Separating internal and external methods of RtpSender/RtpReceiver.

webrtc/api/rtpsenderinterface.h

 /*
  *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 // This file contains interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
 #ifndef WEBRTC_API_RTPSENDERINTERFACE_H_
 #define WEBRTC_API_RTPSENDERINTERFACE_H_
 
 #include <string>
+#include <vector>
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/proxy.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/pc/mediasession.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;
 
-  // Used to set the SSRC of the sender, once a local description has been set.
-  // If |ssrc| is 0, this indiates that the sender should disconnect from the
-  // underlying transport (this occurs if the sender isn't seen in a local
-  // description).
-  virtual void SetSsrc(uint32_t ssrc) = 0;
+  // Returns primary SSRC used by this sender for sending media.
+  // Returns 0 if not yet determined.
+  // TODO(deadbeef): Change to rtc::Optional.
+  // TODO(deadbeef): Remove? With GetParameters this should be redundant.
   virtual uint32_t ssrc() const = 0;
 
   // Audio or video sender?
   virtual cricket::MediaType media_type() 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;
 
-  // TODO(deadbeef): Support one sender having multiple stream ids.
-  virtual void set_stream_id(const std::string& stream_id) = 0;
-  virtual std::string stream_id() const = 0;
-
-  virtual void Stop() = 0;
+  virtual std::vector<std::string> stream_ids() const = 0;
 
   virtual RtpParameters GetParameters() const = 0;
   virtual bool SetParameters(const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~RtpSenderInterface() {}
 };
 
 // Define proxy for RtpSenderInterface.
 BEGIN_SIGNALING_PROXY_MAP(RtpSender)
 PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_METHOD1(void, SetSsrc, uint32_t)
 PROXY_CONSTMETHOD0(uint32_t, ssrc)
 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD1(void, set_stream_id, const std::string&)
-PROXY_CONSTMETHOD0(std::string, stream_id)
-PROXY_METHOD0(void, Stop)
+PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
 PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
 PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
 END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPSENDERINTERFACE_H_