Remove deprecated function NewFilePlayer

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 15 matching lines @@
  public:
   // The largest decoded frame size in samples (60ms with 32kHz sample rate).
   enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
   enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
 
   // Note: will return NULL for unsupported formats.
   static std::unique_ptr<FilePlayer> CreateFilePlayer(
       const uint32_t instanceID,
       const FileFormats fileFormat);
 
-  // Deprecated. Call CreateFilePlayer instead.
-  static std::unique_ptr<FilePlayer> NewFilePlayer(
-      const uint32_t instanceID,
-      const FileFormats fileFormat) {
-    return CreateFilePlayer(instanceID, fileFormat);
-  }
-
   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,
                                    int frequencyInHz) = 0;
 
   // Register callback for receiving file playing notifications.
@@ skipping 26 matching lines @@
   // Set audioCodec to the currently used audio codec.
   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;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_