diff --git a/webrtc/common_audio/fir_filter.cc b/webrtc/common_audio/fir_filter.cc
index a5411b888..a5160fdc8 100644
--- a/webrtc/common_audio/fir_filter.cc
+++ b/webrtc/common_audio/fir_filter.cc
@@ -85,7 +85,7 @@ FIRFilterC::FIRFilterC(const float* coefficients, size_t coefficients_length)
   for (size_t i = 0; i < coefficients_length_; ++i) {
     coefficients_[i] = coefficients[coefficients_length_ - i - 1];
   }
-  memset(state_.get(), 0.f, state_length_ * sizeof(state_[0]));
+  memset(state_.get(), 0, state_length_ * sizeof(state_[0]));
 }
 
 void FIRFilterC::Filter(const float* in, size_t length, float* out) {
diff --git a/webrtc/common_audio/fir_filter_sse.cc b/webrtc/common_audio/fir_filter_sse.cc
index 7d873a735..6e7ae70e3 100644
--- a/webrtc/common_audio/fir_filter_sse.cc
+++ b/webrtc/common_audio/fir_filter_sse.cc
@@ -31,14 +31,14 @@ FIRFilterSSE2::FIRFilterSSE2(const float* coefficients,
                         16))) {
   // Add zeros at the end of the coefficients.
   size_t padding = coefficients_length_ - coefficients_length;
-  memset(coefficients_.get(), 0.f, padding * sizeof(coefficients_[0]));
+  memset(coefficients_.get(), 0, padding * sizeof(coefficients_[0]));
   // The coefficients are reversed to compensate for the order in which the
   // input samples are acquired (most recent last).
   for (size_t i = 0; i < coefficients_length; ++i) {
     coefficients_[i + padding] = coefficients[coefficients_length - i - 1];
   }
   memset(state_.get(),
-         0.f,
+         0,
          (max_input_length + state_length_) * sizeof(state_[0]));
 }
 
diff --git a/webrtc/common_audio/resampler/sinc_resampler.cc b/webrtc/common_audio/resampler/sinc_resampler.cc
index 84f8125b5..165cb356c 100644
--- a/webrtc/common_audio/resampler/sinc_resampler.cc
+++ b/webrtc/common_audio/resampler/sinc_resampler.cc
@@ -222,23 +222,22 @@ void SincResampler::InitializeKernel() {
 
     for (int i = 0; i < kKernelSize; ++i) {
       const int idx = i + offset_idx * kKernelSize;
-      const float pre_sinc = M_PI * (i - kKernelSize / 2 - subsample_offset);
+      const float pre_sinc =
+          static_cast<float>(M_PI * (i - kKernelSize / 2 - subsample_offset));
       kernel_pre_sinc_storage_[idx] = pre_sinc;
 
       // Compute Blackman window, matching the offset of the sinc().
       const float x = (i - subsample_offset) / kKernelSize;
-      const float window = kA0 - kA1 * cos(2.0 * M_PI * x) + kA2
-          * cos(4.0 * M_PI * x);
+      const float window = static_cast<float>(kA0 - kA1 * cos(2.0 * M_PI * x) +
+          kA2 * cos(4.0 * M_PI * x));
       kernel_window_storage_[idx] = window;
 
       // Compute the sinc with offset, then window the sinc() function and store
       // at the correct offset.
-      if (pre_sinc == 0) {
-        kernel_storage_[idx] = sinc_scale_factor * window;
-      } else {
-        kernel_storage_[idx] =
-            window * sin(sinc_scale_factor * pre_sinc) / pre_sinc;
-      }
+      kernel_storage_[idx] = static_cast<float>(window *
+          ((pre_sinc == 0) ?
+              sinc_scale_factor :
+              (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
     }
   }
 }
@@ -260,12 +259,10 @@ void SincResampler::SetRatio(double io_sample_rate_ratio) {
       const float window = kernel_window_storage_[idx];
       const float pre_sinc = kernel_pre_sinc_storage_[idx];
 
-      if (pre_sinc == 0) {
-        kernel_storage_[idx] = sinc_scale_factor * window;
-      } else {
-        kernel_storage_[idx] =
-            window * sin(sinc_scale_factor * pre_sinc) / pre_sinc;
-      }
+      kernel_storage_[idx] = static_cast<float>(window *
+          ((pre_sinc == 0) ?
+              sinc_scale_factor :
+              (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
     }
   }
 }
@@ -289,18 +286,19 @@ void SincResampler::Resample(int frames, float* destination) {
     //
     // Note: The loop construct here can severely impact performance on ARM
     // or when built with clang.  See https://codereview.chromium.org/18566009/
-    for (int i = ceil((block_size_ - virtual_source_idx_) / current_io_ratio);
+    for (int i = static_cast<int>(
+             ceil((block_size_ - virtual_source_idx_) / current_io_ratio));
          i > 0; --i) {
       assert(virtual_source_idx_ < block_size_);
 
