Proper spacing for end-of-namespace comments.

BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_audio/resampler/include/resampler.h

@@ -111,6 +111,6 @@ private:
     Resampler* slave_right_;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_RESAMPLER_RESAMPLER_H_

webrtc/common_audio/resampler/resampler.cc

@@ -1081,4 +1081,4 @@ int Resampler::Pull(int16_t* samplesOut, int desiredLen, int &outLen)
     }
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/common_video/interface/video_image.h

@@ -64,6 +64,6 @@ public:
     bool                        _completeFrame;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H

webrtc/common_video/jpeg/data_manager.cc

@@ -83,4 +83,4 @@ termSource (j_decompress_ptr cinfo)
   //
 }
 
-} // end of namespace webrtc
+}  // namespace webrtc

webrtc/common_video/jpeg/data_manager.h

@@ -62,7 +62,7 @@ skipInputData(j_decompress_ptr cinfo, long num_bytes);
 void
 termSource (j_decompress_ptr cinfo);
 
-} // end of namespace webrtc
+}  // namespace webrtc
 
 
 #endif /* WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER */

webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h

@@ -33,6 +33,6 @@ class ACMDTMFDetection {
   ACMResampler resampler_;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_

webrtc/modules/audio_coding/main/source/acm_isac_macros.h

@@ -67,7 +67,7 @@ namespace webrtc {
 #define ACM_ISAC_GETDECSAMPRATE    ACMISACFixGetDecSampRate  // local Impl
 #endif
 
-}  //namespace
+}  // namespace
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
 

webrtc/modules/audio_coding/neteq/test/NetEqRTPplay.cc

@@ -1269,7 +1269,7 @@ void parsePtypeFile(FILE *ptypeFile, std::map<uint8_t, decoderStruct>* decoders)
         }
 
         n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
-    } // end while
+    }  // end while
 
 }
 

webrtc/modules/audio_coding/neteq/test/RTPencode.cc

@@ -1571,7 +1571,7 @@ int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int
         if (ok != 0) {
             return(ok);
         }
-    } // end for
+    }  // end for
 
     return(0);
 }			
@@ -1974,7 +1974,7 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
         encoded += cdlen;
         totalLen += cdlen;
 
-    } // end for
+    }  // end for
 
 	first_cng=1;
 	return(totalLen);

webrtc/modules/audio_coding/neteq4/test/RTPencode.cc

@@ -1383,7 +1383,7 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bit
         if (ok != 0) {
             return(ok);
         }
-    } // end for
+    }  // end for
 
     return(0);
 }			
@@ -1669,7 +1669,7 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
         encoded += cdlen;
         totalLen += cdlen;
 
-    } // end for
+    }  // end for
 
 	first_cng=1;
 	return(totalLen);

webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h

@@ -80,6 +80,6 @@ public:
 protected:
     AudioConferenceMixer() {}
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_

webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h

@@ -100,6 +100,6 @@ protected:
     AudioRelayReceiver() {}
     virtual ~AudioRelayReceiver() {}
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

@@ -1217,4 +1217,4 @@ bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio)
     }
     return true;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h

@@ -203,6 +203,6 @@ private:
     // Used for inhibiting saturation in mixing.
     scoped_ptr<AudioProcessing> _limiter;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc

@@ -36,7 +36,7 @@ const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
                            0.9114f, 0.9241f, 0.9367f, 0.9494f,
                            0.9620f, 0.9747f, 0.9873f, 1.0000f};
 const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
-} // namespace
+}  // namespace
 
 namespace webrtc {
 void CalculateEnergy(AudioFrame& audioFrame)
@@ -78,4 +78,4 @@ void RampOut(AudioFrame& audioFrame)
            (audioFrame.samples_per_channel_ - rampSize) *
            sizeof(audioFrame.data_[0]));
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h

@@ -21,6 +21,6 @@ void CalculateEnergy(AudioFrame& audioFrame);
 void RampIn(AudioFrame& audioFrame);
 void RampOut(AudioFrame& audioFrame);
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_

webrtc/modules/audio_conference_mixer/source/level_indicator.cc

@@ -73,4 +73,4 @@ int32_t LevelIndicator::GetLevel()
     return _currentLevel;
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_conference_mixer/source/level_indicator.h

