Refactor VideoTracks to forward all sinks to its source

talk/app/webrtc/objctests/RTCPeerConnectionTest.mm

 /*
  * 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,
@@ skipping 72 matching lines @@
   return [session1.type isEqual:session2.type];
 }
 
 - (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
                                  withFactory:(RTCPeerConnectionFactory*)factory
                                  videoSource:(RTCVideoSource*)videoSource
                                  streamLabel:(NSString*)streamLabel
                                 videoTrackID:(NSString*)videoTrackID
                                 audioTrackID:(NSString*)audioTrackID {
   RTCMediaStream* localMediaStream = [factory mediaStreamWithLabel:streamLabel];
-  RTCVideoTrack* videoTrack =
-      [factory videoTrackWithID:videoTrackID source:videoSource];
-  RTCFakeRenderer* videoRenderer = [[RTCFakeRenderer alloc] init];
-  [videoTrack addRenderer:videoRenderer];
-  [localMediaStream addVideoTrack:videoTrack];
-  // Test that removal/re-add works.
-  [localMediaStream removeVideoTrack:videoTrack];
-  [localMediaStream addVideoTrack:videoTrack];
+  // TODO(zeke): Fix this test to create a fake video capturer so that a track
+  // can be created.
+  if (videoSource) {
+    RTCVideoTrack* videoTrack =
+        [factory videoTrackWithID:videoTrackID source:videoSource];
+    RTCFakeRenderer* videoRenderer = [[RTCFakeRenderer alloc] init];
+    [videoTrack addRenderer:videoRenderer];
+    [localMediaStream addVideoTrack:videoTrack];
+    // Test that removal/re-add works.
+    [localMediaStream removeVideoTrack:videoTrack];
+    [localMediaStream addVideoTrack:videoTrack];
+  }
   RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID];
   [localMediaStream addAudioTrack:audioTrack];
   [pc addStream:localMediaStream];
   return localMediaStream;
 }
 
 - (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory {
   NSArray* mandatory = @[
     [[RTCPair alloc] initWithKey:@"DtlsSrtpKeyAgreement" value:@"true"],
     [[RTCPair alloc] initWithKey:@"internalSctpDataChannels" value:@"true"],
@@ skipping 226 matching lines @@
     // factory outlives RTCPeerConnection:dealloc.
     // See https://code.google.com/p/webrtc/issues/detail?id=3100.
     RTCPeerConnectionFactory* factory = [[RTCPeerConnectionFactory alloc] init];
     @autoreleasepool {
       RTCPeerConnectionTest* pcTest = [[RTCPeerConnectionTest alloc] init];
       [pcTest testCompleteSessionWithFactory:factory];
     }
     rtc::CleanupSSL();
   }
 }