diff --git a/webrtc/common_audio/signal_processing/auto_correlation.c b/webrtc/common_audio/signal_processing/auto_correlation.c
index fda4fffeed..58e6d6e0a9 100644
--- a/webrtc/common_audio/signal_processing/auto_correlation.c
+++ b/webrtc/common_audio/signal_processing/auto_correlation.c
@@ -10,7 +10,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-#include <assert.h>
+#include "webrtc/base/checks.h"
 
 size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
                                  size_t in_vector_length,
@@ -22,7 +22,7 @@ size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
   int16_t smax = 0;
   int scaling = 0;
 
-  assert(order <= in_vector_length);
+  RTC_DCHECK_LE(order, in_vector_length);
 
   // Find the maximum absolute value of the samples.
   smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
index 70001a0882..53e800bc63 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // TODO(bjornv): Change the return type to report errors.
@@ -21,8 +21,8 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
   size_t i = 0;
   size_t j = 0;
 
-  assert(data_length > 0);
-  assert(coefficients_length > 1);
+  RTC_DCHECK_GT(data_length, 0);
+  RTC_DCHECK_GT(coefficients_length, 1);
 
   for (i = 0; i < data_length; i++) {
     int32_t output = 0;
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c b/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
index 03847018e3..02fa80b056 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
@@ -25,8 +25,8 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
   int min16 = 0xFFFF8000;
 #endif  // #if !defined(MIPS_DSP_R1_LE)
 
-  assert(data_length > 0);
-  assert(coefficients_length > 1);
+  RTC_DCHECK_GT(data_length, 0);
+  RTC_DCHECK_GT(coefficients_length, 1);
 
   __asm __volatile (
     ".set       push                                             \n\t"
diff --git a/webrtc/common_audio/signal_processing/min_max_operations.c b/webrtc/common_audio/signal_processing/min_max_operations.c
index 4a962f86a0..bc23a9c04d 100644
--- a/webrtc/common_audio/signal_processing/min_max_operations.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations.c
@@ -24,9 +24,9 @@
  *
  */
 
-#include <assert.h>
 #include <stdlib.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // TODO(bjorn/kma): Consolidate function pairs (e.g. combine
@@ -38,7 +38,7 @@ int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length) {
   size_t i = 0;
   int absolute = 0, maximum = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -64,7 +64,7 @@ int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length) {
   uint32_t absolute = 0, maximum = 0;
   size_t i = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -83,7 +83,7 @@ int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length) {
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
   size_t i = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum)
@@ -97,7 +97,7 @@ int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length) {
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
   size_t i = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum)
@@ -111,7 +111,7 @@ int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length) {
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
   size_t i = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum)
@@ -125,7 +125,7 @@ int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length) {
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
   size_t i = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum)
@@ -141,7 +141,7 @@ size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length) {
   size_t i = 0, index = 0;
   int absolute = 0, maximum = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -160,7 +160,7 @@ size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length) {
   size_t i = 0, index = 0;
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum) {
@@ -177,7 +177,7 @@ size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length) {
   size_t i = 0, index = 0;
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum) {
@@ -194,7 +194,7 @@ size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length) {
   size_t i = 0, index = 0;
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum) {
@@ -211,7 +211,7 @@ size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length) {
   size_t i = 0, index = 0;
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum) {
diff --git a/webrtc/common_audio/signal_processing/min_max_operations_mips.c b/webrtc/common_audio/signal_processing/min_max_operations_mips.c
index 28de45b3a5..c769e6a9d4 100644
--- a/webrtc/common_audio/signal_processing/min_max_operations_mips.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations_mips.c
@@ -16,8 +16,7 @@
  *
  */
 
-#include <assert.h>
-
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // Maximum absolute value of word16 vector.
@@ -26,7 +25,7 @@ int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length) {
   int32_t tmp32_0, tmp32_1, tmp32_2, tmp32_3;
   size_t i, loop_size;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
 #if defined(MIPS_DSP_R1)
   const int32_t* tmpvec32 = (int32_t*)vector;
@@ -230,7 +229,7 @@ int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length) {
   uint32_t absolute = 0, maximum = 0;
   int tmp1 = 0, max_value = 0x7fffffff;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -264,7 +263,7 @@ int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length) {
   int tmp1;
   int16_t value;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -292,7 +291,7 @@ int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length) {
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
   int tmp1, value;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -322,7 +321,7 @@ int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length) {
   int tmp1;
   int16_t value;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -351,7 +350,7 @@ int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length) {
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
   int tmp1, value;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
diff --git a/webrtc/common_audio/signal_processing/min_max_operations_neon.c b/webrtc/common_audio/signal_processing/min_max_operations_neon.c
index 6fbbf94ee0..d5aad764a6 100644
--- a/webrtc/common_audio/signal_processing/min_max_operations_neon.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations_neon.c
@@ -9,16 +9,16 @@
  */
 
