Index: webrtc/modules/audio_processing/gain_control_impl.cc |
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc |
index 81469dde01b3f59c278e7639f7f2b77d4d878371..704cfade097cb7797e059f5002b798baf9ad444f 100644 |
--- a/webrtc/modules/audio_processing/gain_control_impl.cc |
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc |
@@ -123,7 +123,7 @@ void GainControlImpl::ProcessRenderAudio( |
void GainControlImpl::PackRenderAudioBuffer( |
AudioBuffer* audio, |
std::vector<int16_t>* packed_buffer) { |
- RTC_DCHECK_GE(160u, audio->num_frames_per_band()); |
+ RTC_DCHECK_GE(160, audio->num_frames_per_band()); |
packed_buffer->clear(); |
packed_buffer->insert( |
@@ -139,7 +139,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { |
} |
RTC_DCHECK(num_proc_channels_); |
- RTC_DCHECK_GE(160u, audio->num_frames_per_band()); |
+ RTC_DCHECK_GE(160, audio->num_frames_per_band()); |
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_); |
RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size()); |
@@ -190,7 +190,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio, |
} |
RTC_DCHECK(num_proc_channels_); |
- RTC_DCHECK_GE(160u, audio->num_frames_per_band()); |
+ RTC_DCHECK_GE(160, audio->num_frames_per_band()); |
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_); |
stream_is_saturated_ = false; |