Fix trivial lint errors in FileRecorder and FilePlayer

webrtc/modules/utility/source/coder.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.
  */
 
@@ skipping 36 matching lines @@
 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
   if (acm_->RegisterReceiveCodec(codec_inst, [&] {
         return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
       }) == -1) {
     return -1;
   }
   memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
   return 0;
 }
 
-int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
+int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
                            uint32_t samp_freq_hz,
                            const int8_t* incoming_payload,
                            size_t payload_length) {
   if (payload_length > 0) {
     const uint8_t payload_type = receive_codec_.pltype;
     decode_timestamp_ += receive_codec_.pacsize;
     if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
                               payload_type, decode_timestamp_) == -1) {
       return -1;
     }
   }
   bool muted;
   int32_t ret =
-      acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
+      acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
   RTC_DCHECK(!muted);
   return ret;
 }
 
-int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
-                                uint16_t& samp_freq_hz) {
+int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
+                                uint16_t samp_freq_hz) {
   bool muted;
-  int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
+  int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
   RTC_DCHECK(!muted);
   return ret;
 }
 
 int32_t AudioCoder::Encode(const AudioFrame& audio,
                            int8_t* encoded_data,
-                           size_t& encoded_length_in_bytes) {
+                           size_t* encoded_length_in_bytes) {
   // Fake a timestamp in case audio doesn't contain a correct timestamp.
   // Make a local copy of the audio frame since audio is const
   AudioFrame audio_frame;
   audio_frame.CopyFrom(audio);
   audio_frame.timestamp_ = encode_timestamp_;
   encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
 
   // For any codec with a frame size that is longer than 10 ms the encoded
   // length in bytes should be zero until a a full frame has been encoded.
   encoded_length_in_bytes_ = 0;
   if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
     return -1;
   }
   encoded_data_ = encoded_data;
-  encoded_length_in_bytes = encoded_length_in_bytes_;
+  *encoded_length_in_bytes = encoded_length_in_bytes_;
   return 0;
 }
 
 int32_t AudioCoder::SendData(FrameType /* frame_type */,
                              uint8_t /* payload_type */,
                              uint32_t /* time_stamp */,
                              const uint8_t* payload_data,
                              size_t payload_size,
                              const RTPFragmentationHeader* /* fragmentation*/) {
   memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
   encoded_length_in_bytes_ = payload_size;
   return 0;
 }
 
 }  // namespace webrtc