Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )

Reason for revert:
Can reland it if backwards compatible API is kept.

Original issue's description:
> Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
>
> Reason for revert:
> The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.
>
> Original issue's description:
> > Enable cpplint and fix cpplint errors in webrtc/*audio
> >
> > Change usage accordingly throughout the codebase
> >
> > BUG=webrtc:5268
> >
> > TESTED=Fixed issues reported by:
> > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
> >
> > Review-Url: https://codereview.webrtc.org/2683033004
> > Cr-Commit-Position: refs/heads/master@{#17133}
> > Committed: aebe55ca6c
>
> TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5268
>
> Review-Url: https://codereview.webrtc.org/2739143002
> Cr-Commit-Position: refs/heads/master@{#17138}
> Committed: e47c1d3ca1

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739073003
Cr-Commit-Position: refs/heads/master@{#17144}

PRESUBMIT.py

@@ -18,6 +18,7 @@ CPPLINT_DIRS = [
   'webrtc/api',
   'webrtc/audio',
   'webrtc/call',
+  'webrtc/common_audio',
   'webrtc/common_video',
   'webrtc/examples',
   'webrtc/modules/audio_mixer',

webrtc/audio/audio_transport_proxy.cc

@@ -52,7 +52,7 @@ int32_t AudioTransportProxy::RecordedDataIsAvailable(
     const int32_t clockDrift,
     const uint32_t currentMicLevel,
     const bool keyPressed,
-    uint32_t& newMicLevel) {
+    uint32_t& newMicLevel) {  // NOLINT: to avoid changing APIs
   // Pass call through to original audio transport instance.
   return voe_audio_transport_->RecordedDataIsAvailable(
       audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,

webrtc/audio/utility/audio_frame_operations.cc

@@ -280,32 +280,32 @@ void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
   }
 }
 
-int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
-  if (frame.num_channels_ != 2) {
+int AudioFrameOperations::Scale(float left, float right, AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
     return -1;
   }
 
-  for (size_t i = 0; i < frame.samples_per_channel_; i++) {
-    frame.data_[2 * i] = static_cast<int16_t>(left * frame.data_[2 * i]);
-    frame.data_[2 * i + 1] =
-        static_cast<int16_t>(right * frame.data_[2 * i + 1]);
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame->data_[2 * i] = static_cast<int16_t>(left * frame->data_[2 * i]);
+    frame->data_[2 * i + 1] =
+        static_cast<int16_t>(right * frame->data_[2 * i + 1]);
   }
   return 0;
 }
 
-int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
+int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame* frame) {
   int32_t temp_data = 0;
 
   // Ensure that the output result is saturated [-32768, +32767].
-  for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
        i++) {
-    temp_data = static_cast<int32_t>(scale * frame.data_[i]);
+    temp_data = static_cast<int32_t>(scale * frame->data_[i]);
     if (temp_data < -32768) {
-      frame.data_[i] = -32768;
+      frame->data_[i] = -32768;
     } else if (temp_data > 32767) {
-      frame.data_[i] = 32767;
+      frame->data_[i] = 32767;
     } else {
-      frame.data_[i] = static_cast<int16_t>(temp_data);
+      frame->data_[i] = static_cast<int16_t>(temp_data);
     }
   }
   return 0;

webrtc/audio/utility/audio_frame_operations.h

@@ -113,9 +113,19 @@ class AudioFrameOperations {
   // Halve samples in |frame|.
   static void ApplyHalfGain(AudioFrame* frame);
 
-  static int Scale(float left, float right, AudioFrame& frame);
+  static int Scale(float left, float right, AudioFrame* frame);
 
