Index: webrtc/modules/audio_processing/transient/transient_suppressor.cc |
diff --git a/webrtc/modules/audio_processing/transient/transient_suppressor.cc b/webrtc/modules/audio_processing/transient/transient_suppressor.cc |
index 46bb574c0be2f40aa4d80252c73c83543955f693..25909b90080de6742fcc4a69622f9bd2f428f340 100644 |
--- a/webrtc/modules/audio_processing/transient/transient_suppressor.cc |
+++ b/webrtc/modules/audio_processing/transient/transient_suppressor.cc |
@@ -17,7 +17,6 @@ |
#include <deque> |
#include <set> |
-#include "webrtc/base/checks.h" |
#include "webrtc/common_audio/fft4g.h" |
#include "webrtc/common_audio/include/audio_util.h" |
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
@@ -101,13 +100,13 @@ |
detector_.reset(new TransientDetector(detection_rate_hz)); |
data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000; |
if (data_length_ > analysis_length_) { |
- RTC_NOTREACHED(); |
+ assert(false); |
return -1; |
} |
buffer_delay_ = analysis_length_ - data_length_; |
complex_analysis_length_ = analysis_length_ / 2 + 1; |
- RTC_DCHECK_GE(complex_analysis_length_, kMaxVoiceBin); |
+ assert(complex_analysis_length_ >= kMaxVoiceBin); |
num_channels_ = num_channels; |
in_buffer_.reset(new float[analysis_length_ * num_channels_]); |
memset(in_buffer_.get(), |