@@ -32,6 +32,6 @@ private:
     uint32_t _count;
     uint32_t _currentLevel;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_

webrtc/modules/audio_conference_mixer/source/memory_pool.h

@@ -117,6 +117,6 @@ int32_t MemoryPool<MemoryType>::PushMemory(MemoryType*& memory)
     }
     return _ptrImpl->PushMemory(memory);
 }
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_

webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h

@@ -163,6 +163,6 @@ int32_t MemoryPoolImpl<MemoryType>::CreateMemory(
     }
     return 0;
 }
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_

webrtc/modules/audio_conference_mixer/source/memory_pool_win.h

@@ -194,6 +194,6 @@ MemoryPoolItem<MemoryType>* MemoryPoolImpl<MemoryType>::CreateMemory()
     ++_createdMemory;
     return returnValue;
 }
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_

webrtc/modules/audio_conference_mixer/source/time_scheduler.cc

@@ -99,4 +99,4 @@ int32_t TimeScheduler::TimeToNextUpdate(
     updateTimeInMS =  (updateTimeInMS < 0) ? 0 : updateTimeInMS;
     return 0;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_conference_mixer/source/time_scheduler.h

@@ -42,6 +42,6 @@ private:
     int64_t  _periodicityInTicks;
     uint32_t _missedPeriods;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_

webrtc/modules/audio_device/android/audio_device_jni_android.cc

@@ -2730,7 +2730,7 @@ bool AudioDeviceAndroidJni::PlayThreadProcess()
 
         Lock();
 
-    } // _playing
+    }  // _playing
 
     if (_shutdownPlayThread)
     {
@@ -2867,7 +2867,7 @@ bool AudioDeviceAndroidJni::RecThreadProcess()
             Lock();
         }
 
-    } // _recording
+    }  // _recording
 
     if (_shutdownRecThread)
     {
@@ -2898,4 +2898,4 @@ bool AudioDeviceAndroidJni::RecThreadProcess()
     return true;
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/android/audio_device_jni_android.h

@@ -264,6 +264,6 @@ class AudioDeviceAndroidJni : public AudioDeviceGeneric {
   static jclass globalScClass;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_JNI_ANDROID_H

webrtc/modules/audio_device/android/audio_device_utility_android.cc

@@ -47,4 +47,4 @@ int32_t AudioDeviceUtilityAndroid::Init()
     return 0;
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/android/audio_device_utility_android.h

@@ -35,6 +35,6 @@ private:
     int32_t _id;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H

webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h

@@ -27,6 +27,6 @@ public:
     virtual int32_t Init() { return 0; }
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_

webrtc/modules/audio_device/ios/audio_device_utility_ios.cc

@@ -42,4 +42,4 @@ int32_t AudioDeviceUtilityIPhone::Init() {
     return 0;
 }
 
-}  //  namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/ios/audio_device_utility_ios.h

@@ -31,6 +31,6 @@ private:
     AudioDeviceModule::ErrorCode _lastError;
 };
 
-}  //  namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H

webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc

@@ -2954,7 +2954,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
         EnableWriteCallback();
         PaUnLock();
 
-    } // _playing
+    }  // _playing
 
     UnLock();
     return true;
@@ -3103,7 +3103,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
         EnableReadCallback();
         PaUnLock();
 
-    } // _recording
+    }  // _recording
 
     UnLock();
     return true;

webrtc/modules/audio_device/linux/audio_device_utility_linux.cc

@@ -51,4 +51,4 @@ int32_t AudioDeviceUtilityLinux::Init()
 }
 
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/linux/audio_device_utility_linux.h

@@ -31,6 +31,6 @@ private:
     int32_t _id;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_

webrtc/modules/audio_device/mac/audio_device_mac.cc

@@ -2271,7 +2271,7 @@ AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
 
             free(bufferList);
             bufferList = NULL;
-        } // for
+        }  // for
     }
 
     if (!listOK)
@@ -3269,4 +3269,4 @@ bool AudioDeviceMac::KeyPressed() const{
   }
   return(key_down);
 }
-} //  namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/mac/audio_device_mac.h

@@ -379,6 +379,6 @@ private:
     int _renderBufSizeSamples;
 };
 