-  static int ScaleWithSat(float scale, AudioFrame& frame);
+  static int Scale(float left, float right, AudioFrame& frame) {  // NOLINT
+    // TODO(oprypin): drop this method
+    return Scale(left, right, &frame);
+  }
+
+  static int ScaleWithSat(float scale, AudioFrame* frame);
+
+  static int ScaleWithSat(float scale, AudioFrame& frame) {  // NOLINT
+    // TODO(oprypin): drop this method
+    return ScaleWithSat(scale, &frame);
+  }
 };
 
 }  // namespace webrtc

webrtc/audio/utility/audio_frame_operations_unittest.cc

@@ -426,20 +426,20 @@ TEST_F(AudioFrameOperationsTest, MuteEndStereoShort) {
 // TODO(andrew): should not allow negative scales.
 TEST_F(AudioFrameOperationsTest, DISABLED_ScaleFailsWithBadParameters) {
   frame_.num_channels_ = 1;
-  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
 
   frame_.num_channels_ = 3;
-  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
 
   frame_.num_channels_ = 2;
-  EXPECT_EQ(-1, AudioFrameOperations::Scale(-1.0, 1.0, frame_));
-  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, -1.0, frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(-1.0, 1.0, &frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, -1.0, &frame_));
 }
 
 // TODO(andrew): fix the wraparound bug. We should always saturate.
 TEST_F(AudioFrameOperationsTest, DISABLED_ScaleDoesNotWrapAround) {
   SetFrameData(4000, -4000, &frame_);
-  EXPECT_EQ(0, AudioFrameOperations::Scale(10.0, 10.0, frame_));
+  EXPECT_EQ(0, AudioFrameOperations::Scale(10.0, 10.0, &frame_));
 
   AudioFrame clipped_frame;
   clipped_frame.samples_per_channel_ = 320;
@@ -450,7 +450,7 @@ TEST_F(AudioFrameOperationsTest, DISABLED_ScaleDoesNotWrapAround) {
 
 TEST_F(AudioFrameOperationsTest, ScaleSucceeds) {
   SetFrameData(1, -1, &frame_);
-  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, frame_));
+  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, &frame_));
 
   AudioFrame scaled_frame;
   scaled_frame.samples_per_channel_ = 320;
@@ -461,13 +461,13 @@ TEST_F(AudioFrameOperationsTest, ScaleSucceeds) {
 
 // TODO(andrew): should fail with a negative scale.
 TEST_F(AudioFrameOperationsTest, DISABLED_ScaleWithSatFailsWithBadParameters) {
-  EXPECT_EQ(-1, AudioFrameOperations::ScaleWithSat(-1.0, frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::ScaleWithSat(-1.0, &frame_));
 }
 
 TEST_F(AudioFrameOperationsTest, ScaleWithSatDoesNotWrapAround) {
   frame_.num_channels_ = 1;
   SetFrameData(4000, &frame_);
-  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, frame_));
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
 
   AudioFrame clipped_frame;
   clipped_frame.samples_per_channel_ = 320;
@@ -476,7 +476,7 @@ TEST_F(AudioFrameOperationsTest, ScaleWithSatDoesNotWrapAround) {
   VerifyFramesAreEqual(clipped_frame, frame_);
 
   SetFrameData(-4000, &frame_);
-  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, frame_));
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
   SetFrameData(-32768, &clipped_frame);
   VerifyFramesAreEqual(clipped_frame, frame_);
 }
@@ -484,7 +484,7 @@ TEST_F(AudioFrameOperationsTest, ScaleWithSatDoesNotWrapAround) {
 TEST_F(AudioFrameOperationsTest, ScaleWithSatSucceeds) {
   frame_.num_channels_ = 1;
   SetFrameData(1, &frame_);
-  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, frame_));
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, &frame_));
 
   AudioFrame scaled_frame;
   scaled_frame.samples_per_channel_ = 320;

webrtc/common_audio/audio_converter.cc

@@ -107,7 +107,8 @@ class ResampleConverter : public AudioConverter {
 // converters must be provided.
 class CompositionConverter : public AudioConverter {
  public:
-  CompositionConverter(std::vector<std::unique_ptr<AudioConverter>> converters)
+  explicit CompositionConverter(
+    std::vector<std::unique_ptr<AudioConverter>> converters)
       : converters_(std::move(converters)) {
     RTC_CHECK_GE(converters_.size(), 2);
     // We need an intermediate buffer after every converter.

