| 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) {
|
|
|