diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index 3a92ab7981..72a110807c 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -303,8 +303,7 @@ void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const int16_t* echo_path)
     memcpy(aecm->channelAdapt16, echo_path, sizeof(int16_t) * PART_LEN1);
     for (i = 0; i < PART_LEN1; i++)
     {
-        aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-            (int32_t)(aecm->channelAdapt16[i]), 16);
+        aecm->channelAdapt32[i] = (int32_t)aecm->channelAdapt16[i] << 16;
     }
 
     // Reset channel storing variables
@@ -370,16 +369,12 @@ static void ResetAdaptiveChannelC(AecmCore_t* aecm)
     // Restore the W32 channel
     for (i = 0; i < PART_LEN; i += 4)
     {
-        aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-                (int32_t)aecm->channelStored[i], 16);
-        aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32(
-                (int32_t)aecm->channelStored[i + 1], 16);
-        aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32(
-                (int32_t)aecm->channelStored[i + 2], 16);
-        aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32(
-                (int32_t)aecm->channelStored[i + 3], 16);
+        aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
+        aecm->channelAdapt32[i + 1] = (int32_t)aecm->channelStored[i + 1] << 16;
+        aecm->channelAdapt32[i + 2] = (int32_t)aecm->channelStored[i + 2] << 16;
+        aecm->channelAdapt32[i + 3] = (int32_t)aecm->channelStored[i + 3] << 16;
     }
-    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)aecm->channelStored[i], 16);
+    aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
 }
 
 // Initialize function pointers for ARM Neon platform.
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
index c1cc7ebadb..f608a285ab 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
@@ -687,7 +687,7 @@ static void ComfortNoise(AecmCore_t* aecm,
   {
     // Shift to the noise domain.
     tmp32 = (int32_t)dfa[i];
-    outLShift32 = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+    outLShift32 = tmp32 << shiftFromNearToNoise;
 
     if (outLShift32 < aecm->noiseEst[i])
     {
@@ -750,7 +750,7 @@ static void ComfortNoise(AecmCore_t* aecm,
     if (tmp32 > 32767)
     {
       tmp32 = 32767;
-      aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+      aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
     }
     noiseRShift16[i] = (int16_t)tmp32;
 
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_mips.c b/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
index 909a52e48e..4a07042ff7 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
@@ -606,8 +606,7 @@ void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore_t* aecm) {
     );
   }
 
-  aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-                              (int32_t)aecm->channelStored[i], 16);
+  aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
 }
 #endif  // #if defined(MIPS_DSP_R1_LE)
 
@@ -1471,11 +1470,11 @@ static void ComfortNoise(AecmCore_t* aecm,
 
     if (tmp32 > 32767) {
       tmp32 = 32767;
-      aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
+      aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
     }
     if (tmp321 > 32767) {
       tmp321 = 32767;
-      aecm->noiseEst[i+1] = WEBRTC_SPL_LSHIFT_W32(tmp321, shiftFromNearToNoise);
+      aecm->noiseEst[i+1] = tmp321 << shiftFromNearToNoise;
     }
 
     __asm __volatile (
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_neon.c b/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
index 4031790843..0a0b768e45 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
@@ -330,8 +330,7 @@ void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore_t* aecm) {
 
   for (i = 0; i < PART_LEN - 7; i += 8) {
     // aecm->channelAdapt16[i] = aecm->channelStored[i];
-    // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((int32_t)
-    //                           aecm->channelStored[i], 16);
+    // aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
     __asm __volatile("vld1.16 {d24, d25}, [%0, :128]" : :
             "r"(&aecm->channelStored[i]) : "q12");
     __asm __volatile("vst1.16 {d24, d25}, [%0, :128]" : :
@@ -342,6 +341,5 @@ void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore_t* aecm) {
             "r"(&aecm->channelAdapt32[i]): "q10", "q11");
   }
   aecm->channelAdapt16[i] = aecm->channelStored[i];
-  aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-      (int32_t)aecm->channelStored[i], 16);
+  aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
 }
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/analog_agc.c
index 32929ddaa0..4dbb4ec14a 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -455,7 +455,7 @@ int WebRtcAgc_VirtualMic(void *agcInst, int16_t *in_near, int16_t *in_near_H,
         stt->lowLevelSignal = 0;
     }
 