 #include <arm_neon.h>
-#include <assert.h>
 #include <stdlib.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // Maximum absolute value of word16 vector. C version for generic platforms.
 int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length) {
   int absolute = 0, maximum = 0;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int16_t* p_start = vector;
   size_t rest = length & 7;
@@ -76,7 +76,7 @@ int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, size_t length) {
   size_t i = 0;
   size_t residual = length & 0x7;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int32_t* p_start = vector;
   uint32x4_t max32x4_0 = vdupq_n_u32(0);
@@ -128,7 +128,7 @@ int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length) {
   size_t i = 0;
   size_t residual = length & 0x7;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int16_t* p_start = vector;
   int16x8_t max16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MIN);
@@ -166,7 +166,7 @@ int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length) {
   size_t i = 0;
   size_t residual = length & 0x7;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int32_t* p_start = vector;
   int32x4_t max32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MIN);
@@ -208,7 +208,7 @@ int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length) {
   size_t i = 0;
   size_t residual = length & 0x7;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int16_t* p_start = vector;
   int16x8_t min16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MAX);
@@ -246,7 +246,7 @@ int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length) {
   size_t i = 0;
   size_t residual = length & 0x7;
 
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0);
 
   const int32_t* p_start = vector;
   int32x4_t min32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MAX);
diff --git a/webrtc/common_audio/signal_processing/spl_sqrt.c b/webrtc/common_audio/signal_processing/spl_sqrt.c
index 579e714a11..511039b650 100644
--- a/webrtc/common_audio/signal_processing/spl_sqrt.c
+++ b/webrtc/common_audio/signal_processing/spl_sqrt.c
@@ -15,10 +15,9 @@
  *
  */
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-#include <assert.h>
-
 int32_t WebRtcSpl_SqrtLocal(int32_t in);
 
 int32_t WebRtcSpl_SqrtLocal(int32_t in)
@@ -166,7 +165,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
     x_norm = (int16_t)(A >> 16);  // x_norm = AH
 
     nshift = (sh / 2);
-    assert(nshift >= 0);
+    RTC_DCHECK_GE(nshift, 0);
 
     A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
     A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
diff --git a/webrtc/common_audio/signal_processing/splitting_filter.c b/webrtc/common_audio/signal_processing/splitting_filter.c
index ba6e77dc6f..1400623a72 100644
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@@ -13,10 +13,9 @@
  *
  */
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-#include <assert.h>
-
 // Maximum number of samples in a low/high-band frame.
 enum
 {
@@ -136,8 +135,8 @@ void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
     const size_t band_length = in_data_length / 2;
-    assert(in_data_length % 2 == 0);
-    assert(band_length <= kMaxBandFrameLength);
+    RTC_DCHECK_EQ(0, in_data_length % 2);
+    RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
 
     // Split even and odd samples. Also shift them to Q10.
     for (i = 0, k = 0; i < band_length; i++, k += 2)
@@ -175,7 +174,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
     int32_t filter2[kMaxBandFrameLength];
     size_t i;
     int16_t k;
-    assert(band_length <= kMaxBandFrameLength);
+    RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
 
     // Obtain the sum and difference channels out of upper and lower-band channels.
     // Also shift to Q10 domain.
diff --git a/webrtc/common_audio/vad/vad_filterbank.c b/webrtc/common_audio/vad/vad_filterbank.c
index 8b9df93b00..5e15696ee1 100644
--- a/webrtc/common_audio/vad/vad_filterbank.c
+++ b/webrtc/common_audio/vad/vad_filterbank.c
@@ -10,8 +10,7 @@
 
 #include "webrtc/common_audio/vad/vad_filterbank.h"
 
-#include <assert.h>
-
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/typedefs.h"
 
@@ -160,8 +159,8 @@ static void LogOfEnergy(const int16_t* data_in, size_t data_length,
   // we eventually will mask out the fractional part.
   uint32_t energy = 0;
 
-  assert(data_in != NULL);
-  assert(data_length > 0);
+  RTC_DCHECK(data_in);
+  RTC_DCHECK_GT(data_length, 0);
 
   energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
                                        &tot_rshifts);
@@ -261,8 +260,8 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
   int16_t* hp_out_ptr = hp_120;  // [2000 - 4000] Hz.
   int16_t* lp_out_ptr = lp_120;  // [0 - 2000] Hz.
 
-  assert(data_length <= 240);
-  assert(4 < kNumChannels - 1);  // Checking maximum |frequency_band|.
+  RTC_DCHECK_LE(data_length, 240);
+  RTC_DCHECK_LT(4, kNumChannels - 1);  // Checking maximum |frequency_band|.
 
