Update audio code to use size_t more correctly, webrtc/modules/ portion.

webrtc/modules/utility/source/file_player_impl.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 77 matching lines @@
 }
 
 int32_t FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const
 {
     audioCodec = _codec;
     return 0;
 }
 
 int32_t FilePlayerImpl::Get10msAudioFromFile(
     int16_t* outBuffer,
-    int& lengthInSamples,
+    size_t& lengthInSamples,
     int frequencyInHz)
 {
     if(_codec.plfreq == 0)
     {
         LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!"
                         << " codec freq = " << _codec.plfreq
                         << ", wanted freq = " << frequencyInHz;
         return -1;
     }
 
@@ skipping 11 matching lines @@
         {
             // End of file reached.
             return -1;
         }
         if(lengthInBytes == 0)
         {
             lengthInSamples = 0;
             return 0;
         }
         // One sample is two bytes.
-        unresampledAudioFrame.samples_per_channel_ =
-            (uint16_t)lengthInBytes >> 1;
+        unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
 
     } else {
         // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
         // expects a full frame. If the frame size is larger than 10 ms,
         // PlayoutAudioData(..) data should be called proportionally less often.
         int16_t encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
         size_t encodedLengthInBytes = 0;
         if(++_numberOf10MsInDecoder >= _numberOf10MsPerFrame)
         {
             _numberOf10MsInDecoder = 0;
             size_t bytesFromFile = sizeof(encodedBuffer);
             if (_fileModule.PlayoutAudioData((int8_t*)encodedBuffer,
                                              bytesFromFile) == -1)
             {
                 // End of file reached.
                 return -1;
             }
             encodedLengthInBytes = bytesFromFile;
         }
         if(_audioDecoder.Decode(unresampledAudioFrame,frequencyInHz,
                                 (int8_t*)encodedBuffer,
                                 encodedLengthInBytes) == -1)
         {
             return -1;
         }
     }
 
-    int outLen = 0;
+    size_t outLen = 0;
     if(_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
                                 frequencyInHz, 1))
     {
         LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
 
         // New sampling frequency. Update state.
-        outLen = frequencyInHz / 100;
+        outLen = static_cast<size_t>(frequencyInHz / 100);
         memset(outBuffer, 0, outLen * sizeof(int16_t));
         return 0;
     }
     _resampler.Push(unresampledAudioFrame.data_,
                     unresampledAudioFrame.samples_per_channel_,
                     outBuffer,
                     MAX_AUDIO_BUFFER_IN_SAMPLES,
                     outLen);
 
     lengthInSamples = outLen;
 
     if(_scaling != 1.0)
     {
-        for (int i = 0;i < outLen; i++)
+        for (size_t i = 0;i < outLen; i++)
         {
             outBuffer[i] = (int16_t)(outBuffer[i] * _scaling);
         }
     }
     _decodedLengthInMS += 10;
     return 0;
 }
 
 int32_t FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback)
 {
@@ skipping 203 matching lines @@
     {
         LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname
                         << " not supported.";
         return -1;
     }
     _numberOf10MsPerFrame = _codec.pacsize / (_codec.plfreq / 100);
     _numberOf10MsInDecoder = 0;
     return 0;
 }
 }  // namespace webrtc