Index: webrtc/modules/audio_processing/ns/nsx_core_c.c |
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c |
index 213320d38c85f51a305f85b4a1a24053e11e7462..abfb2c9e3e7bcf6e97ed92d77c3bde94f89dbb8b 100644 |
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c |
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c |
@@ -8,8 +8,7 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
-#include <assert.h> |
- |
+#include "webrtc/base/checks.h" |
#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h" |
#include "webrtc/modules/audio_processing/ns/nsx_core.h" |
#include "webrtc/modules/audio_processing/ns/nsx_defines.h" |
@@ -149,7 +148,7 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst, |
if (inst->featureSpecDiff) { |
normTmp = WEBRTC_SPL_MIN(20 - inst->stages, |
WebRtcSpl_NormU32(inst->featureSpecDiff)); |
- assert(normTmp >= 0); |
+ RTC_DCHECK_GE(normTmp, 0); |
tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages) |
tmpU32no2 = inst->timeAvgMagnEnergy >> (20 - inst->stages - normTmp); |
if (tmpU32no2 > 0) { |