diff --git a/webrtc/common_audio/signal_processing/complex_fft.c b/webrtc/common_audio/signal_processing/complex_fft.c
index 74b4258a8e..aaeda52ad9 100644
--- a/webrtc/common_audio/signal_processing/complex_fft.c
+++ b/webrtc/common_audio/signal_processing/complex_fft.c
@@ -126,11 +126,9 @@ int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
                          [wri]"r"(wri),
                          [cfftrnd]"r"(CFFTRND));
 #else
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
+                    tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CFFTRND;
 
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
+                    ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CFFTRND;
 #endif
 
                     tr32 >>= 15 - CFFTSFT;
@@ -270,11 +268,9 @@ int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
                     );
 #else
 
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
+                    tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CIFFTRND;
 
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
+                    ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CIFFTRND;
 #endif
                     tr32 >>= 15 - CIFFTSFT;
                     ti32 >>= 15 - CIFFTSFT;
diff --git a/webrtc/common_audio/signal_processing/division_operations.c b/webrtc/common_audio/signal_processing/division_operations.c
index 6aeb0fb2bf..eaa06a1ff9 100644
--- a/webrtc/common_audio/signal_processing/division_operations.c
+++ b/webrtc/common_audio/signal_processing/division_operations.c
@@ -106,8 +106,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
     // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
 
     // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
-            + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
+    tmpW32 = (den_hi * approx << 1) + ((den_low * approx >> 15) << 1);
     // tmpW32 = den * approx
 
     tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
@@ -117,8 +116,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
     tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
 
     // tmpW32 = 1/den in Q29
-    tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
-            >> 15)) << 1);
+    tmpW32 = (tmp_hi * approx + (tmp_low * approx >> 15)) << 1;
 
     // 1/den in hi and low format
     tmp_hi = (int16_t)(tmpW32 >> 16);
@@ -130,8 +128,8 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
 
     // num * (1/den) by 32 bit multiplication (result in Q28)
 
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
-            >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
+    tmpW32 = num_hi * tmp_hi + (num_hi * tmp_low >> 15) +
+        (num_low * tmp_hi >> 15);
 
     // Put result in Q31 (convert from Q28)
     tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
diff --git a/webrtc/common_audio/signal_processing/filter_ar.c b/webrtc/common_audio/signal_processing/filter_ar.c
index 7386808c61..2a1805c7ad 100644
--- a/webrtc/common_audio/signal_processing/filter_ar.c
+++ b/webrtc/common_audio/signal_processing/filter_ar.c
@@ -51,13 +51,13 @@ int WebRtcSpl_FilterAR(const int16_t* a,
         stop = (i < a_length) ? i + 1 : a_length;
         for (j = 1; j < stop; j++)
         {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
+          o -= *a_ptr * *filtered_ptr--;
+          oLOW -= *a_ptr++ * *filtered_low_ptr--;
         }
         for (j = i + 1; j < a_length; j++)
         {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
+          o -= *a_ptr * *state_ptr--;
+          oLOW -= *a_ptr++ * *state_low_ptr--;
         }
 
         o += (oLOW >> 12);
diff --git a/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c b/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
index 943b01c1d9..afec8393d3 100644
--- a/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
@@ -23,18 +23,14 @@ void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
                                int16_t B_length,
                                int16_t length)
 {
-    int32_t o;
     int i, j;
     for (i = 0; i < length; i++)
     {
-        const int16_t* b_ptr = &B[0];
-        const int16_t* x_ptr = &in_ptr[i];
-
-        o = (int32_t)0;
+        int32_t o = 0;
 
         for (j = 0; j < B_length; j++)
         {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
+          o += B[j] * in_ptr[i - j];
         }
 
         // If output is higher than 32768, saturate it. Same with negative side
diff --git a/webrtc/common_audio/signal_processing/ilbc_specific_functions.c b/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
index de870b239b..dae25a445f 100644
--- a/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
+++ b/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
@@ -68,16 +68,11 @@ void WebRtcSpl_AddAffineVectorToVector(int16_t *out, int16_t *in,
                                        int16_t gain, int32_t add_constant,
                                        int16_t right_shifts, int vector_length)
 {
-    int16_t *inPtr;
-    int16_t *outPtr;
     int i;
 
-    inPtr = in;
-    outPtr = out;
     for (i = 0; i < vector_length; i++)
     {
-        (*outPtr++) += (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (int32_t)add_constant) >> right_shifts);
+      out[i] += (int16_t)((in[i] * gain + add_constant) >> right_shifts);
     }
 }
 
@@ -85,15 +80,10 @@ void WebRtcSpl_AffineTransformVector(int16_t *out, int16_t *in,
                                      int16_t gain, int32_t add_constant,
                                      int16_t right_shifts, int vector_length)
 {
-    int16_t *inPtr;
-    int16_t *outPtr;
     int i;
 
-    inPtr = in;
-    outPtr = out;
     for (i = 0; i < vector_length; i++)
     {
-        (*outPtr++) = (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (int32_t)add_constant) >> right_shifts);
+      out[i] = (int16_t)((in[i] * gain + add_constant) >> right_shifts);
     }
 }
diff --git a/webrtc/common_audio/signal_processing/levinson_durbin.c b/webrtc/common_audio/signal_processing/levinson_durbin.c
index e07af5d3b6..eaf51430de 100644
--- a/webrtc/common_audio/signal_processing/levinson_durbin.c
+++ b/webrtc/common_audio/signal_processing/levinson_durbin.c
@@ -76,8 +76,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
     // Alpha = R[0] * (1-K^2)
 
