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 67b18331b3..16befba707 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
@@ -1005,13 +1005,17 @@ int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples)
   /* find new SNR value */
   //consider BottleNeck to be in Q10 ( * 1 in Q10)
   switch(framesamples) {
+  // TODO(bjornv): The comments below confuses me. I don't know if there is a
+  // difference between frame lengths (in which case the implementation is
+  // wrong), or if it is frame length independent in which case we should
+  // correct the comment and simplify the implementation.
     case 480:
       /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10);
       break;
     case 960:
       /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10);
       break;
     default:
       s2nr = -1; /* Error */
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 714a897cdb..5e095da683 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -130,14 +130,15 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
       ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX;    /* DECAY_RATE is in Q15 */
       ISACdec_obj->plcstr_obj.pitchCycles = 0;
 
-      PitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
+      PitchGains_Q12[0] = (int16_t)(PitchGains_Q12[0] * 700 >> 10);
 
       /* ---- Add-overlap ---- */
       WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
       for( k = 0; k < RECOVERY_OVERLAP; k++ )
         Vector_Word16_1[k] = WebRtcSpl_AddSatW16(
-            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
-            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
+            (int16_t)(ISACdec_obj->plcstr_obj.overlapLP[k] *
+                overlapWin[RECOVERY_OVERLAP - k - 1] >> 14),
+            (int16_t)(Vector_Word16_1[k] * overlapWin[k] >> 14));
 
 
 
@@ -176,7 +177,7 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
 
     /* reduce gain to compensate for pitch enhancer */
     /* gain = 1.0f - 0.45f * AvgPitchGain; */
-    tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
+    tmp32a = AvgPitchGain_Q12 * 29;  // Q18
     gainQ13 = (int16_t)((262144 - tmp32a) >> 5);  // Q18 -> Q13.
 
     for (k = 0; k < FRAMESAMPLES/2; k++)
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
index 4243ea3874..75ae542be0 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
@@ -72,7 +72,7 @@ static int16_t plc_filterma_Fast(
     o >>= rshift;
 
     /* decay the output signal; this is specific to plc */
-    *Out++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (int16_t)o, decay, 15); // ((o + (int32_t)2048) >> 12);
+    *Out++ = (int16_t)((int16_t)o * decay >> 15);
 
     /* change the decay */
     decay -= reduceDecay;
@@ -139,7 +139,7 @@ static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
          x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999     */
 
       /* we are sure that x is in the range of int16_t            */
-      x = (int16_t)(in * 15 + WEBRTC_SPL_MUL_16_16_RSFT(in, 983, 12));
+      x = (int16_t)(in * 15 + (in * 983 >> 12));
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
       *B = (int16_t)((x * x + 0x00010000) >> 17);
@@ -157,7 +157,7 @@ static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
     {
       /* This is a mirror case of the above */
       in = 4300 - in;
-      x = (int16_t)(in * 15 + WEBRTC_SPL_MUL_16_16_RSFT(in, 983, 12));
+      x = (int16_t)(in * 15 + (in * 983 >> 12));
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
       *A = (int16_t)((x * x + 0x00010000) >> 17);
@@ -220,7 +220,7 @@ static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t l
       else
       {
         diff = in[ j + 1 ] - in[ j ];
-        out[ i ] = in[ j ] + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
+        out[i] = in[j] + (int16_t)(diff * relativePos >> RESAMP_RES_BIT);
       }
     }
   }
@@ -449,12 +449,9 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
   pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
       ((ISACdec_obj->plcstr_obj).stretchLag<<7);
   pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
-  pitchGains_Q12[2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[3], 1010, 10 );
-  pitchGains_Q12[1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[2], 1010, 10 );
-  pitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[1], 1010, 10 );
+  pitchGains_Q12[2] = (int16_t)(pitchGains_Q12[3] * 1010 >> 10);
+  pitchGains_Q12[1] = (int16_t)(pitchGains_Q12[2] * 1010 >> 10);
+  pitchGains_Q12[0] = (int16_t)(pitchGains_Q12[1] * 1010 >> 10);
 
 
   /* most of the time either B or A are zero so seperating */
@@ -527,9 +524,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
       for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
       {
         /* --- Lowpass                                               */
-        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
-            (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+        pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] *
+            ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
 
         /* --- Highpass                                              */
         pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
@@ -626,9 +622,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
         noise1 = (ISACdec_obj->plcstr_obj.seed >> 10) - 16;
 