       // |virtual_source_idx_| lies in between two kernel offsets so figure out
       // what they are.
-      const int source_idx = virtual_source_idx_;
+      const int source_idx = static_cast<int>(virtual_source_idx_);
       const double subsample_remainder = virtual_source_idx_ - source_idx;
 
       const double virtual_offset_idx =
           subsample_remainder * kKernelOffsetCount;
-      const int offset_idx = virtual_offset_idx;
+      const int offset_idx = static_cast<int>(virtual_offset_idx);
 
       // We'll compute "convolutions" for the two kernels which straddle
       // |virtual_source_idx_|.
@@ -347,7 +345,7 @@ void SincResampler::Resample(int frames, float* destination) {
 #undef CONVOLVE_FUNC
 
 int SincResampler::ChunkSize() const {
-  return block_size_ / io_sample_rate_ratio_;
+  return static_cast<int>(block_size_ / io_sample_rate_ratio_);
 }
 
 void SincResampler::Flush() {
@@ -373,8 +371,8 @@ float SincResampler::Convolve_C(const float* input_ptr, const float* k1,
   }
 
   // Linearly interpolate the two "convolutions".
-  return (1.0 - kernel_interpolation_factor) * sum1
-      + kernel_interpolation_factor * sum2;
+  return static_cast<float>((1.0 - kernel_interpolation_factor) * sum1 +
+      kernel_interpolation_factor * sum2);
 }
 
 }  // namespace webrtc
diff --git a/webrtc/common_audio/resampler/sinc_resampler_sse.cc b/webrtc/common_audio/resampler/sinc_resampler_sse.cc
index bdbe7b1ae..e00e0e5db 100644
--- a/webrtc/common_audio/resampler/sinc_resampler_sse.cc
+++ b/webrtc/common_audio/resampler/sinc_resampler_sse.cc
@@ -41,8 +41,10 @@ float SincResampler::Convolve_SSE(const float* input_ptr, const float* k1,
   }
 
   // Linearly interpolate the two "convolutions".
-  m_sums1 = _mm_mul_ps(m_sums1, _mm_set_ps1(1.0 - kernel_interpolation_factor));
-  m_sums2 = _mm_mul_ps(m_sums2, _mm_set_ps1(kernel_interpolation_factor));
+  m_sums1 = _mm_mul_ps(m_sums1, _mm_set_ps1(
+      static_cast<float>(1.0 - kernel_interpolation_factor)));
+  m_sums2 = _mm_mul_ps(m_sums2, _mm_set_ps1(
+      static_cast<float>(kernel_interpolation_factor)));
   m_sums1 = _mm_add_ps(m_sums1, m_sums2);
 
   // Sum components together.
diff --git a/webrtc/common_audio/signal_processing/copy_set_operations.c b/webrtc/common_audio/signal_processing/copy_set_operations.c
index 0c2493bd3..84d3bc429 100644
--- a/webrtc/common_audio/signal_processing/copy_set_operations.c
+++ b/webrtc/common_audio/signal_processing/copy_set_operations.c
@@ -17,8 +17,6 @@
  * WebRtcSpl_CopyFromEndW16()
  * WebRtcSpl_ZerosArrayW16()
  * WebRtcSpl_ZerosArrayW32()
- * WebRtcSpl_OnesArrayW16()
- * WebRtcSpl_OnesArrayW32()
  *
  * The description header can be found in signal_processing_library.h
  *
@@ -62,47 +60,21 @@ void WebRtcSpl_MemCpyReversedOrder(int16_t* dest, int16_t* source, int length)
     }
 }
 
-int16_t WebRtcSpl_CopyFromEndW16(const int16_t *vector_in,
-                                 int16_t length,
-                                 int16_t samples,
-                                 int16_t *vector_out)
+void WebRtcSpl_CopyFromEndW16(const int16_t *vector_in,
+                              int length,
+                              int samples,
+                              int16_t *vector_out)
 {
     // Copy the last <samples> of the input vector to vector_out
     WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
-
-    return samples;
 }
 
