Revert of 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
new file mode 100644
index 0000000000000000000000000000000000000000..5f441904bee6c823726f440bcd6d136452cffcec
--- /dev/null
+++ b/webrtc/modules/utility/source/coder.h
@@ -0,0 +1,68 @@
+/*
+ *  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_