Revert of Supporting 48kHz PCM file.

webrtc/voice_engine/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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
 #define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
 
 #include <memory>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class FileCallback;
 
 class FilePlayer {
  public:
-  // The largest decoded frame size in samples (120ms with 48kHz sample rate).
-  enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 120 * 48 };
+  // 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);
 
   virtual ~FilePlayer() = default;
 
   // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
@@ skipping 33 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_VOICE_ENGINE_FILE_PLAYER_H_