-} //  namespace webrtc
+}  // namespace webrtc
 
 #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_

webrtc/modules/audio_device/mac/audio_device_utility_mac.cc

@@ -50,4 +50,4 @@ int32_t AudioDeviceUtilityMac::Init()
     return 0;
 }
 
-} //  namespace webrtc
+}  // namespace webrtc

webrtc/modules/audio_device/mac/audio_device_utility_mac.h

@@ -31,6 +31,6 @@ private:
     int32_t _id;
 };
 
-} //  namespace webrtc
+}  // namespace webrtc
 
 #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_

webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc

@@ -1163,5 +1163,5 @@ void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
 #endif
 }
 
-} // namespace webrtc
+}  // namespace webrtc
 // EOF

webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h

@@ -75,6 +75,6 @@ private:
 
 };
 	
-} //namespace webrtc
+}  // namespace webrtc
 
 #endif  // AUDIO_MIXER_MAC_H

webrtc/modules/audio_device/test/func_test_manager.cc

@@ -443,7 +443,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
             }
             _audioList.PopFront();
         }
-    } // if (_fullDuplex)
+    }  // if (_fullDuplex)
 
     if (_playFromFile && _playFile.Open())
     {
@@ -475,7 +475,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                 audio16++;
             }
         }
-    } // if (_playFromFile && _playFile.Open())
+    }  // if (_playFromFile && _playFile.Open())
 
     _playCount++;
 
@@ -550,7 +550,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
         {
             TEST_LOG("++");
         }
-    } // if (_playCount % 100 == 0)
+    }  // if (_playCount % 100 == 0)
 
     nSamplesOut = nSamples;
 
@@ -880,7 +880,7 @@ int32_t FuncTestManager::TestAudioLayerSelection()
                 TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio <=> "
                     "switch was possible\n \n");
         }
-    } // if (tryWinWave || tryWinCore)
+    }  // if (tryWinWave || tryWinCore)
 
     PRINT_TEST_RESULTS;
 
@@ -2460,7 +2460,7 @@ int32_t FuncTestManager::TestDeviceRemoval()
 
             loopCount++;
         }
-    } // loopCount
+    }  // loopCount
 
     EXPECT_EQ(0, audioDevice->Terminate());
     EXPECT_FALSE(audioDevice->Initialized());
@@ -2729,6 +2729,6 @@ int32_t FuncTestManager::TestAdvancedMBAPI()
     return 0;
 }
 
-} // namespace webrtc
+}  // namespace webrtc
 
 // EOF

webrtc/modules/audio_device/test/func_test_manager.h

@@ -218,6 +218,6 @@ private:
     AudioTransportImpl* _audioTransport;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H

webrtc/modules/audio_device/win/audio_device_core_win.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#pragma warning(disable: 4995)  //  name was marked as #pragma deprecated
+#pragma warning(disable: 4995)  // name was marked as #pragma deprecated
 
 #if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
 // Reports the major and minor versions of the compiler.

webrtc/modules/audio_processing/ns/ns_core.c

@@ -276,7 +276,7 @@ void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) {
         inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
             + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s] + 1);
       }
-    } // end loop over magnitude spectrum
+    }  // end loop over magnitude spectrum
 
     if (inst->counter[s] >= END_STARTUP_LONG) {
       inst->counter[s] = 0;
@@ -288,7 +288,7 @@ void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) {
     }
 
     inst->counter[s]++;
-  } // end loop over simultaneous estimates
+  }  // end loop over simultaneous estimates
 
   // Sequentially update the noise during startup
   if (inst->updates < END_STARTUP_LONG) {
@@ -522,7 +522,7 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) {
         inst->histSpecDiff[i] = 0;
       }
     }
-  } // end of flag == 1
+  }  // end of flag == 1
 }
 
 // Compute spectral flatness on input spectrum
@@ -855,7 +855,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
           }
           outFrameHB[i] = (short)dTmp;
         }
-      } // end of H band gain computation
+      }  // end of H band gain computation
       //
       return 0;
     }
@@ -1006,14 +1006,14 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
       snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
                        * snrLocPost[i];
       // post and prior snr needed for step 2
-    } // end of loop over freqs
+    }  // end of loop over freqs
 #ifdef PROCESS_FLOW_1
     for (i = 0; i < inst->magnLen; i++) {
       // gain filter
       tmpFloat1 = inst->overdrive + snrLocPrior[i];
       tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
       theFilter[i] = (float)tmpFloat2;
-    } // end of loop over freqs
+    }  // end of loop over freqs
 #endif
     // done with step 1: dd computation of prior and post snr
 
