Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.

This CL enableds cpplint for webrtc/common_video, webrtc/sound and webrtc/tools.

BUG=webrtc:5273

TESTED=Fixed issues reported by:
find webrtc/common_video webrtc/sound webrtc/tools -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

Review URL: https://codereview.webrtc.org/1511603004

Cr-Commit-Position: refs/heads/master@{#10967}

PRESUBMIT.py

@@ -18,8 +18,11 @@ import sys
 CPPLINT_DIRS = [
   'webrtc/audio',
   'webrtc/call',
+  'webrtc/common_video',
   'webrtc/examples',
   'webrtc/modules/video_processing',
+  'webrtc/sound',
+  'webrtc/tools',
   'webrtc/video',
 ]
 

webrtc/common_video/i420_video_frame_unittest.cc

@@ -8,8 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_frame.h"
-
 #include <math.h>
 #include <string.h>
 
@@ -17,6 +15,7 @@
 #include "webrtc/base/bind.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -174,7 +173,7 @@ TEST(TestVideoFrame, ShallowCopy) {
 
 TEST(TestVideoFrame, Reset) {
   VideoFrame frame;
-  ASSERT_TRUE(frame.CreateEmptyFrame(5, 5, 5, 5, 5) == 0);
+  ASSERT_EQ(frame.CreateEmptyFrame(5, 5, 5, 5, 5), 0);
   frame.set_ntp_time_ms(1);
   frame.set_timestamp(2);
   frame.set_render_time_ms(3);

webrtc/common_video/libyuv/include/webrtc_libyuv.h

@@ -152,6 +152,7 @@ int ConvertNV12ToRGB565(const uint8_t* src_frame,
 double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame);
 // Compute SSIM for an I420 frame (all planes).
 double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame);
-}
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_

webrtc/common_video/libyuv/libyuv_unittest.cc

@@ -346,4 +346,4 @@ TEST_F(TestLibYuv, StrideAlignment) {
   EXPECT_EQ(64, stride_uv);
 }
 
-}  // namespace
+}  // namespace webrtc

webrtc/sound/alsasoundsystem.cc

@@ -11,6 +11,7 @@
 #include "webrtc/sound/alsasoundsystem.h"
 
 #include <algorithm>
+#include <string>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
@@ -259,12 +260,12 @@ class AlsaInputStream :
   }
 
   bool GetVolume(int *volume) override {
-    // TODO: Implement this.
+    // TODO(henrika): Implement this.
     return false;
   }
 
   bool SetVolume(int volume) override {
-    // TODO: Implement this.
+    // TODO(henrika): Implement this.
     return false;
   }
 
@@ -390,12 +391,12 @@ class AlsaOutputStream : public SoundOutputStreamInterface,
   }
 
   bool GetVolume(int *volume) override {
-    // TODO: Implement this.
+    // TODO(henrika): Implement this.
     return false;
   }
 
   bool SetVolume(int volume) override {
-    // TODO: Implement this.
+    // TODO(henrika): Implement this.
     return false;
   }
 
@@ -569,7 +570,6 @@ bool AlsaSoundSystem::EnumerateDevices(
         strcmp(name, ignore_null) != 0 &&
         strcmp(name, ignore_pulse) != 0 &&
         !rtc::starts_with(name, ignore_prefix)) {
-
       // Yes, we do.
       char *desc = symbol_table_.snd_device_name_get_hint()(*list, "DESC");
       if (!desc) {
@@ -622,7 +622,6 @@ StreamInterface *AlsaSoundSystem::OpenDevice(
         int wait_timeout_ms,
         int flags,
         int freq)) {
-
   if (!IsInitialized()) {
     return NULL;
   }