-    micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
+    micLevelTmp = micLevelIn << stt->scale;
     /* Set desired level */
     gainIdx = stt->micVol;
     if (stt->micVol > stt->maxAnalog)
@@ -769,7 +769,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
     Agc_t *stt;
 
     stt = (Agc_t *)state;
-    inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
+    inMicLevelTmp = inMicLevel << stt->scale;
 
     if (inMicLevelTmp > stt->maxAnalog)
     {
@@ -1089,7 +1089,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                     stt->msTooLow = 0;
 
                     /* Normalize the volume level */
-                    tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
+                    tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
                     if (stt->maxInit != stt->minLevel)
                     {
                         volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
@@ -1150,7 +1150,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                     stt->msTooLow = 0;
 
                     /* Normalize the volume level */
-                    tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
+                    tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
                     if (stt->maxInit != stt->minLevel)
                     {
                         volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
@@ -1612,8 +1612,8 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel,
     // TODO(bjornv): Investigate if we really need to scale up a small range now when we have
     // a guard against zero-increments. For now, we do not support scale up (scale = 0).
     stt->scale = 0;
-    maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale);
-    minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale);
+    maxLevel <<= stt->scale;
+    minLevel <<= stt->scale;
 
     /* Make minLevel and maxLevel static in AdaptiveDigital */
     if (stt->agcMode == kAgcModeAdaptiveDigital)
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index 922eafb1ed..12b08b0625 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -116,9 +116,8 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
     //  limiterLvlX = analogTarget - limiterOffset
     //  limiterLvl  = targetLevelDbfs + limiterOffset/compRatio
     limiterLvlX = analogTarget - limiterOffset;
-    limiterIdx = 2
-            + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)limiterLvlX, 13),
-                                        (kLog10_2 / 2));
+    limiterIdx =
+        2 + WebRtcSpl_DivW32W16ResW16((int32_t)limiterLvlX << 13, kLog10_2 / 2);
     tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
     limiterLvl = targetLevelDbfs + tmp16no1;
 
@@ -144,7 +143,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
         inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
 
         // Calculate diffGain-inLevel, to map using the genFuncTable
-        inLevel = WEBRTC_SPL_LSHIFT_W32((int32_t)diffGain, 14) - inLevel; // Q14
+        inLevel = ((int32_t)diffGain << 14) - inLevel;  // Q14
 
         // Make calculations on abs(inLevel) and compensate for the sign afterwards.
         absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14
@@ -186,7 +185,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
                 logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
             }
         }
-        numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
+        numFIX = (maxGain * constMaxGain) << 6;  // Q14
         numFIX -= (int32_t)logApprox * diffGain;  // Q14
 
         // Calculate ratio
@@ -199,7 +198,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
         {
             zeros = WebRtcSpl_NormW32(den) + 8;
         }
-        numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
+        numFIX <<= zeros;  // Q(14+zeros)
 
         // Shift den so we end up in Qy1
         tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
@@ -214,7 +213,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
         if (limiterEnable && (i < limiterIdx))
         {
             tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
-            tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
+            tmp32 -= limiterLvl << 14;  // Q14
             y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
         }
         if (y32 > 39000)
@@ -226,7 +225,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
             tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
             tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
         }
-        tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
+        tmp32 += 16 << 14;  // in Q14 (Make sure final output is in Q16)
 