webrtc/common_audio/audio_converter.h

@@ -31,7 +31,7 @@ class AudioConverter {
                                                 size_t src_frames,
                                                 size_t dst_channels,
                                                 size_t dst_frames);
-  virtual ~AudioConverter() {};
+  virtual ~AudioConverter() {}
 
   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
   // capacity of |dst_capacity|. Both point to a series of buffers containing

webrtc/common_audio/blocker.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
-#define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
+#ifndef WEBRTC_COMMON_AUDIO_BLOCKER_H_
+#define WEBRTC_COMMON_AUDIO_BLOCKER_H_
 
 #include <memory>
 
@@ -124,4 +124,4 @@ class Blocker {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
+#endif  // WEBRTC_COMMON_AUDIO_BLOCKER_H_

webrtc/common_audio/channel_buffer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
+#ifndef WEBRTC_COMMON_AUDIO_CHANNEL_BUFFER_H_
+#define WEBRTC_COMMON_AUDIO_CHANNEL_BUFFER_H_
 
 #include <string.h>
 
@@ -183,4 +183,4 @@ class IFChannelBuffer {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
+#endif  // WEBRTC_COMMON_AUDIO_CHANNEL_BUFFER_H_

webrtc/common_audio/include/audio_util.h

@@ -11,6 +11,7 @@
 #ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 
+#include <algorithm>
 #include <limits>
 #include <cstring>
 

webrtc/common_audio/real_fourier.h

@@ -35,7 +35,7 @@ class RealFourier {
   // Construct a wrapper instance for the given input order, which must be
   // between 1 and kMaxFftOrder, inclusively.
   static std::unique_ptr<RealFourier> Create(int fft_order);
-  virtual ~RealFourier() {};
+  virtual ~RealFourier() {}
 
   // Helper to compute the smallest FFT order (a power of 2) which will contain
   // the given input length.

webrtc/common_audio/resampler/include/resampler.h

@@ -13,8 +13,8 @@
  * A wrapper for resampling a numerous amount of sampling combinations.
  */
 
-#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
-#define WEBRTC_RESAMPLER_RESAMPLER_H_
+#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
+#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
 
 #include <stddef.h>
 
@@ -23,10 +23,8 @@
 namespace webrtc {
 
 // All methods return 0 on success and -1 on failure.
-class Resampler
-{
-
-public:
+class Resampler {
+ public:
   Resampler();
   Resampler(int inFreq, int outFreq, size_t num_channels);
   ~Resampler();
@@ -39,11 +37,10 @@ public:
 
   // Resample samplesIn to samplesOut.
   int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
-             size_t maxLen, size_t &outLen);
+           size_t maxLen, size_t& outLen);  // NOLINT: to avoid changing APIs
 
-private:
-    enum ResamplerMode
-    {
+ private:
+  enum ResamplerMode {
     kResamplerMode1To1,
     kResamplerMode1To2,
     kResamplerMode1To3,
@@ -92,4 +89,4 @@ private:
 
 }  // namespace webrtc
 
-#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
+#endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_

webrtc/common_audio/resampler/push_resampler.cc

@@ -53,7 +53,7 @@ void CheckExpectedBufferSizes(size_t src_length,
   RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
 #endif
 }
-}
+}  // namespace
 
 template <typename T>
 PushResampler<T>::PushResampler()

webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc

@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
 #include <cmath>
 #include <cstring>
 #include <memory>

webrtc/common_audio/resampler/sinc_resampler_unittest.cc

@@ -16,6 +16,7 @@
 
 #include <math.h>
 
+#include <algorithm>
 #include <memory>
 
 #include "webrtc/base/timeutils.h"

