) {
+  track_->RegisterObserver(this);
+  VideoSourceInterface* source = track_->GetSource();
+  if (source) {
+    provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
+  }
+  Reconfigure();
+}
+
+VideoRtpSender::~VideoRtpSender() {
+  track_->UnregisterObserver(this);
+  Stop();
+}
+
+void VideoRtpSender::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  if (track->kind() != "video") {
+    LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
+                  << " track.";
+    return false;
+  }
+  VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
+
+  // Detach from old track.
+  track_->UnregisterObserver(this);
+
+  // Attach to new track.
+  track_ = video_track;
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  Reconfigure();
+  return true;
+}
+
+void VideoRtpSender::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop sending packets.
+  if (!provider_) {
+    return;
+  }
+  provider_->SetCaptureDevice(ssrc_, nullptr);
+  provider_->SetVideoSend(ssrc_, false, nullptr);
+  provider_ = nullptr;
+}
+
+void VideoRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  const cricket::VideoOptions* options = nullptr;
+  VideoSourceInterface* source = track_->GetSource();
+  if (track_->enabled() && source) {
+    options = source->options();
+  }
+  provider_->SetVideoSend(ssrc_, track_->enabled(), options);
+}
+
+}  // namespace webrtc