Revert of 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
new file mode 100644
index 0000000000000000000000000000000000000000..562c4ad306c831f1fd7a3d1a88b56f76b7babdc0
--- /dev/null
+++ b/talk/app/webrtc/test/fakemediastreamsignaling.h
@@ -0,0 +1,140 @@
+/*
+ * 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_