@@ -1085,7 +1085,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
           noise[i] = noiseUpdateTmp;
         }
       }
-    } // end of freq loop
+    }  // end of freq loop
     // done with step 2: noise update
 
     //
@@ -1105,7 +1105,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
       tmpFloat1 = inst->overdrive + snrPrior;
       tmpFloat2 = (float)snrPrior / tmpFloat1;
       theFilter[i] = (float)tmpFloat2;
-    } // end of loop over freqs
+    }  // end of loop over freqs
     // done with step3
 #endif
 #endif
@@ -1204,7 +1204,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
         factor = (float)1.0 / gain;
       }
 #endif
-    } // out of inst->gainmap==1
+    }  // out of inst->gainmap==1
 
     // synthesis
     for (i = 0; i < inst->anaLen; i++) {
@@ -1227,7 +1227,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
         inst->outBuf[i] = fout[i + inst->blockLen10ms];
       }
     }
-  } // end of if out.len==0
+  }  // end of if out.len==0
   else {
     for (i = 0; i < inst->blockLen10ms; i++) {
       fout[i] = inst->outBuf[i];
@@ -1297,7 +1297,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
       }
       outFrameHB[i] = (short)dTmp;
     }
-  } // end of H band gain computation
+  }  // end of H band gain computation
   //
 
   return 0;

webrtc/modules/audio_processing/ns/nsx_core.c

@@ -588,7 +588,7 @@ static void NoiseEstimationC(NsxInst_t* inst,
                      width_factor, countDiv, 15);
         inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
       }
-    } // end loop over magnitude spectrum
+    }  // end loop over magnitude spectrum
 
     if (counter >= END_STARTUP_LONG) {
       inst->noiseEstCounter[s] = 0;
@@ -598,7 +598,7 @@ static void NoiseEstimationC(NsxInst_t* inst,
     }
     inst->noiseEstCounter[s]++;
 
-  } // end loop over simultaneous estimates
+  }  // end loop over simultaneous estimates
 
   // Sequentially update the noise during startup
   if (inst->blockIndex < END_STARTUP_LONG) {
@@ -1146,7 +1146,7 @@ void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
     WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
     WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
     WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
-  } // end of flag == 1
+  }  // end of flag == 1
 }
 
 
@@ -1874,7 +1874,7 @@ void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) {
     tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
                                                         gainFactor2, 14); // Q13;
     gainFactor = tmp16no1 + tmp16no2; // Q13
-  } // out of flag_gain_map==1
+  }  // out of flag_gain_map==1
 
   // Synthesis, read out fully processed segment, and update synthesis buffer.
   WebRtcNsx_SynthesisUpdate(inst, outFrame, gainFactor);
@@ -1960,7 +1960,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
       for (i = 0; i < inst->blockLen10ms; i++) {
         outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
       }
-    } // end of H band gain computation
+    }  // end of H band gain computation
     return 0;
   }
 
@@ -2153,7 +2153,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
     priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
     // priorLocSnr = 1 + 2*priorSnr
     priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
-  } // end of loop over frequencies
+  }  // end of loop over frequencies
   // done with step 1: DD computation of prior and post SNR
 
   // STEP 2: compute speech/noise likelihood
@@ -2314,7 +2314,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
       tmp32no2 += tmp32no1; // Q(qMagn)
     }
     inst->avgMagnPause[i] = tmp32no2;
-  } // end of frequency loop
+  }  // end of frequency loop
 
   norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
   qNoise = inst->prevQNoise + norm32no1 - 5;
@@ -2389,7 +2389,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
       inst->noiseSupFilter[i] = (uint16_t)WebRtcSpl_DivU32U16(tmpU32no1,
                                                                     END_STARTUP_SHORT);
     }
-  } // end of loop over frequencies
+  }  // end of loop over frequencies
   //done with step3
 
