Lint enabled for webrtc/modules/video_coding folder.

webrtc/modules/video_coding/test/video_rtp_play.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 30 matching lines @@
   payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(
       kDefaultRedPayloadType, "RED", webrtc::kVideoCodecRED));
   payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(
       kDefaultVp8PayloadType, "VP8", webrtc::kVideoCodecVP8));
 
   std::string output_file = args.outputFile;
   if (output_file.empty())
     output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv";
 
   webrtc::SimulatedClock clock(0);
-  webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock,
-      kConfigProtectionEnabled, kConfigProtectionMethod, kConfigRttMs,
-      kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
+  webrtc::rtpplayer::VcmPayloadSinkFactory factory(
+      output_file, &clock, kConfigProtectionEnabled, kConfigProtectionMethod,
+      kConfigRttMs, kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
   rtc::scoped_ptr<webrtc::rtpplayer::RtpPlayerInterface> rtp_player(
       webrtc::rtpplayer::Create(args.inputFile, &factory, &clock, payload_types,
                                 kConfigLossRate, kConfigRttMs,
                                 kConfigReordering));
   if (rtp_player.get() == NULL) {
     return -1;
   }
 
   int ret = 0;
   while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
     ret = factory.DecodeAndProcessAll(true);
     if (ret < 0 || (kConfigMaxRuntimeMs > -1 &&
-        clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
+                    clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
       break;
     }
     clock.AdvanceTimeMilliseconds(1);
   }
 
   rtp_player->Print();
 
   switch (ret) {
     case 1:
       printf("Success\n");
       return 0;
     case -1:
       printf("Failed\n");
       return -1;
     case 0:
       printf("Timeout\n");
       return -1;
   }
 
   webrtc::Trace::ReturnTrace();
   return 0;
 }