   // Split at 2000 Hz and downsample.
   SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
diff --git a/webrtc/common_audio/vad/vad_sp.c b/webrtc/common_audio/vad/vad_sp.c
index a54be17daa..4a1cebb0d5 100644
--- a/webrtc/common_audio/vad/vad_sp.c
+++ b/webrtc/common_audio/vad/vad_sp.c
@@ -10,8 +10,7 @@
 
 #include "webrtc/common_audio/vad/vad_sp.h"
 
-#include <assert.h>
-
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/vad_core.h"
 #include "webrtc/typedefs.h"
@@ -72,7 +71,7 @@ int16_t WebRtcVad_FindMinimum(VadInstT* self,
   int16_t* age = &self->index_vector[offset];
   int16_t* smallest_values = &self->low_value_vector[offset];
 
-  assert(channel < kNumChannels);
+  RTC_DCHECK_LT(channel, kNumChannels);
 
   // Each value in |smallest_values| is getting 1 loop older. Update |age|, and
   // remove old values.
diff --git a/webrtc/modules/audio_coding/codecs/isac/empty.cc b/webrtc/modules/audio_coding/codecs/isac/empty.cc
new file mode 100644
index 0000000000..e69de29bb2
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index b074962eae..fa63b46d2e 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -20,9 +20,8 @@
 
 #include "bandwidth_estimator.h"
 
-#include <assert.h>
 #include "settings.h"
-
+#include "webrtc/base/checks.h"
 
 /* array of quantization levels for bottle neck info; Matlab code: */
 /* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
@@ -180,7 +179,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
   int16_t errCode;
 
-  assert(!bweStr->external_bw_info.in_use);
+  RTC_DCHECK(!bweStr->external_bw_info.in_use);
 
   /* UPDATE ESTIMATES FROM OTHER SIDE */
 
@@ -551,7 +550,7 @@ int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
 {
   uint16_t RateInd;
 
-  assert(!bweStr->external_bw_info.in_use);
+  RTC_DCHECK(!bweStr->external_bw_info.in_use);
 
   if ( (Index < 0) || (Index > 23) ) {
     return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
@@ -732,7 +731,7 @@ uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
   int32_t   rec_jitter_short_term_abs_inv; /* Q18 */
   int32_t   temp;
 
-  assert(!bweStr->external_bw_info.in_use);
+  RTC_DCHECK(!bweStr->external_bw_info.in_use);
 
   /* Q18  rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
      2^18 then needs to be shifted 13 bits to 2^31 */
@@ -790,7 +789,7 @@ int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
 {
   int16_t recMaxDelay = (int16_t)(bweStr->recMaxDelay >> 15);
 
-  assert(!bweStr->external_bw_info.in_use);
+  RTC_DCHECK(!bweStr->external_bw_info.in_use);
 
   /* limit range of jitter estimate */
   if (recMaxDelay < MIN_ISAC_MD) {
@@ -804,7 +803,7 @@ int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
 
 /* Clamp val to the closed interval [min,max]. */
 static int16_t clamp(int16_t val, int16_t min, int16_t max) {
-  assert(min <= max);
+  RTC_DCHECK_LE(min, max);
   return val < min ? min : (val > max ? max : val);
 }
 
@@ -822,7 +821,7 @@ int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bweStr) {
 
 void WebRtcIsacfixBw_GetBandwidthInfo(BwEstimatorstr* bweStr,
                                    IsacBandwidthInfo* bwinfo) {
-  assert(!bweStr->external_bw_info.in_use);
+  RTC_DCHECK(!bweStr->external_bw_info.in_use);
   bwinfo->in_use = 1;
   bwinfo->send_bw_avg = WebRtcIsacfix_GetUplinkBandwidth(bweStr);
   bwinfo->send_max_delay_avg = WebRtcIsacfix_GetUplinkMaxDelay(bweStr);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
index 757c0b85c8..248511fec0 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -15,9 +15,9 @@
  *
  */
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 
-#include <assert.h>
 #include <stdio.h>
 
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
@@ -455,7 +455,7 @@ int WebRtcIsacfix_EncodeImpl(int16_t      *in,
 
     while (stream_length < MinBytes)
     {
-      assert(stream_length >= 0);
+      RTC_DCHECK_GE(stream_length, 0);
       if (stream_length & 0x0001){
         ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
         ISACenc_obj->bitstr_obj.stream[stream_length / 2] |=
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
index 0f01a030cb..1e4812aad3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
@@ -17,10 +17,10 @@
 #include "entropy_coding.h"
 
 #include <arm_neon.h>
-#include <assert.h>
 #include <stddef.h>
 
 #include "signal_processing_library.h"
+#include "webrtc/base/checks.h"
 
 void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
                                       const int32_t matrix1[],
@@ -46,8 +46,8 @@ void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
   int32x4_t sum_32x4 =  vdupq_n_s32(0);
   int32x2_t sum_32x2 =  vdup_n_s32(0);
 
