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 19379bf084..3a60c1db8e 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -57,12 +57,12 @@
 #endif
 #endif
 
-#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
-    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
-    + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
-#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
-    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
-    + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b)          \
+  (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 5) + \
+   (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b)          \
+  (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 2) + \
+   (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
 #define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
     ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
     + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x2000) >> 14))
@@ -82,8 +82,7 @@
 
 // Shifting with negative numbers allowed
 // Positive means left shift
-#define WEBRTC_SPL_SHIFT_W32(x, c) \
-    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) ((c) >= 0 ? (x) * (1 << (c)) : (x) >> -(c))
 
 // Shifting with negative numbers not allowed
 // We cannot do casting here due to signed/unsigned problem
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 e3de437a58..cf2d507cb5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -182,7 +182,7 @@ int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
 
     for (k = 0; k < FRAMESAMPLES/2; k++)
     {
-      Vector_Word32_1[k] = (Vector_Word16_2[k] * gainQ13) << 3;  // Q25
+      Vector_Word32_1[k] = (Vector_Word16_2[k] * gainQ13) * (1 << 3);  // Q25
     }
 
 
@@ -192,7 +192,7 @@ int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
 
     /* --- Store Highpass Residual --- */
     for (k = 0; k < FRAMESAMPLES/2; k++)
-      Vector_Word32_1[k] = Vector_Word32_2[k] << 9;  // Q16 -> Q25
+      Vector_Word32_1[k] = Vector_Word32_2[k] * (1 << 9);  // Q16 -> Q25
 
     for( k = 0; k < PITCH_MAX_LAG + 10; k++ )
       (ISACdec_obj->plcstr_obj).prevHP[k] = Vector_Word32_1[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10) + k];
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 c9ddf93580..8e351277c3 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
@@ -255,7 +255,7 @@ static void CalcInvArSpec(const int16_t *ARCoefQ12,
   }
 
   for (k=0; k<FRAMESAMPLES/8; k++) {
-    int32_t diff_q16 = diffQ16[k] << shftVal;
+    int32_t diff_q16 = diffQ16[k] * (1 << shftVal);
     CurveQ16[FRAMESAMPLES / 4 - 1 - k] = CurveQ16[k] - diff_q16;
     CurveQ16[k] += diff_q16;
   }
@@ -864,8 +864,8 @@ void WebRtcIsacfix_MatrixProduct1C(const int16_t matrix0[],
       matrix0_index = matrix0_index_factor1 * (*matrix0_index_factor2);
       matrix1_index = matrix1_index_factor1 * (*matrix1_index_factor2);
       for (n = 0; n < inner_loop_count; n++) {
-        sum32 += (WEBRTC_SPL_MUL_16_32_RSFT16(matrix0[matrix0_index],
-                                              matrix1[matrix1_index] << shift));
+        sum32 += WEBRTC_SPL_MUL_16_32_RSFT16(
+            matrix0[matrix0_index], matrix1[matrix1_index] * (1 << shift));
         matrix0_index += matrix0_index_step;
         matrix1_index += matrix1_index_step;
       }
@@ -1042,7 +1042,8 @@ int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
     /* hi band LAR coeffs */
     for (n=0; n<ORDERHI; n++, pos++, poss++) {
       // ((Q13*Q17)>>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13
-      tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]) << 3;
+      tmp32 =
+          WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]) * (1 << 3);
       tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17
       LPCCoefQ17[pos] = tmp32;
     }
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 b5d91d4219..2e92578cf7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
@@ -50,33 +50,37 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C(
     // Process channel 1:
     in_out = data_ch1[n];
     a = factor_ch1[0] * in_out;  // Q15 * Q0 = Q15
-    a <<= 1;  // Q15 -> Q16
+    a *= 1 << 1;  // Q15 -> Q16
     b = WebRtcSpl_AddSatW32(a, state0_ch1);
     a = -factor_ch1[0] * (int16_t)(b >> 16);  // Q15
-    state0_ch1 = WebRtcSpl_AddSatW32(a << 1, (uint32_t)in_out << 16);  // Q16
+    state0_ch1 =
+        WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16));  // Q16
     in_out = (int16_t) (b >> 16);  // Save as Q0
 
     a = factor_ch1[1] * in_out;  // Q15 * Q0 = Q15
-    a <<= 1; // Q15 -> Q16
+    a *= 1 << 1; // Q15 -> Q16
     b = WebRtcSpl_AddSatW32(a, state1_ch1);  // Q16
     a = -factor_ch1[1] * (int16_t)(b >> 16);  // Q15
-    state1_ch1 = WebRtcSpl_AddSatW32(a << 1, (uint32_t)in_out << 16);  // Q16
+    state1_ch1 =
+        WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16));  // Q16
     data_ch1[n] = (int16_t) (b >> 16);  // Save as Q0
 