-    temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
-            << 1); // temp1W32 = k^2 in Q31
+    temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1;  // = k^2 in Q31
 
     temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
     temp1W32 = (int32_t)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
@@ -87,9 +86,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
     tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
 
     // Calculate Alpha in Q31
-    temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
-            + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
-            + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
+    temp1W32 = (R_hi[0] * tmp_hi + (R_hi[0] * tmp_low >> 15) +
+        (R_low[0] * tmp_hi >> 15)) << 1;
 
     // Normalize Alpha and put it in hi and low format
 
@@ -113,10 +111,10 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
         for (j = 1; j < i; j++)
         {
-            // temp1W32 is in Q31
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
-                    + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
-                            + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
+          // temp1W32 is in Q31
+          temp1W32 += (R_hi[j] * A_hi[i - j] << 1) +
+              (((R_hi[j] * A_low[i - j] >> 15) +
+              (R_low[j] * A_hi[i - j] >> 15)) << 1);
         }
 
         temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
@@ -177,9 +175,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
                     + WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[j],1);
 
             // temp1W32 += K*A[i-j] in Q27
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
-                    + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
-                    + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
+            temp1W32 += (K_hi * A_hi[i - j] + (K_hi * A_low[i - j] >> 15) +
+                (K_low * A_hi[i - j] >> 15)) << 1;
 
             // Put Anew in hi and low format
             A_upd_hi[j] = (int16_t)(temp1W32 >> 16);
@@ -197,8 +194,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
         // Alpha = Alpha * (1-K^2)
 
-        temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
-                + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
+        temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1;  // K*K in Q31
 
         temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
         temp1W32 = (int32_t)0x7fffffffL - temp1W32; // 1 - K*K  in Q31
@@ -208,9 +204,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
         tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
 
         // Calculate Alpha = Alpha * (1-K^2) in Q31
-        temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
+        temp1W32 = (Alpha_hi * tmp_hi + (Alpha_hi * tmp_low >> 15) +
+            (Alpha_low * tmp_hi >> 15)) << 1;
 
         // Normalize Alpha and store it on hi and low format
 
diff --git a/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c b/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
index 5fb4d8596b..edcebd4e63 100644
--- a/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
+++ b/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
@@ -30,7 +30,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
     for (m = use_order - 1; m > 0; m--)
     {
         // (1 - k^2) in Q30
-        tmp_inv_denom32 = ((int32_t)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
+        tmp_inv_denom32 = 1073741823 - k16[m] * k16[m];
         // (1 - k^2) in Q15
         tmp_inv_denom16 = (int16_t)(tmp_inv_denom32 >> 15);
 
@@ -39,8 +39,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
             // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
 
             // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
-            tmp32[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)a16[k], 16)
-                    - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
+            tmp32[k] = (a16[k] << 16) - (k16[m] * a16[m - k + 1] << 1);
 
             tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
         }
diff --git a/webrtc/common_audio/signal_processing/spl_sqrt.c b/webrtc/common_audio/signal_processing/spl_sqrt.c
index 1de6ccd713..24db4f822c 100644
--- a/webrtc/common_audio/signal_processing/spl_sqrt.c
+++ b/webrtc/common_audio/signal_processing/spl_sqrt.c
@@ -49,16 +49,16 @@ int32_t WebRtcSpl_SqrtLocal(int32_t in)
     A >>= 16;
     A = A * A * 2; // A = (x/2)^4
     t16 = (int16_t)(A >> 16);
-    B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
+    B += -20480 * t16 * 2;  // B = B - 0.625*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
 
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
+    A = x_half * t16 * 2;  // A = (x/2)^5
     t16 = (int16_t)(A >> 16);
-    B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
+    B += 28672 * t16 * 2;  // B = B + 0.875*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
 
     t16 = (int16_t)(x2 >> 16);
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
+    A = x_half * t16 * 2;  // A = x/2^3
 
     B = B + (A >> 1); // B = B + 0.5*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
@@ -166,7 +166,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
 
         t16 = (int16_t)(A >> 16);  // t16 = AH
 
-        A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
+        A = k_sqrt_2 * t16 * 2;  // A = 1/sqrt(2)*t16
         A = A + ((int32_t)32768); // Round off
         A = A & ((int32_t)0x7fff0000); // Round off
 
diff --git a/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
index fc438c6373..f9570f313e 100644
--- a/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
+++ b/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -27,7 +27,7 @@ void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
     for (m = 0; m < vector_length; m++)
     {
         tmp = xQ15[m];
-        sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
+        sq = tmp * tmp;  // x^2 in Q30
         sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
         sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
         yQ15[m] = (int16_t)sq;
diff --git a/webrtc/common_audio/signal_processing/vector_scaling_operations.c b/webrtc/common_audio/signal_processing/vector_scaling_operations.c
index 736f62c71f..9ae7480eb7 100644
--- a/webrtc/common_audio/signal_processing/vector_scaling_operations.c
+++ b/webrtc/common_audio/signal_processing/vector_scaling_operations.c
@@ -157,9 +157,8 @@ int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
 
   for (i = 0; i < length; i++) {
     out_vector[i] = (int16_t)((
-        WEBRTC_SPL_MUL_16_16(in_vector1[i], in_vector1_scale)
-        + WEBRTC_SPL_MUL_16_16(in_vector2[i], in_vector2_scale)
-        + round_value) >> right_shifts);
+        in_vector1[i] * in_vector1_scale + in_vector2[i] * in_vector2_scale +
+        round_value) >> right_shifts);
   }
 
   return 0;