         // Calculate power
         if (tmp32 > 0)
@@ -236,10 +235,10 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
             if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
             {
                 tmp16 = (2 << 14) - constLinApprox;
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
+                tmp32no2 = (1 << 14) - fracPart;
                 tmp32no2 *= tmp16;
                 tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
+                tmp32no2 = (1 << 14) - tmp32no2;
             } else
             {
                 tmp16 = constLinApprox - (1 << 14);
@@ -247,8 +246,8 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
                 tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
             }
             fracPart = (uint16_t)tmp32no2;
-            gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
-                    + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
+            gainTable[i] =
+                (1 << intPart) + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
         } else
         {
             gainTable[i] = 0;
@@ -462,7 +461,7 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
         {
             zeros = 31;
         }
-        tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
+        tmp32 = (cur_level << zeros) & 0x7FFFFFFF;
         frac = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
         tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
         gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
@@ -487,7 +486,7 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
     {
         zeros_fast = 31;
     }
-    tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
+    tmp32 = (stt->capacitorFast << zeros_fast) & 0x7FFFFFFF;
     zeros_fast <<= 9;
     zeros_fast -= (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
 
@@ -570,8 +569,8 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
 
     // Apply gain
     // handle first sub frame separately
-    delta = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
-    gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
+    delta = (gains[1] - gains[0]) << (4 - L2);
+    gain32 = gains[0] << 4;
     // iterate over samples
     for (n = 0; n < L; n++)
     {
@@ -615,8 +614,8 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
     // iterate over subframes
     for (k = 1; k < 10; k++)
     {
-        delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
-        gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
+        delta = (gains[k+1] - gains[k]) << (4 - L2);
+        gain32 = gains[k] << 4;
         // iterate over samples
         for (n = 0; n < L; n++)
         {
@@ -648,14 +647,14 @@ void WebRtcAgc_InitVad(AgcVad_t *state)
     state->meanLongTerm = 15 << 10;
 
     // variance of input level (Q8)
-    state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
+    state->varianceLongTerm = 500 << 8;
 
     state->stdLongTerm = 0; // standard deviation of input level in dB
     // short-term average input level (Q10)
     state->meanShortTerm = 15 << 10;
 
     // short-term variance of input level (Q8)
-    state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
+    state->varianceShortTerm = 500 << 8;
 
     state->stdShortTerm = 0; // short-term standard deviation of input level in dB
     state->counter = 3; // counts updates
@@ -760,7 +759,7 @@ int16_t WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
 
     // update short-term estimate of standard deviation in energy level (Q10)
     tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
+    tmp32 = (state->varianceShortTerm << 12) - tmp32;
     state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
 
     // update long-term estimate of mean energy level (Q10)
@@ -776,7 +775,7 @@ int16_t WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
 
     // update long-term estimate of standard deviation in energy level (Q10)
     tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
+    tmp32 = (state->varianceLongTerm << 12) - tmp32;
     state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
 
     // update voice activity measure (Q10)
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
index 0a23ff2355..90a17cd77f 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
@@ -82,8 +82,8 @@ int Filter(FilterState* hpf, int16_t* data, int length) {
     y[2] = y[0];
     y[3] = y[1];
     y[0] = static_cast<int16_t>(tmp_int32 >> 13);
-    y[1] = static_cast<int16_t>((tmp_int32 -
-        WEBRTC_SPL_LSHIFT_W32(static_cast<int32_t>(y[0]), 13)) << 2);
+    y[1] = static_cast<int16_t>(
+        (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
 
     // Rounding in Q12, i.e. add 2^11
     tmp_int32 += 2048;
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 56db109244..024cfa9ceb 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -325,7 +325,7 @@ static void UpdateNoiseEstimate(NsxInst_t* inst, int offset) {
     if (tmp16 < 0) {
       tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
     } else {
-      tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+      tmp32no1 <<= tmp16;
     }
     inst->noiseEstQuantile[i] = WebRtcSpl_SatW32ToW16(tmp32no1);
   }
@@ -604,7 +604,7 @@ void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
 