-        nLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            (int16_t)((noise1)*(ISACdec_obj->plcstr_obj).std),
-            (ISACdec_obj->plcstr_obj).decayCoeffNoise, 15 );
+        nLP = (int16_t)((int16_t)(noise1 * ISACdec_obj->plcstr_obj.std) *
+            ISACdec_obj->plcstr_obj.decayCoeffNoise >> 15);
 
         /* --- Highpass                                              */
         (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
@@ -646,9 +641,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
         /* ------ Periodic Vector ---                                */
         /* --- Lowpass                                               */
-        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
-            (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+        pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] *
+            ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
 
         /* --- Highpass                                              */
         pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
@@ -665,13 +659,11 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
         }
 
         /* ------ Weighting the noisy and periodic vectors -------   */
-        wNoisyLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
-            (ISACdec_obj->plcstr_obj).A, nLP, 15 ) );
+        wNoisyLP = (int16_t)(ISACdec_obj->plcstr_obj.A * nLP >> 15);
         wNoisyHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).A, (nHP) ) );
 
-        wPriodicLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
-            (ISACdec_obj->plcstr_obj).B, pLP, 15));
+        wPriodicLP = (int16_t)(ISACdec_obj->plcstr_obj.B * pLP >> 15);
         wPriodicHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).B, pHP));
 
@@ -752,9 +744,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
   for( i = 0; i < RECOVERY_OVERLAP; i++ )
   {
-    (ISACdec_obj->plcstr_obj).overlapLP[i] = (int16_t)(
-        WEBRTC_SPL_MUL_16_16_RSFT(stretchPitchLP[k],
-                                  (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15) );
+    ISACdec_obj->plcstr_obj.overlapLP[i] = (int16_t)(
+        stretchPitchLP[k] * ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
     k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
   }
 
@@ -767,8 +758,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
   /* reduce gain to compensate for pitch enhancer */
   /* gain = 1.0f - 0.45f * AvgPitchGain; */
-  tmp32a = WEBRTC_SPL_MUL_16_16_RSFT((ISACdec_obj->plcstr_obj).AvgPitchGain_Q12,
-                                     29, 0); // Q18
+  tmp32a = ISACdec_obj->plcstr_obj.AvgPitchGain_Q12 * 29;  // Q18
   tmp32b = 262144 - tmp32a;  // Q18
   gainQ13 = (int16_t) (tmp32b >> 5); // Q13
 
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 d3e7107b6e..1a6372a664 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -115,8 +115,9 @@ int WebRtcIsacfix_EncodeImpl(int16_t      *in,
 
     // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
     // 901/1024 is 0.87988281250000
-    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
-                                             ISACenc_obj->current_framesamples);
+    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr(
+        (int16_t)(ISACenc_obj->BottleNeck * 901 >> 10),
+        ISACenc_obj->current_framesamples);
 
     /* encode frame length */
     status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj);
@@ -352,8 +353,8 @@ int WebRtcIsacfix_EncodeImpl(int16_t      *in,
     // scale FFT coefficients to reduce the bit-rate
     for(k = 0; k < FRAMESAMPLES_HALF; k++)
     {
-      LP16a[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
-      LPandHP[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
+      LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14);
+      LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14);
     }
 
     // Save data for multiple packets memory
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 c150e60692..39653789ef 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
@@ -110,7 +110,7 @@ static int16_t CalcLogN(int32_t arg) {
   zeros=WebRtcSpl_NormU32(arg);
   frac = (int16_t)((uint32_t)((arg << zeros) & 0x7FFFFFFF) >> 23);
   log2 = (int16_t)(((31 - zeros) << 8) + frac);  // log2(x) in Q8
-  logN=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
+  logN = (int16_t)(log2 * 22713 >> 15);  // log(2) = 0.693147 = 22713 in Q15
   logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x.
 
   return logN;
@@ -129,13 +129,12 @@ static int16_t CalcLogN(int32_t arg) {
 */
 
 static int32_t CalcExpN(int16_t x) {
-  int16_t ax, axINT, axFRAC;
+  int16_t axINT, axFRAC;
   int16_t exp16;
   int32_t exp;
+  int16_t ax = (int16_t)(x * 23637 >> 14);  // Q8
 
   if (x>=0) {
-    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
-    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     axINT = ax >> 8;  //Q0
     axFRAC = ax&0x00FF;
     exp16 = 1 << axINT;  // Q0
@@ -143,8 +142,6 @@ static int32_t CalcExpN(int16_t x) {
     exp = exp16 * axFRAC;  // Q0*Q8 = Q8
     exp <<= 9;  // Q17
   } else {
-    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
-    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     ax = -ax;
     axINT = 1 + (ax >> 8);  //Q0
     axFRAC = 0x00FF - (ax&0x00FF);
@@ -679,16 +676,16 @@ static void Rc2LarFix(const int16_t *rcQ15, int32_t *larQ17, int16_t order) {
 
