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..3894fa5b45b15d0c15f2eae8cf6dfff6c386cd6b
--- /dev/null
+++ b/talk/app/webrtc/rtpreceiver.h
@@ -0,0 +1,112 @@
+/*
+ * 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:
+  explicit BaseRtpReceiver(const std::string& id) : id_(id) {}
+  virtual void DetachFromProvider() = 0;
+
+  std::string id() const override { return id_; }

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

If you move DetachFromProvider (hopefully named Stop) into RtpReceiverInterface,
could you make PeerConnection have a vector of RtpReceiverIntefaces instead of
BaseRtpReceivers?

[Taylor Brandstetter, 2015/09/24 20:54:20]

That's the way it was before, but I purposefully made this change because
DetachFromProvider really isn't a "Stop" as in the ORTC spec, and it should only
be called internally by PeerConnection. It means more boilerplate code, but I
think it better separates the interface from implementation.

If DetachFromProvider is really SUPPOSED to be "Stop" (and tear down everything
down to the MediaChannel), I'd be fine with putting things back and leaving a
"TODO" to implement it properly. But with the "RtpTransceiver" PR, are
individual senders/receivers really going to have a Stop method?

+
+ protected:
+  virtual ~BaseRtpReceiver() {}
+
+ private:
+  std::string id_;
+};
+
+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_;
+};
+
+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_