Adding the ability to create an RtpSender without a track.

talk/app/webrtc/rtpsenderinterface.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,
@@ skipping 29 matching lines @@
 
 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).

[pthatcher1, 2015/10/20 17:42:49]

Instead of setting SSRC to 0 here, could the caller instead call Stop() when
it's missing from the local description?

[Taylor Brandstetter, 2015/10/21 00:22:09]

Stop is permanent, SetSsrc(0) is not (as we discussed).

+  // Question for Peter: I know our code always generates nonzero SSRCs,
+  // but what is there to guarantee the peer does?

[pthatcher1, 2015/10/20 17:42:49]

Do you mean the remote peer?  It doesn't get to pick our send SSRCs at all.  It
can only pick its own send SSRCs, which or our receive SSRCs.

[Taylor Brandstetter, 2015/10/21 00:22:09]

I think you already answered this question, I just forgot to remove the comment.
Glad you caught it..

+  virtual void SetSsrc(uint32_t ssrc) = 0;

[pthatcher1, 2015/10/20 17:42:49]

I think we should call this set_ssrc.

[Taylor Brandstetter, 2015/10/21 00:22:09]

I thought a method should only be lowercase_underscore if its action is trivial
(like just setting or returning a variable). SetSsrc does more than that; it's
on the same level as SetTrack, almost.

[pthatcher1, 2015/10/22 07:34:14]

Yes, you're right.

+  virtual uint32_t ssrc() const = 0;
+
+  // "audio" or "video".
+  virtual std::string kind() const = 0;

[pthatcher1, 2015/10/20 17:42:49]

I think this should be a cricket::MediaType.

[Taylor Brandstetter, 2015/10/21 00:22:09]

Done.

+
   // 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;
 
  protected:
   virtual ~RtpSenderInterface() {}
 };
 
 // Define proxy for RtpSenderInterface.
 BEGIN_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(std::string, kind)
 PROXY_CONSTMETHOD0(std::string, id)
+PROXY_METHOD1(void, set_stream_id, const std::string&)
+PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
 END_PROXY()
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_