-int16_t WebRtcSpl_ZerosArrayW16(int16_t *vector, int16_t length)
+void WebRtcSpl_ZerosArrayW16(int16_t *vector, int length)
 {
     WebRtcSpl_MemSetW16(vector, 0, length);
-    return length;
 }
 
-int16_t WebRtcSpl_ZerosArrayW32(int32_t *vector, int16_t length)
+void WebRtcSpl_ZerosArrayW32(int32_t *vector, int length)
 {
     WebRtcSpl_MemSetW32(vector, 0, length);
-    return length;
-}
-
-int16_t WebRtcSpl_OnesArrayW16(int16_t *vector, int16_t length)
-{
-    int16_t i;
-    int16_t *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
-}
-
-int16_t WebRtcSpl_OnesArrayW32(int32_t *vector, int16_t length)
-{
-    int16_t i;
-    int32_t *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
 }
diff --git a/webrtc/common_audio/signal_processing/get_scaling_square.c b/webrtc/common_audio/signal_processing/get_scaling_square.c
index 606902de5..9b6049c24 100644
--- a/webrtc/common_audio/signal_processing/get_scaling_square.c
+++ b/webrtc/common_audio/signal_processing/get_scaling_square.c
@@ -17,14 +17,16 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-int WebRtcSpl_GetScalingSquare(int16_t *in_vector, int in_vector_length, int times)
+int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
+                                   int in_vector_length,
+                                   int times)
 {
-    int nbits = WebRtcSpl_GetSizeInBits(times);
+    int16_t nbits = WebRtcSpl_GetSizeInBits(times);
     int i;
     int16_t smax = -1;
     int16_t sabs;
     int16_t *sptr = in_vector;
-    int t;
+    int16_t t;
     int looptimes = in_vector_length;
 
     for (i = looptimes; i > 0; i--)
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 56bbbe66d..21c5663f8 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -123,9 +123,9 @@ void WebRtcSpl_Init();
 // Get SPL Version
 int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes);
 
-int WebRtcSpl_GetScalingSquare(int16_t* in_vector,
-                               int in_vector_length,
-                               int times);
+int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
+                                   int in_vector_length,
+                                   int times);
 
 // Copy and set operations. Implementation in copy_set_operations.c.
 // Descriptions at bottom of file.
@@ -138,18 +138,14 @@ void WebRtcSpl_MemSetW32(int32_t* vector,
 void WebRtcSpl_MemCpyReversedOrder(int16_t* out_vector,
                                    int16_t* in_vector,
                                    int vector_length);
-int16_t WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
-                                 int16_t in_vector_length,
-                                 int16_t samples,
-                                 int16_t* out_vector);
-int16_t WebRtcSpl_ZerosArrayW16(int16_t* vector,
-                                int16_t vector_length);
-int16_t WebRtcSpl_ZerosArrayW32(int32_t* vector,
-                                int16_t vector_length);
-int16_t WebRtcSpl_OnesArrayW16(int16_t* vector,
-                               int16_t vector_length);
-int16_t WebRtcSpl_OnesArrayW32(int32_t* vector,
-                               int16_t vector_length);
+void WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
+                              int in_vector_length,
+                              int samples,
+                              int16_t* out_vector);
+void WebRtcSpl_ZerosArrayW16(int16_t* vector,
+                             int vector_length);
+void WebRtcSpl_ZerosArrayW32(int32_t* vector,
+                             int vector_length);
 // End: Copy and set operations.
 
 
@@ -931,10 +927,10 @@ void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
  *
  ******************************************************************/
 
-void WebRtcSpl_DownsampleBy2(const int16_t* in, int16_t len,
+void WebRtcSpl_DownsampleBy2(const int16_t* in, int len,
                              int16_t* out, int32_t* filtState);
 
-void WebRtcSpl_UpsampleBy2(const int16_t* in, int16_t len,
+void WebRtcSpl_UpsampleBy2(const int16_t* in, int len,
                            int16_t* out, int32_t* filtState);
 
 /************************************************************
@@ -1110,8 +1106,6 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
 // Output:
 //      - out_vector        : Vector with the requested samples
 //
-// Return value             : Number of copied samples in |out_vector|
-//
 
 //
 // WebRtcSpl_ZerosArrayW16(...)
@@ -1126,24 +1120,6 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
 // Output:
 //      - vector        : Vector containing all zeros
 //
-// Return value         : Number of samples in vector
-//
-
-//
-// WebRtcSpl_OnesArrayW16(...)
-// WebRtcSpl_OnesArrayW32(...)
-//
-// Inserts the value "one" in all positions of a w16 and a w32 vector
-// respectively.
-//
-// Input:
-//      - vector_length : Number of samples in vector
-//
-// Output:
-//      - vector        : Vector containing all ones
-//
-// Return value         : Number of samples in vector
-//
 
