14665ff7d4
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
83 lines
2.7 KiB
C++
83 lines
2.7 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_GAIN_CONTROL_IMPL_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
|
#include "webrtc/modules/audio_processing/processing_component.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioBuffer;
|
|
class CriticalSectionWrapper;
|
|
|
|
class GainControlImpl : public GainControl,
|
|
public ProcessingComponent {
|
|
public:
|
|
GainControlImpl(const AudioProcessing* apm,
|
|
CriticalSectionWrapper* crit);
|
|
virtual ~GainControlImpl();
|
|
|
|
int ProcessRenderAudio(AudioBuffer* audio);
|
|
int AnalyzeCaptureAudio(AudioBuffer* audio);
|
|
int ProcessCaptureAudio(AudioBuffer* audio);
|
|
|
|
// ProcessingComponent implementation.
|
|
int Initialize() override;
|
|
|
|
// GainControl implementation.
|
|
bool is_enabled() const override;
|
|
int stream_analog_level() override;
|
|
|
|
private:
|
|
// GainControl implementation.
|
|
int Enable(bool enable) override;
|
|
int set_stream_analog_level(int level) override;
|
|
int set_mode(Mode mode) override;
|
|
Mode mode() const override;
|
|
int set_target_level_dbfs(int level) override;
|
|
int target_level_dbfs() const override;
|
|
int set_compression_gain_db(int gain) override;
|
|
int compression_gain_db() const override;
|
|
int enable_limiter(bool enable) override;
|
|
bool is_limiter_enabled() const override;
|
|
int set_analog_level_limits(int minimum, int maximum) override;
|
|
int analog_level_minimum() const override;
|
|
int analog_level_maximum() const override;
|
|
bool stream_is_saturated() 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_;
|
|
Mode mode_;
|
|
int minimum_capture_level_;
|
|
int maximum_capture_level_;
|
|
bool limiter_enabled_;
|
|
int target_level_dbfs_;
|
|
int compression_gain_db_;
|
|
std::vector<int> capture_levels_;
|
|
int analog_capture_level_;
|
|
bool was_analog_level_set_;
|
|
bool stream_is_saturated_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
|