| Index: webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| diff --git a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| index 2450e05b768027918c2f90f453000041adbd8988..d2155648d63d2015eb72f90bf491791418045e64 100644
|
| --- a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| +++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| @@ -19,12 +19,13 @@
|
|
|
| #include "webrtc/modules/audio_processing/agc/legacy/analog_agc.h"
|
|
|
| -#include <assert.h>
|
| #include <stdlib.h>
|
| #ifdef WEBRTC_AGC_DEBUG_DUMP
|
| #include <stdio.h>
|
| #endif
|
|
|
| +#include "webrtc/base/checks.h"
|
| +
|
| /* The slope of in Q13*/
|
| static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129,
|
| 2372, 1362, 472, 78};
|
| @@ -155,14 +156,14 @@ int WebRtcAgc_AddMic(void* state,
|
| if (stt->micVol > stt->maxAnalog) {
|
| /* |maxLevel| is strictly >= |micVol|, so this condition should be
|
| * satisfied here, ensuring there is no divide-by-zero. */
|
| - assert(stt->maxLevel > stt->maxAnalog);
|
| + RTC_DCHECK_GT(stt->maxLevel, stt->maxAnalog);
|
|
|
| /* Q1 */
|
| tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
|
| tmp32 = (GAIN_TBL_LEN - 1) * tmp16;
|
| tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
|
| targetGainIdx = tmp32 / tmp16;
|
| - assert(targetGainIdx < GAIN_TBL_LEN);
|
| + RTC_DCHECK_LT(targetGainIdx, GAIN_TBL_LEN);
|
|
|
| /* Increment through the table towards the target gain.
|
| * If micVol drops below maxAnalog, we allow the gain
|
|
|