| Index: webrtc/modules/audio_processing/test/test_utils.h
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| diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
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| index 7ad462c0957e321cebb5d808eea7d41ee756f069..11ef236764101d6204e97a774271cd4130b78e60 100644
 | 
| --- a/webrtc/modules/audio_processing/test/test_utils.h
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| +++ b/webrtc/modules/audio_processing/test/test_utils.h
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| @@ -49,8 +49,8 @@ void WriteIntData(const int16_t* data,
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|                    RawFile* raw_file);
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|  
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|  void WriteFloatData(const float* const* data,
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| -                    int samples_per_channel,
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| -                    int num_channels,
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| +                    size_t samples_per_channel,
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| +                    size_t num_channels,
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|                      WavWriter* wav_file,
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|                      RawFile* raw_file);
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|  
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| @@ -64,7 +64,7 @@ void SetFrameSampleRate(AudioFrame* frame,
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|  
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|  template <typename T>
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|  void SetContainerFormat(int sample_rate_hz,
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| -                        int num_channels,
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| +                        size_t num_channels,
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|                          AudioFrame* frame,
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|                          rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
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|    SetFrameSampleRate(frame, sample_rate_hz);
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| @@ -72,14 +72,14 @@ void SetContainerFormat(int sample_rate_hz,
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|    cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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|  }
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|  
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| -AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
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| +AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
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|  
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|  template <typename T>
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| -float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
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| +float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
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|    float mse = 0;
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|    float mean = 0;
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|    *variance = 0;
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| -  for (int i = 0; i < length; ++i) {
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| +  for (size_t i = 0; i < length; ++i) {
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|      T error = ref[i] - test[i];
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|      mse += error * error;
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|      *variance += ref[i] * ref[i];
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| 
 |