diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index a9cf3842f..56bbbe66d 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -46,8 +46,6 @@
     ((uint32_t) ((uint32_t)(a) * (uint16_t)(b)))
 #define WEBRTC_SPL_MUL_16_U16(a, b) \
     ((int32_t)(int16_t)(a) * (uint16_t)(b))
-#define WEBRTC_SPL_DIV(a, b) \
-    ((int32_t) ((int32_t)(a) / (int32_t)(b)))
 
 #ifndef WEBRTC_ARCH_ARM_V7
 // For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 5d07f16ed..6a70a02ef 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -47,7 +47,6 @@ TEST_F(SplTest, MacroTest) {
 
     a = b;
     b = -3;
-    EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
 
     EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
     EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index e57416580..4efdecc9a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -282,11 +282,11 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
     if (inSqrt < 0)
       inSqrt=-inSqrt;
 
-    newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+    newRes = (inSqrt / res + res) >> 1;
     do
     {
       res = newRes;
-      newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+      newRes = (inSqrt / res + res) >> 1;
     } while (newRes != res && i-- > 0);
 
     tmpARSpecQ8 = (uint16_t)newRes;
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 2ac15350c..d28a6f705 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
@@ -296,9 +296,9 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
             bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((int32_t)bweStr->recBwInv, 13);
 
           } else {
-            /* recBwInv = 1 / (INIT_BN_EST + INIT_HDR_RATE) in Q26 (Q30??)*/
-            bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
-                                               WEBRTC_SPL_LSHIFT_W32(((int32_t)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
+            static const uint32_t kInitRate = INIT_BN_EST + INIT_HDR_RATE;
+            /* recBwInv = 1 / kInitRate  in Q26 (Q30??)*/
+            bweStr->recBwInv = (1073741824 + kInitRate / 2) / kInitRate;
           }
 
           /* reset time-since-update counter */
@@ -854,13 +854,14 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
   } else {
     /* handle burst */
     if (State->BurstCounter) {
-      if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
+      if (State->StillBuffered <
+          (((512 - 512 / BURST_LEN) * DelayBuildUp) >> 9)) {
         /* max bps derived from BottleNeck and DelayBuildUp values */
-        inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples));
+        inv_Q12 = 4096 / (BURST_LEN * FrameSamples);
         MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp, inv_Q12), 3)), BottleNeck);
       } else {
         /* max bps derived from StillBuffered and DelayBuildUp values */
-        inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples);
+        inv_Q12 = 4096 / FrameSamples;
         if (DelayBuildUp > State->StillBuffered) {
           MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp - State->StillBuffered, inv_Q12), 3)), BottleNeck);
         } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
@@ -895,10 +896,10 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
 
   /* keep track of when bottle neck was last exceeded by at least 1% */
   //517/512 ~ 1.01
-  if (WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, FS8), FrameSamples) > (WEBRTC_SPL_MUL(517, BottleNeck) >> 9)) {
+  if ((StreamSize * (int32_t)FS8) / FrameSamples > (517 * BottleNeck) >> 9) {
     if (State->PrevExceed) {
       /* bottle_neck exceded twice in a row, decrease ExceedAgo */
-      State->ExceedAgo -= WEBRTC_SPL_DIV(BURST_INTERVAL, BURST_LEN - 1);
+      State->ExceedAgo -= BURST_INTERVAL / (BURST_LEN - 1);
       if (State->ExceedAgo < 0) {
         State->ExceedAgo = 0;
       }
@@ -922,7 +923,7 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
 
 
   /* Update buffer delay */
-  TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck);    /* ms */
+  TransmissionTime = (StreamSize * 8000) / BottleNeck;  /* ms */
   State->StillBuffered += TransmissionTime;
   State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);  //>>4 =  SAMPLES_PER_MSEC        /* ms */
   if (State->StillBuffered < 0) {
@@ -945,13 +946,12 @@ void WebRtcIsacfix_UpdateRateModel(RateModel *State,
                                    const int16_t FrameSamples,            /* samples per frame */
                                    const int16_t BottleNeck)        /* bottle neck rate; excl headers (bps) */
 {
-  int16_t TransmissionTime;
+  const int16_t TransmissionTime = (StreamSize * 8000) / BottleNeck;  /* ms */
 