-  assert(inner_loop_count % 2 == 0);
-  assert(mid_loop_count % 2 == 0);
+  RTC_DCHECK_EQ(0, inner_loop_count % 2);
+  RTC_DCHECK_EQ(0, mid_loop_count % 2);
 
   if (matrix1_index_init_case != 0 && matrix1_index_factor1 == 1) {
     for (j = 0; j < SUBFRAMES; j++) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
index 2e92578cf7..ce479e2777 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
@@ -20,11 +20,10 @@
 
 #include "filterbank_internal.h"
 
-#include <assert.h>
-
 #include "codec.h"
 #include "filterbank_tables.h"
 #include "settings.h"
+#include "webrtc/base/checks.h"
 
 // Declare a function pointer.
 AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16;
@@ -44,7 +43,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C(
   int32_t a = 0, b = 0;
 
   // Assembly file assumption.
-  assert(length % 2 == 0);
+  RTC_DCHECK_EQ(0, length % 2);
 
   for (n = 0; n < length; n++) {
     // Process channel 1:
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
index 20f80aefec..5dd6e8fe97 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
@@ -14,7 +14,8 @@
 // C code is at end of this file.
 
 #include <arm_neon.h>
-#include <assert.h>
+
+#include "webrtc/base/checks.h"
 
 void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
     int16_t* data_ch1,  // Input and output in channel 1, in Q0
@@ -24,7 +25,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
     const int length,  // Length of the data buffers
     int32_t* filter_state_ch1,  // Filter state for channel 1, in Q16
     int32_t* filter_state_ch2) {  // Filter state for channel 2, in Q16
-  assert(length % 2 == 0);
+  RTC_DCHECK_EQ(0, length % 2);
   int n = 0;
   int16x4_t factorv;
   int16x4_t datav;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c
index 21e49830a1..2e666d6397 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.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_coding/codecs/isac/fix/source/codec.h"
 
 // Autocorrelation function in fixed point.
@@ -27,8 +26,8 @@ int WebRtcIsacfix_AutocorrC(int32_t* __restrict r,
   int64_t prod = 0;
 
   // The ARM assembly code assumptoins.
-  assert(N % 4 == 0);
-  assert(N >= 8);
+  RTC_DCHECK_EQ(0, N % 4);
+  RTC_DCHECK_GE(N, 8);
 
   // Calculate r[0].
   for (i = 0; i < N; i++) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
index eff8dae17c..df4ef6fa76 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
@@ -9,8 +9,8 @@
  */
 
 #include <arm_neon.h>
-#include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 
 // Autocorrelation function in fixed point.
@@ -26,8 +26,8 @@ int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r,
   int64_t prod = 0;
   int64_t prod_tail = 0;
 
-  assert(n % 4 == 0);
-  assert(n >= 8);
+  RTC_DCHECK_EQ(0, n % 4);
+  RTC_DCHECK_GE(n, 8);
 
   // Calculate r[0].
   int16x4_t x0_v;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index e7905ae81f..39ae8beeff 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -17,9 +17,9 @@
 
 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 
-#include <assert.h>
 #include <stdlib.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
@@ -1113,8 +1113,8 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
 void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst,
                                            int bottleneck_bits_per_second) {
   ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
-  assert(bottleneck_bits_per_second >= 10000 &&
-         bottleneck_bits_per_second <= 32000);
+  RTC_DCHECK_GE(bottleneck_bits_per_second, 10000);
+  RTC_DCHECK_LE(bottleneck_bits_per_second, 32000);
   inst->bwestimator_obj.sendBwAvg = ((uint32_t)bottleneck_bits_per_second) << 7;
 }
 
@@ -1539,13 +1539,13 @@ void WebRtcIsacfix_version(char *version)
 void WebRtcIsacfix_GetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
                                     IsacBandwidthInfo* bwinfo) {
   ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
-  assert(inst->initflag & 1);  // Decoder initialized.
+  RTC_DCHECK_NE(0, inst->initflag & 1);  // Decoder initialized.
   WebRtcIsacfixBw_GetBandwidthInfo(&inst->bwestimator_obj, bwinfo);
 }
 
 void WebRtcIsacfix_SetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
                                     const IsacBandwidthInfo* bwinfo) {
   ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
-  assert(inst->initflag & 2);  // Encoder initialized.
+  RTC_DCHECK_NE(0, inst->initflag & 2);  // Encoder initialized.
   WebRtcIsacfixBw_SetBandwidthInfo(&inst->bwestimator_obj, bwinfo);
 }
diff --git a/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi b/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
index 744885d396..e313a7edb1 100644
--- a/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
+++ b/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
@@ -23,6 +23,7 @@
         '<(webrtc_root)',
       ],
       'sources': [
+        'empty.cc',  # force build system to use C++ linker
         './main/test/simpleKenny.c',
         './main/util/utility.c',
       ],
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index 51da3f7c76..f3d9e1b774 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@@ -19,8 +19,8 @@
 #include "bandwidth_estimator.h"
 #include "settings.h"
 #include "isac.h"
+#include "webrtc/base/checks.h"
 