 //
 // WebRtcSpl_VectorBitShiftW16(...)
diff --git a/webrtc/common_audio/signal_processing/include/spl_inl.h b/webrtc/common_audio/signal_processing/include/spl_inl.h
index 8bf452fea..d3cc6dee6 100644
--- a/webrtc/common_audio/signal_processing/include/spl_inl.h
+++ b/webrtc/common_audio/signal_processing/include/spl_inl.h
@@ -84,7 +84,7 @@ static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
 
 #if !defined(MIPS32_LE)
 static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
-  int bits;
+  int16_t bits;
 
   if (0xFFFF0000 & n) {
     bits = 16;
@@ -100,8 +100,8 @@ static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
   return bits;
 }
 
-static __inline int WebRtcSpl_NormW32(int32_t a) {
-  int zeros;
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
+  int16_t zeros;
 
   if (a == 0) {
     return 0;
@@ -123,8 +123,8 @@ static __inline int WebRtcSpl_NormW32(int32_t a) {
   return zeros;
 }
 
-static __inline int WebRtcSpl_NormU32(uint32_t a) {
-  int zeros;
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
+  int16_t zeros;
 
   if (a == 0) return 0;
 
@@ -141,8 +141,8 @@ static __inline int WebRtcSpl_NormU32(uint32_t a) {
   return zeros;
 }
 
-static __inline int WebRtcSpl_NormW16(int16_t a) {
-  int zeros;
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
+  int16_t zeros;
 
   if (a == 0) {
     return 0;
diff --git a/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h b/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
index 3854715d5..0d10129a2 100644
--- a/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
+++ b/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -83,7 +83,7 @@ static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
   return (int16_t)(32 - tmp);
 }
 
-static __inline int WebRtcSpl_NormW32(int32_t a) {
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
   int32_t tmp = 0;
 
   if (a == 0) {
@@ -95,20 +95,20 @@ static __inline int WebRtcSpl_NormW32(int32_t a) {
 
   __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
 
-  return tmp - 1;
+  return (int16_t)(tmp - 1);
 }
 
-static __inline int WebRtcSpl_NormU32(uint32_t a) {
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
   int tmp = 0;
 
   if (a == 0) return 0;
 
   __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
 
-  return tmp;
+  return (int16_t)tmp;
 }
 
-static __inline int WebRtcSpl_NormW16(int16_t a) {
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
   int32_t tmp = 0;
 
   if (a == 0) {
@@ -120,7 +120,7 @@ static __inline int WebRtcSpl_NormW16(int16_t a) {
 
   __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
 
-  return tmp - 17;
+  return (int16_t)(tmp - 17);
 }
 
 // TODO(kma): add unit test.
diff --git a/webrtc/common_audio/signal_processing/include/spl_inl_mips.h b/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
index ab9a60d47..cd04bddcf 100644
--- a/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
+++ b/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
@@ -137,10 +137,10 @@ static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
     : [n] "r" (n), [i32] "r" (i32)
   );
 
-  return bits;
+  return (int16_t)bits;
 }
 
-static __inline int WebRtcSpl_NormW32(int32_t a) {
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
   int zeros = 0;
 
   __asm __volatile(
@@ -160,10 +160,10 @@ static __inline int WebRtcSpl_NormW32(int32_t a) {
     : [a] "r" (a)
   );
 
-  return zeros;
+  return (int16_t)zeros;
 }
 
-static __inline int WebRtcSpl_NormU32(uint32_t a) {
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
   int zeros = 0;
 
   __asm __volatile(
@@ -172,10 +172,10 @@ static __inline int WebRtcSpl_NormU32(uint32_t a) {
     : [a] "r" (a)
   );
 
-  return (zeros & 0x1f);
+  return (int16_t)(zeros & 0x1f);
 }
 
-static __inline int WebRtcSpl_NormW16(int16_t a) {
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
   int zeros = 0;
   int a0 = a << 16;
 
@@ -196,7 +196,7 @@ static __inline int WebRtcSpl_NormW16(int16_t a) {
     : [a0] "r" (a0)
   );
 
