Index: webrtc/common_audio/resampler/push_resampler.cc |
diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc |
index 566acdeaa3c7ac670a768104f62a33546830c7c2..f654e9a3978802df327ddb5fd99eb2a431b2098f 100644 |
--- a/webrtc/common_audio/resampler/push_resampler.cc |
+++ b/webrtc/common_audio/resampler/push_resampler.cc |
@@ -32,7 +32,7 @@ PushResampler<T>::~PushResampler() { |
template <typename T> |
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz, |
int dst_sample_rate_hz, |
- int num_channels) { |
+ size_t num_channels) { |
if (src_sample_rate_hz == src_sample_rate_hz_ && |
dst_sample_rate_hz == dst_sample_rate_hz_ && |
num_channels == num_channels_) |
@@ -68,10 +68,8 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz, |
template <typename T> |
int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst, |
size_t dst_capacity) { |
- const size_t src_size_10ms = |
- static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100); |
- const size_t dst_size_10ms = |
- static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100); |
+ const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100; |
+ const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100; |
if (src_length != src_size_10ms || dst_capacity < dst_size_10ms) |
return -1; |