-#include <assert.h>
 #include <math.h>
 #include <string.h>
 
@@ -159,7 +159,7 @@ int16_t WebRtcIsac_UpdateBandwidthEstimator(
   int immediate_set = 0;
   int num_pkts_expected;
 
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
 
   // We have to adjust the header-rate if the first packet has a
   // frame-size different than the initialized value.
@@ -514,7 +514,7 @@ int16_t WebRtcIsac_UpdateUplinkBwImpl(
     int16_t               index,
     enum IsacSamplingRate encoderSamplingFreq)
 {
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
 
   if((index < 0) || (index > 23))
   {
@@ -572,7 +572,7 @@ int16_t WebRtcIsac_UpdateUplinkJitter(
     BwEstimatorstr*              bwest_str,
     int32_t                  index)
 {
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
 
   if((index < 0) || (index > 23))
   {
@@ -711,7 +711,7 @@ int32_t WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
   float   jitter_sign;
   float   bw_adjust;
 
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
 
   /* create a value between -1.0 and 1.0 indicating "average sign" of jitter */
   jitter_sign = bwest_str->rec_jitter_short_term /
@@ -741,7 +741,7 @@ WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
 {
   int32_t rec_max_delay;
 
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
 
   rec_max_delay = (int32_t)(bwest_str->rec_max_delay);
 
@@ -759,7 +759,7 @@ WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
 
 /* Clamp val to the closed interval [min,max]. */
 static int32_t clamp(int32_t val, int32_t min, int32_t max) {
-  assert(min <= max);
+  RTC_DCHECK_LE(min, max);
   return val < min ? min : (val > max ? max : val);
 }
 
@@ -778,7 +778,7 @@ int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str) {
 void WebRtcIsacBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
                                    enum IsacSamplingRate decoder_sample_rate_hz,
                                    IsacBandwidthInfo* bwinfo) {
-  assert(!bwest_str->external_bw_info.in_use);
+  RTC_DCHECK(!bwest_str->external_bw_info.in_use);
   bwinfo->in_use = 1;
   bwinfo->send_bw_avg = WebRtcIsac_GetUplinkBandwidth(bwest_str);
   bwinfo->send_max_delay_avg = WebRtcIsac_GetUplinkMaxDelay(bwest_str);
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index 875e7ac521..e59f16f2c1 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -17,12 +17,12 @@
 
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 
-#include <assert.h>
 #include <math.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h"
@@ -1539,8 +1539,8 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
 void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
                                         int bottleneck_bits_per_second) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-  assert(bottleneck_bits_per_second >= 10000 &&
-         bottleneck_bits_per_second <= 32000);
+  RTC_DCHECK_GE(bottleneck_bits_per_second, 10000);
+  RTC_DCHECK_LE(bottleneck_bits_per_second, 32000);
   instISAC->bwestimator_obj.send_bw_avg = (float)bottleneck_bits_per_second;
 }
 
@@ -2341,7 +2341,7 @@ uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
 void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst,
                                  IsacBandwidthInfo* bwinfo) {
   ISACMainStruct* instISAC = (ISACMainStruct*)inst;
-  assert(instISAC->initFlag & BIT_MASK_DEC_INIT);
+  RTC_DCHECK_NE(0, instISAC->initFlag & BIT_MASK_DEC_INIT);
   WebRtcIsacBw_GetBandwidthInfo(&instISAC->bwestimator_obj,
                                 instISAC->decoderSamplingRateKHz, bwinfo);
 }
@@ -2349,15 +2349,15 @@ void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst,
 void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
                                  const IsacBandwidthInfo* bwinfo) {
   ISACMainStruct* instISAC = (ISACMainStruct*)inst;
-  assert(instISAC->initFlag & BIT_MASK_ENC_INIT);
+  RTC_DCHECK_NE(0, instISAC->initFlag & BIT_MASK_ENC_INIT);
   WebRtcIsacBw_SetBandwidthInfo(&instISAC->bwestimator_obj, bwinfo);
 }
 
 void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst,
                                         int sample_rate_hz) {
   ISACMainStruct* instISAC = (ISACMainStruct*)inst;
-  assert(instISAC->initFlag & BIT_MASK_DEC_INIT);
-  assert(!(instISAC->initFlag & BIT_MASK_ENC_INIT));
-  assert(sample_rate_hz == 16000 || sample_rate_hz == 32000);
+  RTC_DCHECK_NE(0, instISAC->initFlag & BIT_MASK_DEC_INIT);
+  RTC_DCHECK(!(instISAC->initFlag & BIT_MASK_ENC_INIT));
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000);
   instISAC->encoderSamplingRateKHz = sample_rate_hz / 1000;
 }
diff --git a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
index 2450e05b76..d2155648d6 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
diff --git a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
index 231a2044a3..dd24845cf2 100644
--- a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
@@ -14,12 +14,12 @@
 