-  return zeros;
+  return (int16_t)zeros;
 }
 
 static __inline int32_t WebRtc_MulAccumW16(int16_t a,
diff --git a/webrtc/common_audio/signal_processing/resample_by_2.c b/webrtc/common_audio/signal_processing/resample_by_2.c
index eab091ba4..9c0784edc 100644
--- a/webrtc/common_audio/signal_processing/resample_by_2.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2.c
@@ -67,10 +67,10 @@ static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
 
 // decimator
 #if !defined(MIPS32_LE)
-void WebRtcSpl_DownsampleBy2(const int16_t* in, int16_t len,
+void WebRtcSpl_DownsampleBy2(const int16_t* in, int len,
                              int16_t* out, int32_t* filtState) {
   int32_t tmp1, tmp2, diff, in32, out32;
-  int16_t i;
+  int i;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
@@ -125,10 +125,10 @@ void WebRtcSpl_DownsampleBy2(const int16_t* in, int16_t len,
 #endif  // #if defined(MIPS32_LE)
 
 
-void WebRtcSpl_UpsampleBy2(const int16_t* in, int16_t len,
+void WebRtcSpl_UpsampleBy2(const int16_t* in, int len,
                            int16_t* out, int32_t* filtState) {
   int32_t tmp1, tmp2, diff, in32, out32;
-  int16_t i;
+  int i;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
diff --git a/webrtc/common_audio/signal_processing/resample_by_2_mips.c b/webrtc/common_audio/signal_processing/resample_by_2_mips.c
index 8e840bb9a..6ffce551f 100644
--- a/webrtc/common_audio/signal_processing/resample_by_2_mips.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2_mips.c
@@ -29,11 +29,11 @@ static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
 
 // decimator
 void WebRtcSpl_DownsampleBy2(const int16_t* in,
-                             int16_t len,
+                             int len,
                              int16_t* out,
                              int32_t* filtState) {
   int32_t out32;
-  int16_t i, len1;
+  int i, len1;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 6a70a02ef..3fa3d2ee5 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -172,26 +172,18 @@ TEST_F(SplTest, BasicArrayOperationsTest) {
     for (int kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(3, b16[kk]);
     }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
+    WebRtcSpl_ZerosArrayW16(b16, kVectorSize);
     for (int kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(0, b16[kk]);
     }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(1, b16[kk]);
-    }
     WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
     for (int kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(3, b32[kk]);
     }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
+    WebRtcSpl_ZerosArrayW32(b32, kVectorSize);
     for (int kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(0, b32[kk]);
     }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(1, b32[kk]);
-    }
     for (int kk = 0; kk < kVectorSize; ++kk) {
         bTmp16[kk] = (int16_t)kk;
         bTmp32[kk] = (int32_t)kk;
@@ -204,7 +196,7 @@ TEST_F(SplTest, BasicArrayOperationsTest) {
 //    for (int kk = 0; kk < kVectorSize; ++kk) {
 //        EXPECT_EQ(b32[kk], bTmp32[kk]);
 //    }
-    EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
+    WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16);
     for (int kk = 0; kk < 2; ++kk) {
         EXPECT_EQ(kk+2, bTmp16[kk]);
     }
diff --git a/webrtc/common_audio/signal_processing/splitting_filter.c b/webrtc/common_audio/signal_processing/splitting_filter.c
index 966eef78b..4f6430c28 100644
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@@ -45,7 +45,7 @@ static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
 //                            |data_length|
 //
 
-void WebRtcSpl_AllPassQMF(int32_t* in_data, int16_t data_length,
+void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
                           int32_t* out_data, const uint16_t* filter_coefficients,
                           int32_t* filter_state)
 {
@@ -65,7 +65,7 @@ void WebRtcSpl_AllPassQMF(int32_t* in_data, int16_t data_length,
     // filter operation takes the |in_data| (which is the output from the previous cascade
     // filter) and store the output in |out_data|.
     // Note that the input vector values are changed during the process.
-    int16_t k;
+    int k;
     int32_t diff;
     // First all-pass cascade; filter from in_data to out_data.
 
diff --git a/webrtc/common_audio/signal_processing/vector_scaling_operations.c b/webrtc/common_audio/signal_processing/vector_scaling_operations.c
index d83920668..c3eaa2893 100644
--- a/webrtc/common_audio/signal_processing/vector_scaling_operations.c
+++ b/webrtc/common_audio/signal_processing/vector_scaling_operations.c
@@ -75,7 +75,7 @@ void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, int length,
       (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
     }
   } else {
-    int16_t left_shifts = -right_shifts;
+    int left_shifts = -right_shifts;
     for (i = length; i > 0; i--) {
       tmp_w32 = (*in++) << left_shifts;
       (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);