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]; |