 #include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
 
-#include <assert.h>
 #include <string.h>
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
 
 // To generate the gaintable, copy&paste the following lines to a Matlab window:
@@ -109,7 +109,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,       // Q16
   diffGain =
       WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
   if (diffGain < 0 || diffGain >= kGenFuncTableSize) {
-    assert(0);
+    RTC_DCHECK(0);
     return -1;
   }
 
@@ -268,7 +268,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc* stt, int16_t agcMode) {
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
                                      const int16_t* in_far,
                                      size_t nrSamples) {
-  assert(stt != NULL);
+  RTC_DCHECK(stt);
   // VAD for far end
   WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
 
diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
index 5ce64cee29..76589c5fe9 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;
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 94b6449776..c58fc39bac 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -10,11 +10,11 @@
 
 #include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 
-#include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
@@ -344,8 +344,8 @@ static void NoiseEstimationC(NoiseSuppressionFixedC* inst,
   size_t i, s, offset;
 
   tabind = inst->stages - inst->normData;
-  assert(tabind < 9);
-  assert(tabind > -9);
+  RTC_DCHECK_LT(tabind, 9);
+  RTC_DCHECK_GT(tabind, -9);
   if (tabind < 0) {
     logval = -WebRtcNsx_kLogTable[-tabind];
   } else {
@@ -362,7 +362,7 @@ static void NoiseEstimationC(NoiseSuppressionFixedC* inst,
       frac = (int16_t)((((uint32_t)magn[i] << zeros)
                               & 0x7FFFFFFF) >> 23);
       // log2(magn(i))
-      assert(frac < 256);
+      RTC_DCHECK_LT(frac, 256);
       log2 = (int16_t)(((31 - zeros) << 8)
                              + WebRtcNsx_kLogTableFrac[frac]);
       // log2(magn(i))*log(2)
@@ -380,7 +380,7 @@ static void NoiseEstimationC(NoiseSuppressionFixedC* inst,
 
     // Get counter values from state
     counter = inst->noiseEstCounter[s];
-    assert(counter < 201);
+    RTC_DCHECK_LT(counter, 201);
     countDiv = WebRtcNsx_kCounterDiv[counter];
     countProd = (int16_t)(counter * countDiv);
 
@@ -543,7 +543,7 @@ static void NormalizeRealBufferC(NoiseSuppressionFixedC* inst,
                                  const int16_t* in,
                                  int16_t* out) {
   size_t i = 0;
-  assert(inst->normData >= 0);
+  RTC_DCHECK_GE(inst->normData, 0);
   for (i = 0; i < inst->anaLen; ++i) {
     out[i] = in[i] << inst->normData;  // Q(normData)
   }
@@ -594,8 +594,8 @@ void WebRtcNsx_CalcParametricNoiseEstimate(NoiseSuppressionFixedC* inst,
 
   // Use pink noise estimate
   // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
-  assert(freq_index >= 0);
-  assert(freq_index < 129);
+  RTC_DCHECK_GE(freq_index, 0);
+  RTC_DCHECK_LT(freq_index, 129);
   tmp32no2 = (pink_noise_exp_avg * kLogIndex[freq_index]) >> 15;  // Q11
   tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
 
@@ -1038,7 +1038,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NoiseSuppressionFixedC* inst,
       frac = (int16_t)(((uint32_t)((uint32_t)(magn[i]) << zeros)
                               & 0x7FFFFFFF) >> 23);
       // log2(magn(i))
-      assert(frac < 256);
+      RTC_DCHECK_LT(frac, 256);
       tmpU32 = (uint32_t)(((31 - zeros) << 8)
                                 + WebRtcNsx_kLogTableFrac[frac]); // Q8
       avgSpectralFlatnessNum += tmpU32; // Q8
@@ -1053,7 +1053,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NoiseSuppressionFixedC* inst,
   zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
   frac = (int16_t)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
   // log2(avgSpectralFlatnessDen)
-  assert(frac < 256);
+  RTC_DCHECK_LT(frac, 256);
   tmp32 = (int32_t)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]); // Q8
   logCurSpectralFlatness = (int32_t)avgSpectralFlatnessNum;
   logCurSpectralFlatness += ((int32_t)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
@@ -1286,7 +1286,7 @@ void WebRtcNsx_DataAnalysis(NoiseSuppressionFixedC* inst,
       frac = (int16_t)((((uint32_t)magnU16[inst->anaLen2] << zeros) &
                               0x7FFFFFFF) >> 23); // Q8
       // log2(magnU16(i)) in Q8
-      assert(frac < 256);
+      RTC_DCHECK_LT(frac, 256);
       log2 = (int16_t)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]);
     }
 