   /* avoid the initial "high-rate" burst */
   State->InitCounter = 0;
 
   /* Update buffer delay */
-  TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck);    /* ms */
   State->StillBuffered += TransmissionTime;
   State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4);            /* ms */
   if (State->StillBuffered < 0) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index 0f5c81938..263f88a4f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -59,8 +59,8 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
 
 
   int16_t frame_nb; /* counter */
-  int16_t frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
-  int16_t processed_samples;
+  int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */
+  static const int16_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */
 
   /* PLC */
   int16_t overlapWin[ 240 ];
@@ -76,14 +76,14 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
   if (err<0)  // error check
     return err;
 
-  frame_mode = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
-  processed_samples = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
+  frame_mode = *current_framesamples / MAX_FRAMESAMPLES;  /* 0, or 1 */
 
   err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
   if (err<0)  // error check
     return err;
 
-  /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames bundled together (60ms) */
+  /* one loop if it's one frame (30ms), two loops if two frames bundled together
+   * (60ms) */
   for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
 
     /* decode & dequantize pitch parameters */
@@ -210,7 +210,10 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
       Vector_Word16_2[k] = tmp_2;
     }
 
-    WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1, Vector_Word16_2, signal_out16 + frame_nb * processed_samples, &ISACdec_obj->postfiltbankstr_obj);
+    WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1,
+                                    Vector_Word16_2,
+                                    signal_out16 + frame_nb * kProcessedSamples,
+                                    &ISACdec_obj->postfiltbankstr_obj);
 
   }
   return len;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 435f57232..27d1c1fc1 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -350,11 +350,11 @@ static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
     if(in_sqrt<0)
       in_sqrt=-in_sqrt;
 
-    newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+    newRes = (in_sqrt / res + res) >> 1;
     do
     {
       res = newRes;
-      newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+      newRes = (in_sqrt / res + res) >> 1;
     } while (newRes != res && i-- > 0);
 
     CurveQ8[k] = (int16_t)newRes;
@@ -368,11 +368,11 @@ static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
     if(in_sqrt<0)
       in_sqrt=-in_sqrt;
 
-    newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+    newRes = (in_sqrt / res + res) >> 1;
     do
     {
       res = newRes;
-      newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+      newRes = (in_sqrt / res + res) >> 1;
     } while (newRes != res && i-- > 0);
 
     CurveQ8[k] = (int16_t)newRes;
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/analog_agc.c
index 4f110cc20..0376dae51 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -208,7 +208,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
         tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
         tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
         tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
-        targetGainIdx = (uint16_t)WEBRTC_SPL_DIV(tmp32, tmp16);
+        targetGainIdx = tmp32 / tmp16;
         assert(targetGainIdx < GAIN_TBL_LEN);
 
         /* Increment through the table towards the target gain.
@@ -1078,8 +1078,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                     tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
                     if (stt->maxInit != stt->minLevel)
                     {
-                        volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
-                                                              (stt->maxInit - stt->minLevel));
+                        volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
                     }
 
                     /* Find correct curve */
@@ -1138,8 +1137,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                     tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
                     if (stt->maxInit != stt->minLevel)
                     {
-                        volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
-                                                              (stt->maxInit - stt->minLevel));
+                        volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
                     }
 
                     /* Find correct curve */
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index 7b515a57d..b15b6e39d 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -210,7 +210,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
         {
             numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
         }
-        y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
+        y32 = numFIX / tmp32no1;  // in Q14
         if (limiterEnable && (i < limiterIdx))
         {
             tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 41244d4c9..5a88c12fb 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -1493,8 +1493,7 @@ void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) {
     }
 
     assert(inst->energyIn > 0);
-    energyRatio = (int16_t)WEBRTC_SPL_DIV(energyOut
-        + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
+    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);
 
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index de92441ac..b50d4f858 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -258,8 +258,8 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
           tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)inst->priorNonSpeechProb,
                                            8); // Q22
 
-          nonSpeechProbFinal[i] = (uint16_t)WEBRTC_SPL_DIV(tmp32no1,
-              (int32_t)inst->priorNonSpeechProb + invLrtFX); // Q8
+          nonSpeechProbFinal[i] = tmp32no1 /
+              (inst->priorNonSpeechProb + invLrtFX);  // Q8
         }
       }
     }