Fix trivial lint errors in FileRecorder and FilePlayer

webrtc/modules/utility/include/file_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.
  */
 
@@ skipping 26 matching lines @@
   static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
                                       const FileFormats fileFormat);
   static void DestroyFilePlayer(FilePlayer* player);
 
   virtual ~FilePlayer() = default;
 
   // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
   // will be set to the number of samples read (not the number of samples per
   // channel).
   virtual int Get10msAudioFromFile(int16_t* outBuffer,
-                                   size_t& lengthInSamples,
+                                   size_t* lengthInSamples,
                                    int frequencyInHz) = 0;
 
   // Register callback for receiving file playing notifications.
   virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
 
   // API for playing audio from fileName to channel.
   // Note: codecInst is used for pre-encoded files.
   virtual int32_t StartPlayingFile(const char* fileName,
                                    bool loop,
                                    uint32_t startPosition,
                                    float volumeScaling,
                                    uint32_t notification,
                                    uint32_t stopPosition,
                                    const CodecInst* codecInst) = 0;
 
   // Note: codecInst is used for pre-encoded files.
-  virtual int32_t StartPlayingFile(InStream& sourceStream,
+  virtual int32_t StartPlayingFile(InStream* sourceStream,
                                    uint32_t startPosition,
                                    float volumeScaling,
                                    uint32_t notification,
                                    uint32_t stopPosition,
                                    const CodecInst* codecInst) = 0;
 
   virtual int32_t StopPlayingFile() = 0;
 
   virtual bool IsPlayingFile() const = 0;
 
-  virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+  virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
 
   // Set audioCodec to the currently used audio codec.
-  virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+  virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
 
   virtual int32_t Frequency() const = 0;
 
   // Note: scaleFactor is in the range [0.0 - 2.0]
   virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+
+  // Deprecated functions. Use the functions above with the same name instead.
+  int Get10msAudioFromFile(int16_t* outBuffer,
+                           size_t& lengthInSamples,
+                           int frequencyInHz) {
+    return Get10msAudioFromFile(outBuffer, &lengthInSamples, frequencyInHz);
+  }
+  int32_t StartPlayingFile(InStream& sourceStream,
+                           uint32_t startPosition,
+                           float volumeScaling,
+                           uint32_t notification,
+                           uint32_t stopPosition,
+                           const CodecInst* codecInst) {
+    return StartPlayingFile(&sourceStream, startPosition, volumeScaling,
+                            notification, stopPosition, codecInst);
+  }
+  int32_t GetPlayoutPosition(uint32_t& durationMs) {
+    return GetPlayoutPosition(&durationMs);
+  }
+  int32_t AudioCodec(CodecInst& audioCodec) const {
+    return AudioCodec(&audioCodec);
+  }
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_