     if (rc<24956) {  //0.7615966 in Q15
       // (Q15*Q13)>>11 = Q17
-      larAbsQ17 = WEBRTC_SPL_MUL_16_16_RSFT(rc, 21512, 11);
+      larAbsQ17 = rc * 21512 >> 11;
     } else if (rc<30000) { //0.91552734375 in Q15
       // Q17 + (Q15*Q12)>>10 = Q17
-      larAbsQ17 = -465024 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 29837, 10);
+      larAbsQ17 = -465024 + (rc * 29837 >> 10);
     } else if (rc<32500) { //0.99182128906250 in Q15
       // Q17 + (Q15*Q10)>>8 = Q17
-      larAbsQ17 = -3324784 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 31863, 8);
+      larAbsQ17 = -3324784 + (rc * 31863 >> 8);
     } else  {
       // Q17 + (Q15*Q5)>>3 = Q17
-      larAbsQ17 = -88546020 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 21973, 3);
+      larAbsQ17 = -88546020 + (rc * 21973 >> 3);
     }
 
     if (rcQ15[k]>0) {
@@ -717,7 +714,7 @@ static void Lar2RcFix(const int32_t *larQ17, int16_t *rcQ15,  int16_t order) {
 
     if (larAbsQ11<4097) { //2.000012018559 in Q11
       // Q11*Q16>>12 = Q15
-      rc = WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24957, 12);
+      rc = larAbsQ11 * 24957 >> 12;
     } else if (larAbsQ11<6393) { //3.121320351712 in Q11
       // (Q11*Q17 + Q13)>>13 = Q15
       rc = (larAbsQ11 * 17993 + 130738688) >> 13;
@@ -995,7 +992,8 @@ int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
       pos = LPC_SHAPE_ORDER * j;
       pos2 = LPC_SHAPE_ORDER * k;
       for (n=0; n<LPC_SHAPE_ORDER; n++) {
-        sumQQ += WEBRTC_SPL_MUL_16_16_RSFT(tmpcoeffs_sQ10[pos], WebRtcIsacfix_kT1ShapeQ15[model][pos2], 7); // (Q10*Q15)>>7 = Q18
+        sumQQ += tmpcoeffs_sQ10[pos] *
+            WebRtcIsacfix_kT1ShapeQ15[model][pos2] >> 7;  // (Q10*Q15)>>7 = Q18
         pos++;
         pos2++;
       }
@@ -1609,7 +1607,7 @@ int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
 
   /* get the approximate arcsine (almost linear)*/
   for (k=0; k<PITCH_SUBFRAMES; k++)
-    SQ15[k] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
+    SQ15[k] = (int16_t)(PitchGains_Q12[k] * 33 >> 2);  // Q15
 
 
   /* find quantization index; only for the first three transform coefficients */
@@ -1618,7 +1616,7 @@ int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
     /*  transform */
     CQ17=0;
     for (j=0; j<PITCH_SUBFRAMES; j++) {
-      CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], SQ15[j],10); // Q17
+      CQ17 += WebRtcIsacfix_kTransform[k][j] * SQ15[j] >> 10;  // Q17
     }
 
     index[k] = (int16_t)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
@@ -1733,14 +1731,14 @@ int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
+    tmp32b = WebRtcIsacfix_kTransform[1][k] * (int16_t)CQ10 >> 10;
     tmp16c = (int16_t)(tmp32b >> 5);
     PitchLags_Q7[k] += tmp16c;
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
+    tmp32b = WebRtcIsacfix_kTransform[3][k] * (int16_t)CQ10 >> 10;
     tmp16c = (int16_t)(tmp32b >> 5);
     PitchLags_Q7[k] += tmp16c;
   }
@@ -1809,7 +1807,7 @@ int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
     /*  transform */
     CQ17=0;
     for (j=0; j<PITCH_SUBFRAMES; j++)
-      CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], PitchLagsQ7[j],2); // Q17
+      CQ17 += WebRtcIsacfix_kTransform[k][j] * PitchLagsQ7[j] >> 2;  // Q17
 
     CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize
 
@@ -1840,14 +1838,14 @@ int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
+    tmp32b = WebRtcIsacfix_kTransform[1][k] * (int16_t)CQ10 >> 10;
     tmp16c = (int16_t)(tmp32b >> 5);  // Q7.
     PitchLagsQ7[k] += tmp16c;
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
+    tmp32b = WebRtcIsacfix_kTransform[3][k] * (int16_t)CQ10 >> 10;
     tmp16c = (int16_t)(tmp32b >> 5);  // Q7.
     PitchLagsQ7[k] += tmp16c;
   }
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
index 5ae951a9e3..97e4ce03a4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
@@ -393,7 +393,7 @@ void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12,
   chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]);
   chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog);
   tmp = chng1+chng2+chng3+chng4;
-  chngQ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
+  chngQ = (int16_t)(tmp * kChngFactor >> 10);  /* Q12 */
   chngQ += 2926; /* + 1.0/1.4 in Q12 */
 
   /* Find average pitch gain */
@@ -403,10 +403,10 @@ void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12,
     pgQ += pitchGains_Q12[k];
   }
 
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18  */
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10  kMulPitchGain = -25 = -200 in Q-3. */
-
+  pg3 = (int16_t)(pgQ * pgQ >> 11);  // pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17
+  pg3 = (int16_t)(pgQ * pg3 >> 13);  /* Q14*Q17>>13 =>Q18  */
+  /* kMulPitchGain = -25 = -200 in Q-3. */
+  pg3 = (int16_t)(pg3 * kMulPitchGain >> 5);  // Q10
   tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
   if (tmp16<0) {
     tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
@@ -580,9 +580,9 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
   snrq=snrQ10;
 
   /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/
-  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
+  tmp16 = (int16_t)(snrq * 172 >> 10);  // Q10
   tmp16b = exp2_Q10_T(tmp16); // Q10
-  snrq = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
+  snrq = (int16_t)(tmp16b * 285 >> 10);  // Q10
 
   /* change quallevel depending on pitch gains and level fluctuations */
   WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14);
@@ -595,12 +595,12 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
   aaQ14 = (int16_t)((22938 * (8192 + (varscaleQ14 >> 1)) + 32768) >> 16);
 
   /* Calculate tmp = (1.0 + aa*aa); in Q12 */
-  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
+  tmp16 = (int16_t)(aaQ14 * aaQ14 >> 15);  // Q14*Q14>>15 = Q13
   tmpQQlo = 4096 + (tmp16 >> 1);  // Q12 + Q13>>1 = Q12.
 
   /* Calculate tmp = (1.0+aa) * (1.0+aa); */
   tmp16 = 8192 + (aaQ14 >> 1);  // 1+a in Q13.
-  tmpQQhi = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
+  tmpQQhi = (int16_t)(tmp16 * tmp16 >> 14);  // Q13*Q13>>14 = Q12
 
   /* replace data in buffer by new look-ahead data */
   for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) {
@@ -613,19 +613,19 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
     for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
       maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2];
       maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2];
-      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+      DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
+      DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
     }
     pos2 = (int16_t)(k * UPDATE / 2);
     for (n = 0; n < UPDATE/2; n++, pos1++) {
       maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2];
       maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++];
-      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+      DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
+      DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
     }
 
     /* Get correlation coefficients */
@@ -868,14 +868,12 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
       /* add hearing threshold and compute the gain */
       /* lo_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
 
-
-      //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
       tmp32a = varscaleQ14 >> 1;  // H_T_HQ19=65536 (16-17=-1)
       ssh = sh_lo >> 1;  // sqrt_nrg is in Qssh.
       sh = ssh - 14;
       tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
       tmp32c = sqrt_nrg + tmp32b;  // Qssh  (denominator)
-      tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0);  //Q24 (numerator)
+      tmp32a = varscaleQ14 * snrq;  // Q24 (numerator)
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
@@ -918,14 +916,13 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
       /* add hearing threshold and compute the gain */
       /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
 
-      //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
       tmp32a = varscaleQ14 >> 1;  // H_T_HQ19=65536 (16-17=-1)
 
       ssh = sh_hi >> 1;  // |sqrt_nrg| is in Qssh.
       sh = ssh - 14;
       tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
       tmp32c = sqrt_nrg + tmp32b;  // Qssh  (denominator)
-      tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0);  //Q24 (numerator)
+      tmp32a = varscaleQ14 * snrq;  // Q24 (numerator)
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
index e6ca7987fd..8018a120aa 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
@@ -212,7 +212,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
   /* bias towards pitch lag of previous frame */
   tmp32a = WebRtcIsacfix_Log2Q8((uint32_t) old_lagQ8) - 2304;
       // log2(0.5*oldlag) in Q8
-  tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0;
+  tmp32b = oldgQ12 * oldgQ12 >> 10;  // Q12 & * 4.0;
   gain_bias16 = (int16_t) tmp32b;  //Q12
   if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12
 
