Adding the ability to create an RtpSender without a track.

talk/app/webrtc/rtpsender.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 18 matching lines @@
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
 #ifndef TALK_APP_WEBRTC_RTPSENDER_H_
 #define TALK_APP_WEBRTC_RTPSENDER_H_
 
 #include <string>
 
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
+#include "talk/app/webrtc/statscollector.h"
 #include "talk/media/base/audiorenderer.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 // LocalAudioSinkAdapter receives data callback as a sink to the local
 // AudioTrack, and passes the data to the sink of AudioRenderer.
 class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
@@ skipping 14 matching lines @@
   void SetSink(cricket::AudioRenderer::Sink* sink) override;
 
   cricket::AudioRenderer::Sink* sink_;
   // Critical section protecting |sink_|.
   rtc::CriticalSection lock_;
 };
 
 class AudioRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
+  // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
+  // at the appropriate times.
   AudioRtpSender(AudioTrackInterface* track,
-                 uint32_t ssrc,
-                 AudioProviderInterface* provider);
+                 const std::string& stream_id,
+                 AudioProviderInterface* provider,
+                 StatsCollector* stats);
+
+  // Randomly generates id and stream_id.
+  AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
 
   virtual ~AudioRtpSender();
 
   // ObserverInterface implementation
   void OnChanged() override;
 
   // RtpSenderInterface implementation
   bool SetTrack(MediaStreamTrackInterface* track) override;
   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
     return track_.get();
   }
 
+  void SetSsrc(uint32_t ssrc) override;
+
+  uint32_t ssrc() const override { return ssrc_; }
+
+  std::string kind() const override { return "audio"; }
+
   std::string id() const override { return id_; }
 
+  void set_stream_id(const std::string& stream_id) override {
+    stream_id_ = stream_id;
+  }
+  std::string stream_id() const override { return stream_id_; }
+
   void Stop() override;
 
  private:
-  void Reconfigure();
+  // Helper function to construct options for
+  // AudioProviderInterface::SetAudioSend.
+  void SetAudioSend();
 
   std::string id_;
+  std::string stream_id_;
+  AudioProviderInterface* provider_;
+  StatsCollector* stats_;
   rtc::scoped_refptr<AudioTrackInterface> track_;
-  uint32_t ssrc_;
-  AudioProviderInterface* provider_;
-  bool cached_track_enabled_;
+  uint32_t ssrc_ = 0;

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

Can you put a comment about how 0 is "unset" and what the Sender does when it's
in that state?

[Taylor Brandstetter, 2015/10/21 00:22:08]

It's in rtpsenderinterface.h.

+  bool cached_track_enabled_ = false;
+  bool stopped_ = false;
 
   // Used to pass the data callback from the |track_| to the other end of
   // cricket::AudioRenderer.
   rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 class VideoRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   VideoRtpSender(VideoTrackInterface* track,
-                 uint32_t ssrc,
+                 const std::string& stream_id,
                  VideoProviderInterface* provider);
 
+  // Randomly generates id and stream_id.
+  explicit VideoRtpSender(VideoProviderInterface* provider);
+
   virtual ~VideoRtpSender();
 
   // ObserverInterface implementation
   void OnChanged() override;
 
   // RtpSenderInterface implementation
   bool SetTrack(MediaStreamTrackInterface* track) override;
   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
     return track_.get();
   }
 
+  void SetSsrc(uint32_t ssrc) override;
+
+  uint32_t ssrc() const override { return ssrc_; }
+
+  std::string kind() const override { return "video"; }
+
   std::string id() const override { return id_; }
 
+  void set_stream_id(const std::string& stream_id) override {
+    stream_id_ = stream_id;
+  }
+  std::string stream_id() const override { return stream_id_; }
+
   void Stop() override;
 
  private:
-  void Reconfigure();
+  // Helper function to construct options for
+  // VideoProviderInterface::SetVideoSend.
+  void SetVideoSend();
 
   std::string id_;
+  std::string stream_id_;
+  VideoProviderInterface* provider_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
-  uint32_t ssrc_;
-  VideoProviderInterface* provider_;
-  bool cached_track_enabled_;
+  uint32_t ssrc_ = 0;
+  bool cached_track_enabled_ = false;
+  bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDER_H_