From d72b3d6c81e487b879bd36c58740af41f822c312 Mon Sep 17 00:00:00 2001 From: "andrew@webrtc.org" Date: Thu, 15 Nov 2012 21:46:06 +0000 Subject: [PATCH] Fix cpplint errors in audio_processing.h TBR=leozwang@webrtc.org BUG=1114 Review URL: https://webrtc-codereview.appspot.com/934015 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3108 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../include/audio_processing.h | 32 +++++++++---------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h index a70dd2cd2..192b5399d 100644 --- a/webrtc/modules/audio_processing/include/audio_processing.h +++ b/webrtc/modules/audio_processing/include/audio_processing.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_ +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ -#include // size_t +#include // size_t -#include "module.h" -#include "typedefs.h" +#include "webrtc/modules/interface/module.h" +#include "webrtc/typedefs.h" namespace webrtc { @@ -113,7 +113,7 @@ class AudioProcessing : public Module { // for each far-end stream which requires processing. On the server-side, // this would typically be one instance for every incoming stream. static AudioProcessing* Create(int id); - virtual ~AudioProcessing() {}; + virtual ~AudioProcessing() {} // TODO(andrew): remove this method. We now allow users to delete instances // directly, useful for scoped_ptr. @@ -251,8 +251,8 @@ class AudioProcessing : public Module { }; // Inherited from Module. - virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; }; - virtual WebRtc_Word32 Process() { return -1; }; + virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; } + virtual WebRtc_Word32 Process() { return -1; } }; // The acoustic echo cancellation (AEC) component provides better performance @@ -343,7 +343,7 @@ class EchoCancellation { virtual int GetDelayMetrics(int* median, int* std) = 0; protected: - virtual ~EchoCancellation() {}; + virtual ~EchoCancellation() {} }; // The acoustic echo control for mobile (AECM) component is a low complexity @@ -399,7 +399,7 @@ class EchoControlMobile { static size_t echo_path_size_bytes(); protected: - virtual ~EchoControlMobile() {}; + virtual ~EchoControlMobile() {} }; // The automatic gain control (AGC) component brings the signal to an @@ -491,7 +491,7 @@ class GainControl { virtual bool stream_is_saturated() const = 0; protected: - virtual ~GainControl() {}; + virtual ~GainControl() {} }; // A filtering component which removes DC offset and low-frequency noise. @@ -502,7 +502,7 @@ class HighPassFilter { virtual bool is_enabled() const = 0; protected: - virtual ~HighPassFilter() {}; + virtual ~HighPassFilter() {} }; // An estimation component used to retrieve level metrics. @@ -525,7 +525,7 @@ class LevelEstimator { virtual int RMS() = 0; protected: - virtual ~LevelEstimator() {}; + virtual ~LevelEstimator() {} }; // The noise suppression (NS) component attempts to remove noise while @@ -555,7 +555,7 @@ class NoiseSuppression { virtual float speech_probability() const = 0; protected: - virtual ~NoiseSuppression() {}; + virtual ~NoiseSuppression() {} }; // The voice activity detection (VAD) component analyzes the stream to @@ -605,8 +605,8 @@ class VoiceDetection { virtual int frame_size_ms() const = 0; protected: - virtual ~VoiceDetection() {}; + virtual ~VoiceDetection() {} }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_ +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_