Lint fix for webrtc/modules/video_coding PART 3!

webrtc/modules/video_coding/test/rtp_player.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 class Clock;
 
 namespace rtpplayer {
 
 class PayloadCodecTuple {
  public:
-  PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
+  PayloadCodecTuple(uint8_t payload_type,
+                    const std::string& codec_name,
                     VideoCodecType codec_type)
       : name_(codec_name),
         payload_type_(payload_type),
-        codec_type_(codec_type) {
-  }
+        codec_type_(codec_type) {}
 
   const std::string& name() const { return name_; }
   uint8_t payload_type() const { return payload_type_; }
   VideoCodecType codec_type() const { return codec_type_; }
 
  private:
   std::string name_;
   uint8_t payload_type_;
   VideoCodecType codec_type_;
 };
@@ skipping 37 matching lines @@
 class RtpPlayerInterface {
  public:
   virtual ~RtpPlayerInterface() {}
 
   virtual int NextPacket(int64_t timeNow) = 0;
   virtual uint32_t TimeUntilNextPacket() const = 0;
   virtual void Print() const = 0;
 };
 
 RtpPlayerInterface* Create(const std::string& inputFilename,
-    PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
-    const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
-    bool reordering);
+                           PayloadSinkFactoryInterface* payloadSinkFactory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float lossRate,
+                           int64_t rttMs,
+                           bool reordering);
 
 }  // namespace rtpplayer
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_