@@ -1320,7 +1320,7 @@ void WebRtcNsx_DataAnalysis(NoiseSuppressionFixedC* inst,
           frac = (int16_t)((((uint32_t)magnU16[i] << zeros) &
                                   0x7FFFFFFF) >> 23);
           // log2(magnU16(i)) in Q8
-          assert(frac < 256);
+          RTC_DCHECK_LT(frac, 256);
           log2 = (int16_t)(((31 - zeros) << 8)
                                  + WebRtcNsx_kLogTableFrac[frac]);
         }
@@ -1347,14 +1347,14 @@ void WebRtcNsx_DataAnalysis(NoiseSuppressionFixedC* inst,
     // Shift to same Q-domain as whiteNoiseLevel
     tmpU32no1 >>= right_shifts_in_magnU16;
     // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
-    assert(END_STARTUP_SHORT < 128);
+    RTC_DCHECK_LT(END_STARTUP_SHORT, 128);
     inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
 
     // Estimate Pink noise parameters
     // Denominator used in both parameter estimates.
     // The value is only dependent on the size of the frequency band (kStartBand)
     // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
-    assert(kStartBand < 66);
+    RTC_DCHECK_LT(kStartBand, 66);
     matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
     sum_log_i = kSumLogIndex[kStartBand]; // Q5
     sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
@@ -1469,13 +1469,13 @@ void WebRtcNsx_DataSynthesis(NoiseSuppressionFixedC* inst, short* outFrame) {
       inst->energyIn >>= 8 + scaleEnergyOut - inst->scaleEnergyIn;
     }
 
-    assert(inst->energyIn > 0);
+    RTC_DCHECK_GT(inst->energyIn, 0);
     energyRatio = (energyOut + inst->energyIn / 2) / inst->energyIn;  // Q8
     // Limit the ratio to [0, 1] in Q8, i.e., [0, 256]
     energyRatio = WEBRTC_SPL_SAT(256, energyRatio, 0);
 
     // all done in lookup tables now
-    assert(energyRatio < 257);
+    RTC_DCHECK_LT(energyRatio, 257);
     gainFactor1 = kFactor1Table[energyRatio]; // Q8
     gainFactor2 = inst->factor2Table[energyRatio]; // Q8
 
@@ -1534,24 +1534,24 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
   int q_domain_to_use = 0;
 
   // Code for ARMv7-Neon platform assumes the following:
-  assert(inst->anaLen > 0);
-  assert(inst->anaLen2 > 0);
-  assert(inst->anaLen % 16 == 0);
-  assert(inst->anaLen2 % 8 == 0);
-  assert(inst->blockLen10ms > 0);
-  assert(inst->blockLen10ms % 16 == 0);
-  assert(inst->magnLen == inst->anaLen2 + 1);
+  RTC_DCHECK_GT(inst->anaLen, 0);
+  RTC_DCHECK_GT(inst->anaLen2, 0);
+  RTC_DCHECK_EQ(0, inst->anaLen % 16);
+  RTC_DCHECK_EQ(0, inst->anaLen2 % 8);
+  RTC_DCHECK_GT(inst->blockLen10ms, 0);
+  RTC_DCHECK_EQ(0, inst->blockLen10ms % 16);
+  RTC_DCHECK_EQ(inst->magnLen, inst->anaLen2 + 1);
 
 #ifdef NS_FILEDEBUG
   if (fwrite(spframe, sizeof(short),
              inst->blockLen10ms, inst->infile) != inst->blockLen10ms) {
-    assert(false);
+    RTC_DCHECK(false);
   }
 #endif
 
   // Check that initialization has been done
-  assert(inst->initFlag == 1);
-  assert((num_bands - 1) <= NUM_HIGH_BANDS_MAX);
+  RTC_DCHECK_EQ(1, inst->initFlag);
+  RTC_DCHECK_LE(num_bands - 1, NUM_HIGH_BANDS_MAX);
 
   const short* const* speechFrameHB = NULL;
   short* const* outFrameHB = NULL;
@@ -1989,7 +1989,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
 
     //gain filter
     tmpU32no1 = inst->overdrive + ((priorSnr + 8192) >> 14);  // Q8
-    assert(inst->overdrive > 0);
+    RTC_DCHECK_GT(inst->overdrive, 0);
     tmpU16no1 = (priorSnr + tmpU32no1 / 2) / tmpU32no1;  // Q14
     inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
 
