Index: webrtc/common_audio/resampler/push_sinc_resampler.cc |
diff --git a/webrtc/common_audio/resampler/push_sinc_resampler.cc b/webrtc/common_audio/resampler/push_sinc_resampler.cc |
index 72ed56b86a99648a1f8661eae5a5deb342e06951..a740423eec3771a2cc5bd2f3871ca6a2867ee2df 100644 |
--- a/webrtc/common_audio/resampler/push_sinc_resampler.cc |
+++ b/webrtc/common_audio/resampler/push_sinc_resampler.cc |
@@ -50,8 +50,8 @@ size_t PushSincResampler::Resample(const float* source, |
size_t source_length, |
float* destination, |
size_t destination_capacity) { |
- CHECK_EQ(source_length, resampler_->request_frames()); |
- CHECK_GE(destination_capacity, destination_frames_); |
+ RTC_CHECK_EQ(source_length, resampler_->request_frames()); |
+ RTC_CHECK_GE(destination_capacity, destination_frames_); |
// Cache the source pointer. Calling Resample() will immediately trigger |
// the Run() callback whereupon we provide the cached value. |
source_ptr_ = source; |
@@ -81,7 +81,7 @@ size_t PushSincResampler::Resample(const float* source, |
void PushSincResampler::Run(size_t frames, float* destination) { |
// Ensure we are only asked for the available samples. This would fail if |
// Run() was triggered more than once per Resample() call. |
- CHECK_EQ(source_available_, frames); |
+ RTC_CHECK_EQ(source_available_, frames); |
if (first_pass_) { |
// Provide dummy input on the first pass, the output of which will be |