   // save noise and magnitude spectrum for next frame
@@ -2484,7 +2484,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
       outFrameHB[i]
         = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
     }
-  } // end of H band gain computation
+  }  // end of H band gain computation
 
   return 0;
 }

webrtc/modules/audio_processing/ns/nsx_core_neon.c

@@ -291,7 +291,7 @@ void WebRtcNsx_NoiseEstimationNeon(NsxInst_t* inst,
       __asm__("vbit %q0, %q1, %q2"::
               "w"(tmp16x8_1), "w"(tmp16x8_0), "w"(tmp16x8_4));
       vst1q_s16(&inst->noiseEstDensity[offset + i], tmp16x8_1);
-    } // End loop over magnitude spectrum
+    }  // End loop over magnitude spectrum
 
     // Last iteration over magnitude spectrum:
     // compute delta
@@ -348,7 +348,7 @@ void WebRtcNsx_NoiseEstimationNeon(NsxInst_t* inst,
     }
     inst->noiseEstCounter[s]++;
 
-  } // end loop over simultaneous estimates
+  }  // end loop over simultaneous estimates
 
   // Sequentially update the noise during startup
   if (inst->blockIndex < END_STARTUP_LONG) {

webrtc/modules/desktop_capture/mac/desktop_configuration.mm

@@ -83,7 +83,7 @@ MacDisplayConfiguration GetConfigurationForScreen(NSScreen* screen) {
   return display_config;
 }
 
-} // namespace
+}  // namespace
 
 MacDisplayConfiguration::MacDisplayConfiguration()
     : id(0),

webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.cc

@@ -52,4 +52,4 @@ void ScopedPixelBufferObject::Release() {
   }
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h

@@ -36,6 +36,6 @@ class ScopedPixelBufferObject {
   DISALLOW_COPY_AND_ASSIGN(ScopedPixelBufferObject);
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_DESKTOP_CAPTURE_SCOPED_PIXEL_BUFFER_OBJECT_H_

webrtc/modules/interface/module_common_types.h

@@ -1067,6 +1067,6 @@ inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
       timestamp2;
 }
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // MODULE_COMMON_TYPES_H

webrtc/modules/media_file/interface/media_file.h

@@ -223,5 +223,5 @@ protected:
     MediaFile() {}
     virtual ~MediaFile() {}
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_

webrtc/modules/media_file/interface/media_file_defines.h

@@ -47,5 +47,5 @@ public:
 protected:
     FileCallback() {}
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_

webrtc/modules/media_file/source/avi_file.cc

@@ -1763,4 +1763,4 @@ void AviFile::WriteIndex()
     }
     PutLE32LengthFromCurrent(static_cast<long>(idxChunkSize));
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/media_file/source/avi_file.h

@@ -271,6 +271,6 @@ private:
 
     ListWrapper* _indexList; // Elements are of type AVIINDEXENTRY.
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_

webrtc/modules/media_file/source/media_file_impl.cc

@@ -1368,4 +1368,4 @@ bool MediaFileImpl::ValidFrequency(const uint32_t frequency)
                  "Frequency should be 8000, 16000 or 32000 (Hz)");
     return false;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/media_file/source/media_file_impl.h

@@ -236,6 +236,6 @@ private:
 
     FileCallback* _ptrCallback;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_

webrtc/modules/media_file/source/media_file_utility.cc

@@ -55,7 +55,7 @@ struct WAVE_CHUNK_header
    int8_t  fmt_ckID[4];
    int32_t fmt_ckSize;
 };
-} // unnamed namespace
+}  // unnamed namespace
 
 namespace webrtc {
 ModuleFileUtility::ModuleFileUtility(const int32_t id)
@@ -2559,4 +2559,4 @@ uint32_t ModuleFileUtility::PlayoutPositionMs()
         return 0;
     }
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/media_file/source/media_file_utility.h

@@ -343,5 +343,5 @@ private:
     VideoCodec _videoCodec;
 #endif
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_

webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h

@@ -52,6 +52,6 @@ public:
     uint32_t      _incomingBitRate;
     double              _noiseVar;
 };
-} //namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_

webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc

@@ -381,4 +381,4 @@ void RemoteRateControl::StateStr(BandwidthUsage state, char* str) {
       assert(false);
   }
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/remote_bitrate_estimator/remote_rate_control.h

