Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiver.h

+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
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+ */
+
+// 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_