Move coder, file_player, and file_recorder to webrtc/voice_engine

webrtc/voice_engine/file_recorder.cc

 /*
  *  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.
  */
 
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_recorder.h"
 
 #include <list>
 
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/coder.h"
 
 namespace webrtc {
 
 namespace {
 
 // 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 };
 enum { kMaxAudioBufferQueueLength = 100 };
 
@@ skipping 218 matching lines @@
 }  // namespace
 
 std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
     uint32_t instanceID,
     FileFormats fileFormat) {
   return std::unique_ptr<FileRecorder>(
       new FileRecorderImpl(instanceID, fileFormat));
 }
 
 }  // namespace webrtc