Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiver.h

Index: talk/app/webrtc/rtpreceiver.h
diff --git a/talk/app/webrtc/rtpreceiver.h b/talk/app/webrtc/rtpreceiver.h
new file mode 100644
index 0000000000000000000000000000000000000000..d4bced6fc04a7bd27c8a84ab9870c545d4d7bfcc
--- /dev/null
+++ b/talk/app/webrtc/rtpreceiver.h
@@ -0,0 +1,106 @@
+/*
+ * 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.
+ */
+
+// This file contains classes that implement RtpReceiverInterface.
+// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
+// transport (provided by AudioProviderInterface/VideoProviderInterface)
+
+#ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_
+#define TALK_APP_WEBRTC_RTPRECEIVER_H_
+
+#include <string>
+
+#include "talk/app/webrtc/mediastreamprovider.h"
+#include "talk/app/webrtc/rtpreceiverinterface.h"
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+
+class BaseRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
+ public:
+  virtual void DetachFromProvider() = 0;

[pthatcher1, 2015/09/23 14:47:40]

I think it would make sense to call this method Stop(), because it's exactly
like the RtpReceiver::Stop method in ORTC.  In fact, I think we should put the
method on RtpReceiverInterface.  Then you might not need the base class.

[pthatcher1, 2015/09/24 06:32:29]

I still think we should call this method Stop.

+
+ protected:
+  virtual ~BaseRtpReceiver() {}
+};
+
+class AudioRtpReceiver : public ObserverInterface,
+                         public AudioSourceInterface::AudioObserver,
+                         public BaseRtpReceiver {
+ public:
+  AudioRtpReceiver(AudioTrackInterface* track,
+                   uint32 ssrc,
+                   AudioProviderInterface* provider);
+
+  virtual ~AudioRtpReceiver();
+
+  // ObserverInterface implementation
+  void OnChanged() override;
+
+  // AudioSourceInterface::AudioObserver implementation
+  void OnSetVolume(double volume) override;
+
+  // RtpReceiverInterface implementation
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+
+  void DetachFromProvider() override;
+
+ private:
+  void UpdateProvider();
+
+  rtc::scoped_refptr<AudioTrackInterface> track_;
+  uint32 ssrc_;
+  AudioProviderInterface* provider_;
+  bool cached_track_enabled_;

[pthatcher1, 2015/09/23 14:47:40]

Why not just track_enabled_, with a comment explaining that we keep it separate
to know when track_->enabled() changed.

+};
+
+class VideoRtpReceiver : public BaseRtpReceiver {
+ public:
+  VideoRtpReceiver(VideoTrackInterface* track,
+                   uint32 ssrc,
+                   VideoProviderInterface* provider);
+
+  virtual ~VideoRtpReceiver();
+
+  // RtpReceiverInterface implementation
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+
+  void DetachFromProvider() override;
+
+ private:
+  rtc::scoped_refptr<VideoTrackInterface> track_;
+  uint32 ssrc_;
+  VideoProviderInterface* provider_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_RTPRECEIVER_H_