| Index: webrtc/modules/audio_processing/ns/ns_core.c
|
| diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
|
| index 5ce64cee29eba0c2f8e65de989424afe2426e5f3..76589c5fe9172ad2b7ff9787f707e5db7b8e7b53 100644
|
| --- a/webrtc/modules/audio_processing/ns/ns_core.c
|
| +++ b/webrtc/modules/audio_processing/ns/ns_core.c
|
| @@ -8,11 +8,11 @@
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
|
|
| -#include <assert.h>
|
| #include <math.h>
|
| #include <string.h>
|
| #include <stdlib.h>
|
|
|
| +#include "webrtc/base/checks.h"
|
| #include "webrtc/common_audio/fft4g.h"
|
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
| #include "webrtc/modules/audio_processing/ns/noise_suppression.h"
|
| @@ -857,7 +857,7 @@ static void UpdateBuffer(const float* frame,
|
| size_t frame_length,
|
| size_t buffer_length,
|
| float* buffer) {
|
| - assert(buffer_length < 2 * frame_length);
|
| + RTC_DCHECK_LT(buffer_length, 2 * frame_length);
|
|
|
| memcpy(buffer,
|
| buffer + frame_length,
|
| @@ -893,7 +893,7 @@ static void FFT(NoiseSuppressionC* self,
|
| float* magn) {
|
| size_t i;
|
|
|
| - assert(magnitude_length == time_data_length / 2 + 1);
|
| + RTC_DCHECK_EQ(magnitude_length, time_data_length / 2 + 1);
|
|
|
| WebRtc_rdft(time_data_length, 1, time_data, self->ip, self->wfft);
|
|
|
| @@ -929,7 +929,7 @@ static void IFFT(NoiseSuppressionC* self,
|
| float* time_data) {
|
| size_t i;
|
|
|
| - assert(time_data_length == 2 * (magnitude_length - 1));
|
| + RTC_DCHECK_EQ(time_data_length, 2 * (magnitude_length - 1));
|
|
|
| time_data[0] = real[0];
|
| time_data[1] = real[magnitude_length - 1];
|
| @@ -1062,7 +1062,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
|
| float parametric_num = 0.0;
|
|
|
| // Check that initiation has been done.
|
| - assert(self->initFlag == 1);
|
| + RTC_DCHECK_EQ(1, self->initFlag);
|
| updateParsFlag = self->modelUpdatePars[0];
|
|
|
| // Update analysis buffer for L band.
|
| @@ -1206,8 +1206,8 @@ void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
|
| float sumMagnAnalyze, sumMagnProcess;
|
|
|
| // Check that initiation has been done.
|
| - assert(self->initFlag == 1);
|
| - assert((num_bands - 1) <= NUM_HIGH_BANDS_MAX);
|
| + RTC_DCHECK_EQ(1, self->initFlag);
|
| + RTC_DCHECK_LE(num_bands - 1, NUM_HIGH_BANDS_MAX);
|
|
|
| const float* const* speechFrameHB = NULL;
|
| float* const* outFrameHB = NULL;
|
|
|