Files
platform-external-webrtc/webrtc/modules/audio_processing/voice_detection_impl.h
kjellander@webrtc.org 14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00

65 lines
2.1 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioBuffer;
class CriticalSectionWrapper;
class VoiceDetectionImpl : public VoiceDetection,
public ProcessingComponent {
public:
VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
virtual ~VoiceDetectionImpl();
int ProcessCaptureAudio(AudioBuffer* audio);
// VoiceDetection implementation.
bool is_enabled() const override;
// ProcessingComponent implementation.
int Initialize() override;
private:
// VoiceDetection implementation.
int Enable(bool enable) override;
int set_stream_has_voice(bool has_voice) override;
bool stream_has_voice() const override;
int set_likelihood(Likelihood likelihood) override;
Likelihood likelihood() const override;
int set_frame_size_ms(int size) override;
int frame_size_ms() const override;
// ProcessingComponent implementation.
void* CreateHandle() const override;
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
int num_handles_required() const override;
int GetHandleError(void* handle) const override;
const AudioProcessing* apm_;
CriticalSectionWrapper* crit_;
bool stream_has_voice_;
bool using_external_vad_;
Likelihood likelihood_;
int frame_size_ms_;
int frame_size_samples_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_