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