Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiver.h

+/*
+ * 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 AudioRtpReceiver : public ObserverInterface,
+                         public AudioSourceInterface::AudioObserver,
+                         public RtpReceiverInterface {
+ public:
+  AudioRtpReceiver(AudioTrackInterface* track,
+                   uint32 ssrc,
+                   const std::string& mid,
+                   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();
+  }
+  std::string mid() const override { return mid_; }

[pthatcher1, 2015/09/17 04:25:46]

I think we should either remove this or just call it "id" so that it can work
well between Plan B (MSID), Unified Plan (MID), and ORTC (an ID unto itself).

[Taylor Brandstetter, 2015/09/23 00:10:45]

Removed.

+
+  // Detach from AudioProviderInterface
+  void Stop() override;

[pthatcher1, 2015/09/17 04:25:46]

Does this also shutdown the media, or does setLocal/setRemoteDescription do
that?  We should probably make it clear in the comment one way or the other.

[Taylor Brandstetter, 2015/09/23 00:10:45]

I'm not completely sure what "shutdown the media" entails, but currently this
just detaches the track from the AudioProviderInterface (the media channel).

+
+ private:
+  void UpdateEnabled();
+
+  rtc::scoped_refptr<AudioTrackInterface> track_;
+  uint32 ssrc_;

[pthatcher1, 2015/09/17 04:25:46]

Can you put a TODO to change the key we use for passing into
BaseChannel/MediaChannel to a "sender_id" string?

[Taylor Brandstetter, 2015/09/23 00:10:45]

Done.

+  std::string mid_;
+  AudioProviderInterface* provider_;

[pthatcher1, 2015/09/17 04:25:46]

Can you comment that this is just a crazy way to get to the BaseChannel, and
leave a TODO for refactoring that?

[Taylor Brandstetter, 2015/09/23 00:10:45]

Done.

+  bool enabled_;

[pthatcher1, 2015/09/17 04:25:46]

Can you rename this to track_enabled_?  That would make it much more clear,
especially if you add a comment like "stored separately from track_ because we
need to know whether it changes or not."

[Taylor Brandstetter, 2015/09/23 00:10:45]

Changed to cached_track_enabled_.

+};
+
+class VideoRtpReceiver : public RtpReceiverInterface {

[pthatcher1, 2015/09/17 04:25:46]

All the comment I left for audio apply to video (except the track_enabled_).

+ public:
+  VideoRtpReceiver(VideoTrackInterface* track,
+                   uint32 ssrc,
+                   const std::string& mid,
+                   VideoProviderInterface* provider);
+
+  virtual ~VideoRtpReceiver();
+
+  // RtpReceiverInterface implementation
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+  std::string mid() const override { return mid_; }
+
+  // Detach from VideoProviderInterface
+  void Stop() override;
+
+ private:
+  void UpdateEnabled();
+
+  rtc::scoped_refptr<VideoTrackInterface> track_;
+  uint32 ssrc_;
+  std::string mid_;
+  VideoProviderInterface* provider_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_RTPRECEIVER_H_