@@ -2025,7 +2025,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
 #ifdef NS_FILEDEBUG
   if (fwrite(outframe, sizeof(short),
              inst->blockLen10ms, inst->outfile) != inst->blockLen10ms) {
-    assert(false);
+    RTC_DCHECK(false);
   }
 #endif
 
@@ -2052,7 +2052,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
       tmpU16no1 += nonSpeechProbFinal[i]; // Q8
       tmpU32no1 += (uint32_t)(inst->noiseSupFilter[i]); // Q14
     }
-    assert(inst->stages >= 7);
+    RTC_DCHECK_GE(inst->stages, 7);
     avgProbSpeechHB = (4096 - (tmpU16no1 >> (inst->stages - 7)));  // Q12
     avgFilterGainHB = (int16_t)(tmpU32no1 >> (inst->stages - 3));  // Q14
 
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index 213320d38c..abfb2c9e3e 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) {
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index 3922308c7c..2baf7dfa9e 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <assert.h>
 #include <string.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"
 
@@ -184,7 +184,7 @@ 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) {
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_neon.c b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
index 516dd095cc..fb1b32350d 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_neon.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
@@ -11,7 +11,8 @@
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 
 #include <arm_neon.h>
-#include <assert.h>
+
+#include "webrtc/base/checks.h"
 
 // Constants to compensate for shifting signal log(2^shifts).
 const int16_t WebRtcNsx_kLogTable[9] = {
@@ -144,8 +145,8 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
   size_t i, s, offset;
 
   tabind = inst->stages - inst->normData;
-  assert(tabind < 9);
-  assert(tabind > -9);
+  RTC_DCHECK_LT(tabind, 9);
+  RTC_DCHECK_GT(tabind, -9);
   if (tabind < 0) {
     logval = -WebRtcNsx_kLogTable[-tabind];
   } else {
@@ -163,7 +164,7 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
       zeros = WebRtcSpl_NormU32((uint32_t)magn[i]);
       frac = (int16_t)((((uint32_t)magn[i] << zeros)
                         & 0x7FFFFFFF) >> 23);
-      assert(frac < 256);
+      RTC_DCHECK_LT(frac, 256);
       // log2(magn(i))
       log2 = (int16_t)(((31 - zeros) << 8)
                        + WebRtcNsx_kLogTableFrac[frac]);
@@ -190,7 +191,7 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
 
     // Get counter values from state
     counter = inst->noiseEstCounter[s];
-    assert(counter < 201);
+    RTC_DCHECK_LT(counter, 201);
     countDiv = WebRtcNsx_kCounterDiv[counter];
     countProd = (int16_t)(counter * countDiv);
 
@@ -354,8 +355,8 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
 // Filter the data in the frequency domain, and create spectrum.
 void WebRtcNsx_PrepareSpectrumNeon(NoiseSuppressionFixedC* inst,
                                    int16_t* freq_buf) {
-  assert(inst->magnLen % 8 == 1);
-  assert(inst->anaLen2 % 16 == 0);
+  RTC_DCHECK_EQ(1, inst->magnLen % 8);
+  RTC_DCHECK_EQ(0, inst->anaLen2 % 16);
 
   // (1) Filtering.
 
@@ -445,8 +446,8 @@ void WebRtcNsx_PrepareSpectrumNeon(NoiseSuppressionFixedC* inst,
 void WebRtcNsx_SynthesisUpdateNeon(NoiseSuppressionFixedC* inst,
                                    int16_t* out_frame,
                                    int16_t gain_factor) {
-  assert(inst->anaLen % 16 == 0);
-  assert(inst->blockLen10ms % 16 == 0);
+  RTC_DCHECK_EQ(0, inst->anaLen % 16);
+  RTC_DCHECK_EQ(0, inst->blockLen10ms % 16);
 
   int16_t* preal_start = inst->real;
   const int16_t* pwindow = inst->window;
@@ -537,8 +538,8 @@ void WebRtcNsx_SynthesisUpdateNeon(NoiseSuppressionFixedC* inst,
 void WebRtcNsx_AnalysisUpdateNeon(NoiseSuppressionFixedC* inst,
                                   int16_t* out,
                                   int16_t* new_speech) {
-  assert(inst->blockLen10ms % 16 == 0);
-  assert(inst->anaLen % 16 == 0);
+  RTC_DCHECK_EQ(0, inst->blockLen10ms % 16);
+  RTC_DCHECK_EQ(0, inst->anaLen % 16);
 
   // For lower band update analysis buffer.
   // memcpy(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
diff --git a/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c b/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
index 0b05e224eb..cd1ff72d17 100644
--- a/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
+++ b/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
@@ -10,7 +10,6 @@
 
 #include "webrtc/system_wrappers/source/data_log_c_helpers_unittest.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>