webrtc/common_audio/ring_buffer_unittest.cc

@@ -58,7 +58,8 @@ static void RandomStressTest(int** data_ptr) {
   printf("seed=%u\n", seed);
   srand(seed);
   for (int i = 0; i < kNumTests; i++) {
-    const int buffer_size = std::max(rand() % kMaxBufferSize, 1);
+    // rand_r is not supported on many platforms, so rand is used.
+    const int buffer_size = std::max(rand() % kMaxBufferSize, 1);  // NOLINT
     std::unique_ptr<int[]> write_data(new int[buffer_size]);
     std::unique_ptr<int[]> read_data(new int[buffer_size]);
     scoped_ring_buffer buffer(WebRtc_CreateBuffer(buffer_size, sizeof(int)));
@@ -68,8 +69,8 @@ static void RandomStressTest(int** data_ptr) {
     int write_element = 0;
     int read_element = 0;
     for (int j = 0; j < kNumOps; j++) {
-      const bool write = rand() % 2 == 0 ? true : false;
-      const int num_elements = rand() % buffer_size;
+      const bool write = rand() % 2 == 0 ? true : false;  // NOLINT
+      const int num_elements = rand() % buffer_size;  // NOLINT
       if (write) {
         const int buffer_available = buffer_size - buffer_consumed;
         ASSERT_EQ(static_cast<size_t>(buffer_available),

webrtc/common_audio/signal_processing/include/signal_processing_library.h

@@ -15,8 +15,8 @@
  * For specific function calls, see bottom of file.
  */
 
-#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
 
 #include <string.h>
 #include "webrtc/common_audio/signal_processing/dot_product_with_scale.h"
@@ -77,7 +77,7 @@
 
 // C + the 32 most significant bits of A * B
 #define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
-    (C + (B >> 16) * A + (((uint32_t)(0x0000FFFF & B) * A) >> 16))
+    (C + (B >> 16) * A + (((uint32_t)(B & 0x0000FFFF) * A) >> 16))
 
 #define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
 
@@ -906,7 +906,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
 #ifdef __cplusplus
 }
 #endif  // __cplusplus
-#endif  // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
 
 //
 // WebRtcSpl_AddSatW16(...)

webrtc/common_audio/signal_processing/include/spl_inl.h

@@ -12,8 +12,8 @@
 // This header file includes the inline functions in
 // the fix point signal processing library.
 
-#ifndef WEBRTC_SPL_SPL_INL_H_
-#define WEBRTC_SPL_SPL_INL_H_
+#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
 
 #include "webrtc/system_wrappers/include/compile_assert_c.h"
 
@@ -56,7 +56,7 @@ static __inline int WebRtcSpl_CountLeadingZeros32(uint32_t n) {
 // Returns the number of leading zero bits in the argument.
 static __inline int WebRtcSpl_CountLeadingZeros64(uint64_t n) {
 #ifdef __GNUC__
-  COMPILE_ASSERT(sizeof(unsigned long long) == sizeof(uint64_t));
+  COMPILE_ASSERT(sizeof(unsigned long long) == sizeof(uint64_t));  // NOLINT
   return n == 0 ? 64 : __builtin_clzll(n);
 #else
   return WebRtcSpl_CountLeadingZeros64_NotBuiltin(n);
@@ -151,4 +151,4 @@ static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
 
 #endif  // WEBRTC_ARCH_ARM_V7
 
-#endif  // WEBRTC_SPL_SPL_INL_H_
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_

webrtc/common_audio/signal_processing/include/spl_inl_armv7.h

@@ -13,8 +13,8 @@
  * the fix point signal processing library.
  */
 
-#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
-#define WEBRTC_SPL_SPL_INL_ARMV7_H_
+#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
 