@@ -75,6 +75,6 @@ class RemoteRateControl {
   float beta_;
   unsigned int rtt_;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_

webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h

@@ -932,5 +932,5 @@ class RtpRtcp : public Module {
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

@@ -307,5 +307,5 @@ class NullRtpAudioFeedback : public RtpAudioFeedback {
                                     const uint8_t volume) {}
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_

webrtc/modules/rtp_rtcp/source/H264/bitstream_builder.cc

@@ -577,4 +577,4 @@ BitstreamBuilder::AddSuffix(const uint8_t numBits, const uint32_t rest)
         }
     }
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/H264/bitstream_builder.h

@@ -47,6 +47,6 @@ private:
     uint32_t _byteOffset;
     uint8_t  _bitOffset;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_

webrtc/modules/rtp_rtcp/source/H264/bitstream_parser.cc

@@ -214,4 +214,4 @@ BitstreamParser::GetUE()
     }
     return retVal;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/H264/bitstream_parser.h

@@ -41,6 +41,6 @@ private:
     uint32_t _byteOffset;
     uint8_t  _bitOffset;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_

webrtc/modules/rtp_rtcp/source/H264/h264_information.cc

@@ -815,4 +815,4 @@ H264Information::SetLayerLengths()
 
     return 0;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/H264/h264_information.h

@@ -165,6 +165,6 @@ private:
     uint32_t          _remLength;
     H264Info          _info;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_

webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc

@@ -466,7 +466,7 @@ RTPSenderH264::SendH264_STAP_A(const FrameType frameType,
         }
     }
     return 0;
-} // end STAP-A
+}  // end STAP-A
 
 // STAP-A for H.264 SVC
 int32_t
@@ -660,7 +660,7 @@ RTPSenderH264::SendH264_STAP_A_PACSI(const FrameType frameType,
         }
     }
     return 0;
-} // end STAP-A
+}  // end STAP-A
 
 int32_t
 RTPSenderH264::SendH264_FU_A(const FrameType frameType,
@@ -1277,4 +1277,4 @@ RTPSenderH264::SendH264(const FrameType frameType,
     }
     return 0;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.h

@@ -174,6 +174,6 @@ private:
 
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_

webrtc/modules/rtp_rtcp/source/forward_error_correction.h

@@ -308,5 +308,5 @@ class ForwardErrorCorrection {
   FecPacketList fec_packet_list_;
   bool fec_packet_received_;
 };
-}       // namespace webrtc
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_

webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc

@@ -370,7 +370,7 @@ void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
                           num_mask_bytes, packet_mask, mask_table);
 
   }  // End of UEP modification
-}    //End of GetPacketMasks
+}  //End of GetPacketMasks
 
 }  // namespace internal
 }  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc

@@ -124,4 +124,4 @@ TEST_F(RtcpFormatRembTest, TestCompund) {
   EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 2, SSRCs));
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRemb));
 }
-} // namespace
+}  // namespace

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -1449,4 +1449,4 @@ void RTCPReceiver::PacketTimeout()
     }
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_receiver.h

@@ -243,5 +243,5 @@ protected:
   uint16_t _rtt;
 
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc

@@ -197,4 +197,4 @@ void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
     const uint32_t minimumSize) {
   TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h

@@ -130,7 +130,7 @@ private:
     std::vector<int64_t> _tmmbrSetTimeouts;
 };
 
-} // end namespace RTCPHelp
+}  // end namespace RTCPHelp
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -2136,4 +2136,4 @@ RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
     }
     return -1;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

@@ -269,6 +269,6 @@ private:
     uint32_t      _pliCount;
     uint32_t      _fullIntraRequestCount;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_

webrtc/modules/rtp_rtcp/source/rtcp_utility.cc

@@ -1524,4 +1524,4 @@ RTCPUtility::RTCPPacketIterator::Current()
 
     return &_header;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtcp_utility.h

@@ -441,6 +441,6 @@ namespace RTCPUtility {
 
         RTCPCommonHeader         _header;
     };
-} // RTCPUtility
+}  // RTCPUtility
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_

webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc

@@ -178,4 +178,4 @@ void RtpHeaderExtensionMap::GetCopy(RtpHeaderExtensionMap* map) const {
     it++;
   }
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtp_receiver.cc