@@ -222,12 +222,12 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
     if (crrvecQ8_1[k]>0) {
       tmp32b = WebRtcIsacfix_Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2)));
       tmp16a = (int16_t) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
-      tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10
+      tmp32c = tmp16a * tmp16a >> 6;  // Q10
       tmp16b = (int16_t) tmp32c; // Q10 & <8
-      tmp32d = WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 177 , 8); // mult with ln2 in Q8
+      tmp32d = tmp16b * 177 >> 8;  // mult with ln2 in Q8
       tmp16c = (int16_t) tmp32d; // Q10 & <4
       tmp16d = Exp2Q10((int16_t) -tmp16c); //Q10
-      tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10  & * 0.5
+      tmp32c = gain_bias16 * tmp16d >> 13;  // Q10  & * 0.5
       bias16 = (int16_t) (1024 + tmp32c); // Q10
       tmp32b = WebRtcIsacfix_Log2Q8((uint32_t)bias16) - 2560;
           // Q10 in -> Q8 out with 10*2^8 offset
@@ -306,7 +306,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
       tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -42, 8);
+      tmp32b = (int16_t)tmp32a * -42 >> 8;
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
@@ -330,7 +330,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
   {
     tmp32a = k << 7; // 0.5*k Q8
     tmp32b = tmp32a * 2 - ratq;  // Q8
-    tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32b, (int16_t) tmp32b, 8); // Q8
+    tmp32c = (int16_t)tmp32b * (int16_t)tmp32b >> 8;  // Q8
 
     tmp32b = tmp32c + (ratq >> 1);
         // (k-r)^2 + 0.5 * r  Q8
@@ -380,7 +380,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */
       tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -82, 8);
+      tmp32b = (int16_t)tmp32a * -82 >> 8;
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
index 7ba7b69daa..81bdff5e4f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
@@ -34,12 +34,8 @@ void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) {
   csum32 = 0;
   x = in + PITCH_MAX_LAG / 2 + 2;
   for (n = 0; n < PITCH_CORR_LEN2; n++) {
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[n],
-                                        (int16_t)in[n],
-                                        scaling);  // Q0
-    csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)x[n],
-                                        (int16_t)in[n],
-                                        scaling);  // Q0
+    ysum32 += in[n] * in[n] >> scaling;  // Q0
+    csum32 += x[n] * in[n] >> scaling;  // Q0
   }
   logcorQ8 += PITCH_LAG_SPAN2 - 1;
   lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32) >> 1; // Q8, sqrt(ysum)
@@ -57,12 +53,9 @@ void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) {
 
   for (k = 1; k < PITCH_LAG_SPAN2; k++) {
     inptr = &in[k];
-    ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[k - 1],
-                                        (int16_t)in[k - 1],
-                                        scaling);
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[PITCH_CORR_LEN2 + k - 1],
-                                        (int16_t)in[PITCH_CORR_LEN2 + k - 1],
-                                        scaling);
+    ysum32 -= in[k - 1] * in[k - 1] >> scaling;
+    ysum32 += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1] >>
+        scaling;
 #ifdef WEBRTC_ARCH_ARM_NEON
     {
       int32_t vbuff[4];
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
index 06471beb33..eaf131df61 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
@@ -89,14 +89,12 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
 
     // Make output more periodic.
     for (k = 0; k < PITCH_SUBFRAMES; k++) {
-      gainsQ12[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-          gainsQ12[k], Gain, 14);
+      gainsQ12[k] = (int16_t)(gainsQ12[k] * Gain >> 14);
     }
   }
 
   // No interpolation if pitch lag step is big.
-  if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
-      (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
+  if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) {
     oldLagQ7 = lagsQ7[0];
     oldGainQ12 = gainsQ12[0];
   }
@@ -110,8 +108,7 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
                   lagdeltaQ7, kDivFactor, 15);
     curLagQ7 = oldLagQ7;
     gaindeltaQ12 = gainsQ12[k] - oldGainQ12;
-    gaindeltaQ12 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-                    gaindeltaQ12, kDivFactor, 15);
+    gaindeltaQ12 = (int16_t)(gaindeltaQ12 * kDivFactor >> 15);
 
     curGainQ12 = oldGainQ12;
     oldLagQ7 = lagsQ7[k];
@@ -173,8 +170,7 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
   oldLagQ7 = pfp->oldlagQ7;
 
   // No interpolation if pitch lag step is big.
-  if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
-      (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
+  if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) {
     oldLagQ7 = lagsQ7[0];
   }