Fix trivial lint errors in FileRecorder and FilePlayer

webrtc/modules/utility/source/coder.h

 /*
  *  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);
+  explicit 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,
+  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 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);
+                 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_