Move FilePlayer and FileRecorder to Voice Engine

webrtc/modules/utility/source/coder.h

Index: webrtc/modules/utility/source/coder.h
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
deleted file mode 100644
index 5f441904bee6c823726f440bcd6d136452cffcec..0000000000000000000000000000000000000000
--- a/webrtc/modules/utility/source/coder.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 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_SOURCE_CODER_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
-
-#include <memory>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class AudioFrame;
-
-class AudioCoder : public AudioPacketizationCallback {
- public:
-  AudioCoder(uint32_t instance_id);
-  ~AudioCoder();
-
-  int32_t SetEncodeCodec(const CodecInst& codec_inst);
-
-  int32_t SetDecodeCodec(const CodecInst& codec_inst);
-
-  int32_t Decode(AudioFrame& decoded_audio,
-                 uint32_t samp_freq_hz,
-                 const int8_t* incoming_payload,
-                 size_t payload_length);
-
-  int32_t PlayoutData(AudioFrame& decoded_audio, uint16_t& samp_freq_hz);
-
-  int32_t Encode(const AudioFrame& audio,
-                 int8_t* encoded_data,
-                 size_t& encoded_length_in_bytes);
-
- protected:
-  int32_t SendData(FrameType frame_type,
-                   uint8_t payload_type,
-                   uint32_t time_stamp,
-                   const uint8_t* payload_data,
-                   size_t payload_size,
-                   const RTPFragmentationHeader* fragmentation) override;
-
- private:
-  std::unique_ptr<AudioCodingModule> acm_;
-  acm2::CodecManager codec_manager_;
-  acm2::RentACodec rent_a_codec_;
-
-  CodecInst receive_codec_;
-
-  uint32_t encode_timestamp_;
-  int8_t* encoded_data_;
-  size_t encoded_length_in_bytes_;
-
-  uint32_t decode_timestamp_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_