 /* TODO(kma): Replace some assembly code with GCC intrinsics
  * (e.g. __builtin_clz).
@@ -88,8 +88,7 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
 
   if (a == 0) {
     return 0;
-  }
-  else if (a < 0) {
+  } else if (a < 0) {
     a ^= 0xFFFFFFFF;
   }
 
@@ -114,8 +113,7 @@ static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
 
   if (a_32 == 0) {
     return 0;
-  }
-  else if (a_32 < 0) {
+  } else if (a_32 < 0) {
     a_32 ^= 0xFFFFFFFF;
   }
 
@@ -133,4 +131,4 @@ static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
   return (int16_t)out;
 }
 
-#endif  // WEBRTC_SPL_SPL_INL_ARMV7_H_
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_

webrtc/common_audio/signal_processing/include/spl_inl_mips.h

@@ -12,8 +12,8 @@
 // This header file includes the inline functions in
 // the fix point signal processing library.
 
-#ifndef WEBRTC_SPL_SPL_INL_MIPS_H_
-#define WEBRTC_SPL_SPL_INL_MIPS_H_
+#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
 
 static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a,
                                              int32_t b) {
@@ -33,8 +33,7 @@ static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a,
     "mul    %[value32],     %[a1],  %[b1]       \n\t"
     : [value32] "=r" (value32), [a1] "=&r" (a1), [b1] "=&r" (b1)
     : [a] "r" (a), [b] "r" (b)
-    : "hi", "lo"
-  );
+    : "hi", "lo");
   return value32;
 }
 
@@ -61,8 +60,7 @@ static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a,
     : [value32] "=&r" (value32), [b1] "=&r" (b1), [b2] "=&r" (b2),
       [a1] "=&r" (a1)
     : [a] "r" (a), [b] "r" (b)
-    : "hi", "lo"
-  );
+    : "hi", "lo");
   return value32;
 }
 
@@ -72,8 +70,7 @@ static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
     "shll_s.w   %[value32], %[value32], 16      \n\t"
     "sra        %[value32], %[value32], 16      \n\t"
     : [value32] "+r" (value32)
-    :
-  );
+    :);
   int16_t out16 = (int16_t)value32;
   return out16;
 }
@@ -84,8 +81,7 @@ static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
   __asm __volatile(
     "addq_s.ph      %[value32],     %[a],   %[b]    \n\t"
     : [value32] "=r" (value32)
-    : [a] "r" (a), [b] "r" (b)
-  );
+    : [a] "r" (a), [b] "r" (b) );
   return (int16_t)value32;
 }
 
@@ -95,8 +91,7 @@ static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
   __asm __volatile(
     "addq_s.w   %[l_sum],       %[l_var1],      %[l_var2]    \n\t"
     : [l_sum] "=r" (l_sum)
-    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2)
-  );
+    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2) );
 
   return l_sum;
 }
@@ -107,8 +102,7 @@ static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
   __asm __volatile(
     "subq_s.ph  %[value32], %[var1],    %[var2]     \n\t"
     : [value32] "=r" (value32)
-    : [var1] "r" (var1), [var2] "r" (var2)
-  );
+    : [var1] "r" (var1), [var2] "r" (var2) );
 
   return (int16_t)value32;
 }
@@ -119,8 +113,7 @@ static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
   __asm __volatile(
     "subq_s.w   %[l_diff],      %[l_var1],      %[l_var2]    \n\t"
     : [l_diff] "=r" (l_diff)
-    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2)
-  );
+    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2) );
 
   return l_diff;
 }
@@ -134,8 +127,7 @@ static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
     "clz    %[bits],    %[n]                    \n\t"
     "subu   %[bits],    %[i32],     %[bits]     \n\t"
     : [bits] "=&r" (bits)
-    : [n] "r" (n), [i32] "r" (i32)
-  );
+    : [n] "r" (n), [i32] "r" (i32) );
 
