Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiver.cc

Index: talk/app/webrtc/rtpreceiver.cc
diff --git a/talk/app/webrtc/rtpreceiver.cc b/talk/app/webrtc/rtpreceiver.cc
new file mode 100644
index 0000000000000000000000000000000000000000..3be297ff3acd1ec06eb90a03da466c769233a269
--- /dev/null
+++ b/talk/app/webrtc/rtpreceiver.cc
@@ -0,0 +1,104 @@
+/*
+ * 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/rtpreceiver.h"
+
+#include "talk/app/webrtc/videosourceinterface.h"
+
+namespace webrtc {
+
+AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
+                                   uint32 ssrc,
+                                   AudioProviderInterface* provider)
+    : id_(track->id()),
+      track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
+      cached_track_enabled_(track->enabled()) {
+  track_->RegisterObserver(this);
+  track_->GetSource()->RegisterAudioObserver(this);
+  Reconfigure();
+}
+
+AudioRtpReceiver::~AudioRtpReceiver() {
+  track_->GetSource()->UnregisterAudioObserver(this);
+  track_->UnregisterObserver(this);
+  if (provider_) {
+    Stop();
+  }
+}
+
+void AudioRtpReceiver::OnChanged() {
+  if (cached_track_enabled_ != track_->enabled()) {
+    cached_track_enabled_ = track_->enabled();
+    Reconfigure();
+  }
+}
+
+void AudioRtpReceiver::OnSetVolume(double volume) {
+  // 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 (provider_ && track_->enabled())
+    provider_->SetAudioPlayoutVolume(ssrc_, volume);
+}
+
+void AudioRtpReceiver::Stop() {
+  // TODO(deadbeef): Need to do more here to fully stop receiving packets.
+  provider_->SetAudioPlayout(ssrc_, false, nullptr);
+  provider_ = nullptr;
+}
+
+void AudioRtpReceiver::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
+  provider_->SetAudioPlayout(ssrc_, track_->enabled(), track_->GetRenderer());
+}
+
+VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track,
+                                   uint32 ssrc,
+                                   VideoProviderInterface* provider)
+    : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
+  provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
+}
+
+VideoRtpReceiver::~VideoRtpReceiver() {
+  if (provider_) {
+    // Since cricket::VideoRenderer is not reference counted,
+    // we need to remove it from the provider before we are deleted.
+    Stop();
+  }
+}
+
+void VideoRtpReceiver::Stop() {

[pthatcher1, 2015/09/24 21:35:22]

We should probably put 

if (provider_) {
}

In the Stop method rather than in the caller of the Stop method (above in the
desctructor).

[Taylor Brandstetter, 2015/09/24 22:00:53]

Done.

+  // TODO(deadbeef): Need to do more here to fully stop receiving packets.
+  provider_->SetVideoPlayout(ssrc_, false, nullptr);
+  provider_ = nullptr;
+}
+
+}  // namespace webrtc