     // Process channel 2:
     in_out = data_ch2[n];
     a = factor_ch2[0] * in_out;  // Q15 * Q0 = Q15
-    a <<= 1;  // Q15 -> Q16
+    a *= 1 << 1;  // Q15 -> Q16
     b = WebRtcSpl_AddSatW32(a, state0_ch2);  // Q16
     a = -factor_ch2[0] * (int16_t)(b >> 16);  // Q15
-    state0_ch2 = WebRtcSpl_AddSatW32(a << 1, (uint32_t)in_out << 16);  // Q16
+    state0_ch2 =
+        WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16));  // Q16
     in_out = (int16_t) (b >> 16);  // Save as Q0
 
     a = factor_ch2[1] * in_out;  // Q15 * Q0 = Q15
-    a <<= 1;  // Q15 -> Q16
+    a *= (1 << 1);  // Q15 -> Q16
     b = WebRtcSpl_AddSatW32(a, state1_ch2);  // Q16
     a = -factor_ch2[1] * (int16_t)(b >> 16);  // Q15
-    state1_ch2 = WebRtcSpl_AddSatW32(a << 1, (uint32_t)in_out << 16);  // Q16
+    state1_ch2 =
+        WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16));  // Q16
     data_ch2[n] = (int16_t) (b >> 16);  // Save as Q0
   }
 
@@ -144,11 +148,11 @@ void WebRtcIsacfix_HighpassFilterFixDec32C(int16_t *io,
     c = in + ((a1 + b1) >> 7);  // Q0.
     io[k] = (int16_t)WebRtcSpl_SatW32ToW16(c);  // Write output as Q0.
 
-    c = (in << 2) - a2 - b2;  // In Q2.
+    c = in * (1 << 2) - a2 - b2;  // In Q2.
     c = (int32_t)WEBRTC_SPL_SAT(536870911, c, -536870912);
 
     state1 = state0;
-    state0 = c << 2;  // Write state as Q4
+    state0 = c * (1 << 2);  // Write state as Q4
   }
   state[0] = state0;
   state[1] = state1;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
index 22224a8071..2e9d0664ef 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
@@ -274,7 +274,7 @@ void WebRtcIsacfix_NormLatticeFilterAr(size_t orderCoef,
     for (i=0;i<HALF_SUBFRAMELEN;i++)
     {
 
-      tmp32 = lat_inQ25[i + temp1] << 1;  // Q25->Q26
+      tmp32 = lat_inQ25[i + temp1] * (1 << 1);  // Q25->Q26
       tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh)
       tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
index 621f921013..362610a1c4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
@@ -132,10 +132,10 @@ void WebRtcIsacfix_Spec2TimeC(int16_t *inreQ7, int16_t *inimQ7, int32_t *outre1Q
     tmp1rQ14 = -WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 - 1 - k];
     tmp1iQ14 = WebRtcIsacfix_kSinTab2[k];
 
-    tmpInRe = inreQ7[k] << 9;  // Q7 -> Q16
-    tmpInIm = inimQ7[k] << 9;  // Q7 -> Q16
-    tmpInRe2 = inreQ7[FRAMESAMPLES / 2 - 1 - k] << 9;  // Q7 -> Q16
-    tmpInIm2 = inimQ7[FRAMESAMPLES / 2 - 1 - k] << 9;  // Q7 -> Q16
+    tmpInRe = inreQ7[k] * (1 << 9);  // Q7 -> Q16
+    tmpInIm = inimQ7[k] * (1 << 9);  // Q7 -> Q16
+    tmpInRe2 = inreQ7[FRAMESAMPLES / 2 - 1 - k] * (1 << 9);  // Q7 -> Q16
+    tmpInIm2 = inimQ7[FRAMESAMPLES / 2 - 1 - k] * (1 << 9);  // Q7 -> Q16
 
     xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm);
     xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe);
@@ -184,8 +184,8 @@ void WebRtcIsacfix_Spec2TimeC(int16_t *inreQ7, int16_t *inimQ7, int32_t *outre1Q
     }
   } else {
     for (k=0; k<240; k++) {
-      outre1Q16[k] = inreQ7[k] << -sh;  // Q(16+sh) -> Q16
-      outre2Q16[k] = inimQ7[k] << -sh;  // Q(16+sh) -> Q16
+      outre1Q16[k] = inreQ7[k] * (1 << -sh);  // Q(16+sh) -> Q16
+      outre2Q16[k] = inimQ7[k] * (1 << -sh);  // Q(16+sh) -> Q16
     }
   }