Move FilePlayer and FileRecorder to Voice Engine

webrtc/modules/utility/include/file_player.h

Index: webrtc/modules/utility/include/file_player.h
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h
deleted file mode 100644
index 1adbb9d70e3d28d6bd660edf5c1db7ab16ce8ddf..0000000000000000000000000000000000000000
--- a/webrtc/modules/utility/include/file_player.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_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 (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|
-  // 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.
-  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,
-                                   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;
-
-  // 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_