Adding the ability to create an RtpSender without a track.

talk/app/webrtc/test/fakemediastreamsignaling.h

Index: talk/app/webrtc/test/fakemediastreamsignaling.h
diff --git a/talk/app/webrtc/test/fakemediastreamsignaling.h b/talk/app/webrtc/test/fakemediastreamsignaling.h
deleted file mode 100644
index 562c4ad306c831f1fd7a3d1a88b56f76b7babdc0..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/test/fakemediastreamsignaling.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*

[Taylor Brandstetter, 2015/10/19 23:51:34]

I could have deleted this in the MediaStreamSignaling removal CL, I just forgot.

- * libjingle
- * Copyright 2013 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,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
-#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
-
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
-#include "talk/app/webrtc/videotrack.h"
-
-static const char kStream1[] = "stream1";
-static const char kVideoTrack1[] = "video1";
-static const char kAudioTrack1[] = "audio1";
-
-static const char kStream2[] = "stream2";
-static const char kVideoTrack2[] = "video2";
-static const char kAudioTrack2[] = "audio2";
-
-class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
-                                 public webrtc::MediaStreamSignalingObserver {
- public:
-  explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
-    webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
-                                 channel_manager) {
-  }
-
-  void SendAudioVideoStream1() {
-    ClearLocalStreams();
-    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
-  }
-
-  void SendAudioVideoStream2() {
-    ClearLocalStreams();
-    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
-  }
-
-  void SendAudioVideoStream1And2() {
-    ClearLocalStreams();
-    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
-    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
-  }
-
-  void SendNothing() {
-    ClearLocalStreams();
-  }
-
-  void UseOptionsAudioOnly() {
-    ClearLocalStreams();
-    AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
-  }
-
-  void UseOptionsVideoOnly() {
-    ClearLocalStreams();
-    AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
-  }
-
-  void ClearLocalStreams() {
-    while (local_streams()->count() != 0) {
-      RemoveLocalStream(local_streams()->at(0));
-    }
-  }
-
-  // Implements MediaStreamSignalingObserver.
-  virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}
-  virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}
-  virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}
-  virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
-                                    webrtc::AudioTrackInterface* audio_track,
-                                    uint32_t ssrc) {}
-  virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
-                                    webrtc::VideoTrackInterface* video_track,
-                                    uint32_t ssrc) {}
-  virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
-                                     webrtc::AudioTrackInterface* audio_track,
-                                     uint32_t ssrc) {}
-  virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
-                                     webrtc::VideoTrackInterface* video_track,
-                                     uint32_t ssrc) {}
-  virtual void OnRemoveRemoteAudioTrack(
-      webrtc::MediaStreamInterface* stream,
-      webrtc::AudioTrackInterface* audio_track) {}
-  virtual void OnRemoveRemoteVideoTrack(
-      webrtc::MediaStreamInterface* stream,
-      webrtc::VideoTrackInterface* video_track) {}
-  virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,
-                                       webrtc::AudioTrackInterface* audio_track,
-                                       uint32_t ssrc) {}
-  virtual void OnRemoveLocalVideoTrack(
-      webrtc::MediaStreamInterface* stream,
-      webrtc::VideoTrackInterface* video_track) {}
-  virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}
-
- private:
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
-      const std::string& stream_label,
-      const std::string& audio_track_id,
-      const std::string& video_track_id) {
-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
-        webrtc::MediaStream::Create(stream_label));
-
-    if (!audio_track_id.empty()) {
-      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-          webrtc::AudioTrack::Create(audio_track_id, NULL));
-      stream->AddTrack(audio_track);
-    }
-
-    if (!video_track_id.empty()) {
-      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
-          webrtc::VideoTrack::Create(video_track_id, NULL));
-      stream->AddTrack(video_track);
-    }
-    return stream;
-  }
-};
-
-#endif  // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_