Exposing RtpSenders and RtpReceivers from PeerConnection.
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.
It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.
Review URL: https://codereview.webrtc.org/1351803002
Cr-Commit-Position: refs/heads/master@{#10100}
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index aee77e1..fca98f2 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -25,4 +25,46 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-// This file is currently stubbed so that Chromium's build files can be updated.
+// This file contains interfaces for RtpSenders
+// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
+
+#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+
+#include <string>
+
+#include "talk/app/webrtc/proxy.h"
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class RtpSenderInterface : public rtc::RefCountInterface {
+ public:
+ // Returns true if successful in setting the track.
+ // Fails if an audio track is set on a video RtpSender, or vice-versa.
+ virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
+ virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+ // Not to be confused with "mid", this is a field we can temporarily use
+ // to uniquely identify a receiver until we implement Unified Plan SDP.
+ virtual std::string id() const = 0;
+
+ virtual void Stop() = 0;
+
+ protected:
+ virtual ~RtpSenderInterface() {}
+};
+
+// Define proxy for RtpSenderInterface.
+BEGIN_PROXY_MAP(RtpSender)
+PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(std::string, id)
+PROXY_METHOD0(void, Stop)
+END_PROXY()
+
+} // namespace webrtc
+
+#endif // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_