   // Calculate output: 2^tmp32no1
   // Output in Q(minNorm-stages)
-  tmp32no1 += WEBRTC_SPL_LSHIFT_W32((int32_t)(inst->minNorm - inst->stages), 11);
+  tmp32no1 += (inst->minNorm - inst->stages) << 11;
   if (tmp32no1 > 0) {
     int_part = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
     frac_part = (int16_t)(tmp32no1 & 0x000007ff); // Q11
@@ -1071,7 +1071,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, uint16_t* magn) {
   logCurSpectralFlatness = (int32_t)avgSpectralFlatnessNum;
   logCurSpectralFlatness += ((int32_t)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
   logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
-  logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
+  logCurSpectralFlatness <<= (10 - inst->stages);  // Q17
   tmp32 = (int32_t)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
                                         & 0x0001FFFF)); //Q17
   intPart = -(int16_t)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
@@ -1079,7 +1079,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, uint16_t* magn) {
   if (intPart > 0) {
     currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
   } else {
-    currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
+    currentSpectralFlatness = tmp32 << -intPart;
   }
 
   //time average update of spectral flatness feature
@@ -1387,7 +1387,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
       tmp_1_w32 = (int32_t)matrix_determinant;
       tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
       tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
-      tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((int32_t)sum_log_i_square, 4);
+      tmp_1_w32 -= (int32_t)sum_log_i_square << 4;
       tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)
                        (inst->magnLen - kStartBand), kSumSquareLogIndex[65], 2);
       matrix_determinant = (int16_t)tmp_1_w32;
@@ -1400,7 +1400,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
     if (zeros < 0) {
       zeros = 0;
     }
-    tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
+    tmp_1_w32 = sum_log_magn << 1;  // Q9
     sum_log_magn_u16 = (uint16_t)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
 
     // Calculate and update pinkNoiseNumerator. Result in Q11.
@@ -1417,7 +1417,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
     tmp_2_w32 -= (int32_t)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
     matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
     tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
-    tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((int32_t)net_norm, 11); // Q11
+    tmp_2_w32 += (int32_t)net_norm << 11;  // Q11
     if (tmp_2_w32 < 0) {
       tmp_2_w32 = 0;
     }
@@ -1944,8 +1944,8 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
         tmp32no1 *= ONE_MINUS_GAMMA_PAUSE_Q8;  // Q(8+prevQMagn+nShifts)
         tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
       } else {
-        tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)magnU16[i], nShifts)
-                   - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
+        // In Q(qMagn+nShifts)
+        tmp32no1 = ((int32_t)magnU16[i] << nShifts) - inst->avgMagnPause[i];
         tmp32no1 *= ONE_MINUS_GAMMA_PAUSE_Q8;  // Q(8+prevQMagn+nShifts)
         tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
       }
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index 2fce49b905..9e40060591 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -234,8 +234,8 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
         // Quadratic approximation of 2^frac
         tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
         tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
-        invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
-                   + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
+        invLrtFX = (1 << (8 + intPart)) +
+            WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
 
         normTmp = WebRtcSpl_NormW32(invLrtFX);
         normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
@@ -253,8 +253,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
             invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
           }
 
-          tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)inst->priorNonSpeechProb,
-                                           8); // Q22
+          tmp32no1 = (int32_t)inst->priorNonSpeechProb << 8;  // Q22
 
           nonSpeechProbFinal[i] = tmp32no1 /
               (inst->priorNonSpeechProb + invLrtFX);  // Q8
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_neon.c b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
index 6041785503..0a72a08ef2 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_neon.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
@@ -102,7 +102,7 @@ static void UpdateNoiseEstimateNeon(NsxInst_t* inst, int offset) {
     // if (tmp16 < 0) {
     //   tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
     // } else {
-    //   tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+    //   tmp32no1 <<= tmp16;
     // }
     v32x4B = vshlq_s32(v32x4A, v32x4B);
 
@@ -127,7 +127,7 @@ static void UpdateNoiseEstimateNeon(NsxInst_t* inst, int offset) {
   if (tmp16 < 0) {
     tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
   } else {
-    tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
+    tmp32no1 <<= tmp16;
   }
   *ptr_noiseEstQuantile = WebRtcSpl_SatW32ToW16(tmp32no1);
 }