@@ -822,7 +822,7 @@ int32_t RTPReceiver::CheckPayloadChanged(
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
       is_red = false;
     }
-  }   // End critsect.
+  }  // End critsect.
 
   if (re_initialize_decoder) {
     if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
@@ -1137,4 +1137,4 @@ void RTPReceiver::ProcessBitrate() {
                     "RTPReceiverPacketRate", ssrc_, PacketRate());
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtp_receiver.h

@@ -237,6 +237,6 @@ class RTPReceiver : public Bitrate {
   int payload_type_rtx_;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_

webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h

@@ -47,7 +47,7 @@ enum { DTMF_OUTBAND_MAX         = 20};
 enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
 enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
 enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
-} // namespace webrtc
+}  // namespace webrtc
 
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -474,7 +474,7 @@ int32_t RTPSenderAudio::SendAudio(
       }
     }
     _lastPayloadType = payloadType;
-  }   // end critical section
+  }  // end critical section
   TRACE_EVENT_INSTANT2("webrtc_rtp", "Audio::Send",
                        "timestamp", captureTimeStamp,
                        "seqnum", _rtpSender->SequenceNumber());
@@ -631,4 +631,4 @@ RTPSenderAudio::SendTelephoneEventPacket(const bool ended,
 
     return retVal;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -121,6 +121,6 @@ private:
     uint8_t     _audioLevelIndicationID;
     uint8_t     _audioLevel_dBov;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -507,4 +507,4 @@ int RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   return 0;
 }
 
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -137,6 +137,6 @@ private:
     // Bitrate used for video payload and RTP headers
     Bitrate                   _videoBitrate;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_

webrtc/modules/rtp_rtcp/source/ssrc_database.cc

@@ -214,4 +214,4 @@ uint32_t SSRCDatabase::GenerateRandom()
 
     return ssrc;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/ssrc_database.h

@@ -57,6 +57,6 @@ private:
 
     CriticalSectionWrapper* _critSect;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_

webrtc/modules/rtp_rtcp/source/tmmbr_help.cc

@@ -479,4 +479,4 @@ bool TMMBRHelp::CalcMinBitRate( uint32_t* minBitrateKbit) const {
   }
   return true;
 }
-} // namespace webrtc
+}  // namespace webrtc

webrtc/modules/rtp_rtcp/source/tmmbr_help.h

@@ -110,6 +110,6 @@ private:
     float*                  _ptrIntersectionBoundingSet;
     float*                  _ptrMaxPRBoundingSet;
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_

webrtc/modules/rtp_rtcp/source/video_codec_information.h

@@ -23,6 +23,6 @@ public:
     virtual RtpVideoCodecTypes Type() = 0;
     virtual ~VideoCodecInformation(){};
 };
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_

webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h

@@ -185,4 +185,4 @@ const float kMaxResidualLossBurstyMask[kNumberCodes] = {
   0.009657f
 };
 
-}   // namespace
+}  // namespace

webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc

@@ -473,11 +473,11 @@ TEST(FecTest, FecTest) {
             }
             timeStamp += 90000 / 30;
           }  // loop over numImpPackets
-        }    // loop over FecPackets
+        }  // loop over FecPackets
-      }      // loop over numMediaPackets
+      }  // loop over numMediaPackets
       delete[] packetMask;
     }  // loop over loss rates
-  }    // loop over mask types
+  }  // loop over mask types
 
   // Have DecodeFEC free allocated memory.
   fec.ResetState(&recoveredPacketList);

webrtc/modules/utility/interface/audio_frame_operations.h

@@ -53,6 +53,6 @@ class AudioFrameOperations {
   static int ScaleWithSat(float scale, AudioFrame& frame);
 };
 
-}  //  namespace webrtc
+}  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_

webrtc/modules/utility/interface/file_player.h

@@ -107,5 +107,5 @@ protected:
     virtual ~FilePlayer() {}
 
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_

webrtc/modules/utility/interface/file_recorder.h

@@ -86,5 +86,5 @@ protected:
     virtual ~FileRecorder() {}
 
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_

webrtc/modules/utility/interface/process_thread.h

@@ -30,5 +30,5 @@ public:
 protected:
     virtual ~ProcessThread();
 };
-} // namespace webrtc
+}  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_