   return (int16_t)bits;
 }
@@ -157,8 +149,7 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
    "2:                                              \n\t"
     ".set       pop                                 \n\t"
     : [zeros]"=&r"(zeros)
-    : [a] "r" (a)
-  );
+    : [a] "r" (a) );
 
   return (int16_t)zeros;
 }
@@ -169,8 +160,7 @@ static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
   __asm __volatile(
     "clz    %[zeros],   %[a]    \n\t"
     : [zeros] "=r" (zeros)
-    : [a] "r" (a)
-  );
+    : [a] "r" (a) );
 
   return (int16_t)(zeros & 0x1f);
 }
@@ -193,8 +183,7 @@ static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
    "2:                                              \n\t"
     ".set       pop                                 \n\t"
     : [zeros]"=&r"(zeros)
-    : [a0] "r" (a0)
-  );
+    : [a0] "r" (a0) );
 
   return (int16_t)zeros;
 }
@@ -217,9 +206,8 @@ static __inline int32_t WebRtc_MulAccumW16(int16_t a,
     "addu   %[c1],      %[c],   %[res]  \n\t"
     : [c1] "=r" (c1), [res] "=&r" (res)
     : [a] "r" (a), [b] "r" (b), [c] "r" (c)
-    : "hi", "lo"
-  );
+    : "hi", "lo");
   return (c1);
 }
 
-#endif  // WEBRTC_SPL_SPL_INL_MIPS_H_
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_

webrtc/common_audio/signal_processing/real_fft_unittest.cc

@@ -60,7 +60,7 @@ TEST_F(RealFFTTest, RealAndComplexMatch) {
   for (i = 0, j = 0; i < kTimeDataLength; i += 1, j += 2) {
     complex_fft_buff[j] = kRefData[i];
     complex_fft_buff[j + 1] = 0;  // Insert zero's to imaginary parts.
-  };
+  }
 
   // Create and run real forward FFT.
   RealFFT* fft = WebRtcSpl_CreateRealFFT(kOrder);

webrtc/common_audio/signal_processing/resample_by_2_internal.h

@@ -14,8 +14,8 @@
  *
  */
 
-#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
 
 #include "webrtc/typedefs.h"
 
@@ -44,4 +44,4 @@ void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len,
 void WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
                              int32_t* state);
 
-#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_

webrtc/common_audio/signal_processing/signal_processing_unittest.cc

@@ -328,7 +328,8 @@ TEST_F(SplTest, VectorOperationsTest) {
     for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
     }
-    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
+    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16,
+                                          kVectorSize);
     for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
     }
@@ -355,7 +356,8 @@ TEST_F(SplTest, VectorOperationsTest) {
     for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
     }
-    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
+    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3],
+                                            kVectorSize, 2);
     for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
     }
@@ -558,7 +560,7 @@ TEST_F(SplTest, FFTTest) {
 //    }
     WebRtcSpl_ComplexBitReverse(B, 3);
     for (int kk = 0; kk < 16; ++kk) {
-        //EXPECT_EQ(A[kk], B[kk]);
+//      EXPECT_EQ(A[kk], B[kk]);
     }
 }
 

webrtc/common_audio/vad/vad_core.h

@@ -24,9 +24,7 @@ enum { kNumGaussians = 2 };  // Number of Gaussians per channel in the GMM.
 enum { kTableSize = kNumChannels * kNumGaussians };
 enum { kMinEnergy = 10 };  // Minimum energy required to trigger audio signal.
 