webrtc/sound/pulseaudiosoundsystem.cc

@@ -13,6 +13,7 @@
 #ifdef HAVE_LIBPULSE
 
 #include <algorithm>
+#include <string>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
@@ -216,17 +217,6 @@ class PulseAudioStream {
 class PulseAudioInputStream :
     public SoundInputStreamInterface,
     private rtc::Worker {
-
-  struct GetVolumeCallbackData {
-    PulseAudioInputStream *instance;
-    pa_cvolume *channel_volumes;
-  };
-
-  struct GetSourceChannelCountCallbackData {
-    PulseAudioInputStream *instance;
-    uint8_t *channels;
-  };
-
  public:
   PulseAudioInputStream(PulseAudioSoundSystem *pulse,
                         pa_stream *stream,
@@ -386,6 +376,16 @@ class PulseAudioInputStream :
   }
 
  private:
+  struct GetVolumeCallbackData {
+    PulseAudioInputStream* instance;
+    pa_cvolume* channel_volumes;
+  };
+
+  struct GetSourceChannelCountCallbackData {
+    PulseAudioInputStream* instance;
+    uint8_t* channels;
+  };
+
   void Lock() {
     stream_.Lock();
   }
@@ -580,12 +580,6 @@ class PulseAudioInputStream :
 class PulseAudioOutputStream :
     public SoundOutputStreamInterface,
     private rtc::Worker {
-
-  struct GetVolumeCallbackData {
-    PulseAudioOutputStream *instance;
-    pa_cvolume *channel_volumes;
-  };
-
  public:
   PulseAudioOutputStream(PulseAudioSoundSystem *pulse,
                          pa_stream *stream,
@@ -733,7 +727,7 @@ class PulseAudioOutputStream :
   }
 
 #if 0
-  // TODO: Versions 0.9.16 and later of Pulse have a new API for
+  // TODO(henrika): Versions 0.9.16 and later of Pulse have a new API for
   // zero-copy writes, but Hardy is not new enough to have that so we can't
   // rely on it. Perhaps auto-detect if it's present or not and use it if we
   // can?
@@ -777,6 +771,11 @@ class PulseAudioOutputStream :
 #endif
 
  private:
+  struct GetVolumeCallbackData {
+    PulseAudioOutputStream* instance;
+    pa_cvolume* channel_volumes;
+  };
+
   void Lock() {
     stream_.Lock();
   }
@@ -1165,7 +1164,7 @@ bool PulseAudioSoundSystem::ConnectToPulse(pa_context *context) {
 pa_context *PulseAudioSoundSystem::CreateNewConnection() {
   // Create connection context.
   std::string app_name;
-  // TODO: Pulse etiquette says this name should be localized. Do
+  // TODO(henrika): Pulse etiquette says this name should be localized. Do
   // we care?
   rtc::Filesystem::GetApplicationName(&app_name);
   pa_context *context = symbol_table_.pa_context_new()(

webrtc/sound/soundinputstreaminterface.h

@@ -65,4 +65,4 @@ class SoundInputStreamInterface {
 
 }  // namespace rtc
 
-#endif  // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
+#endif  // WEBRTC_SOUND_SOUNDINPUTSTREAMINTERFACE_H_

webrtc/tools/agc/activity_metric.cc

@@ -65,7 +65,7 @@ static void DitherSilence(AudioFrame* frame) {
     sum_squared += frame->data_[n] * frame->data_[n];
   if (sum_squared <= sum_squared_silence) {
     for (size_t n = 0; n < frame->samples_per_channel_; n++)
-      frame->data_[n] = (rand() & 0xF) - 8;
+      frame->data_[n] = (rand() & 0xF) - 8;  // NOLINT: ignore non-threadsafe.
   }
 }
 

webrtc/tools/frame_editing/frame_editing_lib.cc

@@ -15,6 +15,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/tools/frame_editing/frame_editing_lib.h"
 #include "webrtc/typedefs.h"
 
 using std::string;

webrtc/tools/frame_editing/frame_editing_lib.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
-#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
+#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
+#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
 
 #include <string>
 
@@ -36,4 +36,4 @@ int EditFrames(const std::string& in_path, int width, int height,
                 int last_frame_to_process, const std::string& out_path);
 }  // namespace webrtc
 
-#endif  // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
+#endif  // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_