-typedef struct VadInstT_
-{
-
+typedef struct VadInstT_ {
     int vad;
     int32_t downsampling_filter_states[4];
     WebRtcSpl_State48khzTo8khz state_48_to_8;
@@ -52,7 +50,6 @@ typedef struct VadInstT_
     int16_t total[3];
 
     int init_flag;
-
 } VadInstT;
 
 // Initializes the core VAD component. The default aggressiveness mode is

webrtc/common_audio/vad/vad_core_unittest.cc

@@ -18,7 +18,8 @@ extern "C" {
 #include "webrtc/common_audio/vad/vad_core.h"
 }
 
-namespace {
+namespace webrtc {
+namespace test {
 
 TEST_F(VadTest, InitCore) {
   // Test WebRtcVad_InitCore().
@@ -102,4 +103,5 @@ TEST_F(VadTest, CalcVad) {
 
   free(self);
 }
-}  // namespace
+}  // namespace test
+}  // namespace webrtc

webrtc/common_audio/vad/vad_filterbank_unittest.cc

@@ -19,7 +19,8 @@ extern "C" {
 #include "webrtc/common_audio/vad/vad_filterbank.h"
 }
 
-namespace {
+namespace webrtc {
+namespace test {
 
 const int kNumValidFrameLengths = 3;
 
@@ -89,4 +90,5 @@ TEST_F(VadTest, vad_filterbank) {
 
   free(self);
 }
-}  // namespace
+}  // namespace test
+}  // namespace webrtc

webrtc/common_audio/vad/vad_gmm_unittest.cc

@@ -16,7 +16,8 @@ extern "C" {
 #include "webrtc/common_audio/vad/vad_gmm.h"
 }
 
-namespace {
+namespace webrtc {
+namespace test {
 
 TEST_F(VadTest, vad_gmm) {
   int16_t delta = 0;
@@ -40,4 +41,5 @@ TEST_F(VadTest, vad_gmm) {
   EXPECT_EQ(0, WebRtcVad_GaussianProbability(105, 0, 128, &delta));
   EXPECT_EQ(13440, delta);
 }
-}  // namespace
+}  // namespace test
+}  // namespace webrtc

webrtc/common_audio/vad/vad_sp_unittest.cc

@@ -19,7 +19,8 @@ extern "C" {
 #include "webrtc/common_audio/vad/vad_sp.h"
 }
 
-namespace {
+namespace webrtc {
+namespace test {
 
 TEST_F(VadTest, vad_sp) {
   VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
@@ -71,4 +72,5 @@ TEST_F(VadTest, vad_sp) {
 
   free(self);
 }
-}  // namespace
+}  // namespace test
+}  // namespace webrtc

webrtc/common_audio/vad/vad_unittest.cc

@@ -52,7 +52,8 @@ bool VadTest::ValidRatesAndFrameLengths(int rate, size_t frame_length) {
   return false;
 }
 
-namespace {
+namespace webrtc {
+namespace test {
 
 TEST_F(VadTest, ApiTest) {
   // This API test runs through the APIs for all possible valid and invalid
@@ -152,4 +153,5 @@ TEST_F(VadTest, ValidRatesFrameLengths) {
 
 // TODO(bjornv): Add a process test, run on file.
 
-}  // namespace
+}  // namespace test
+}  // namespace webrtc

webrtc/common_audio/vad/vad_unittest.h

@@ -8,15 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
-#define WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H_
 
 #include <stddef.h>  // size_t
 
 #include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace test {
 
 // Modes we support
 const int kModes[] = { 0, 1, 2, 3 };
@@ -32,7 +33,8 @@ const size_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640, 960,
     kMaxFrameLength };
 const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
 
-}  // namespace
+}  // namespace test
+}  // namespace webrtc
 
 class VadTest : public ::testing::Test {
  protected:
@@ -44,4 +46,4 @@ class VadTest : public ::testing::Test {
   bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
 };
 
-#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H_

webrtc/voice_engine/channel.cc

@@ -675,7 +675,7 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
   // Output volume scaling
   if (output_gain < 0.99f || output_gain > 1.01f) {
     // TODO(solenberg): Combine with mute state - this can cause clicks!
-    AudioFrameOperations::ScaleWithSat(output_gain, *audioFrame);
+    AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
   }
 
   // Mix decoded PCM output with file if file mixing is enabled