Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/accelerate.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

80 lines
3.0 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_CODING_NETEQ_ACCELERATE_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
#include <assert.h>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Forward declarations.
class BackgroundNoise;
// This class implements the Accelerate operation. Most of the work is done
// in the base class TimeStretch, which is shared with the PreemptiveExpand
// operation. In the Accelerate class, the operations that are specific to
// Accelerate are implemented.
class Accelerate : public TimeStretch {
public:
Accelerate(int sample_rate_hz, size_t num_channels,
const BackgroundNoise& background_noise)
: TimeStretch(sample_rate_hz, num_channels, background_noise) {
}
virtual ~Accelerate() {}
// This method performs the actual Accelerate operation. The samples are
// read from |input|, of length |input_length| elements, and are written to
// |output|. The number of samples removed through time-stretching is
// is provided in the output |length_change_samples|. The method returns
// the outcome of the operation as an enumerator value.
ReturnCodes Process(const int16_t* input,
size_t input_length,
AudioMultiVector* output,
int16_t* length_change_samples);
protected:
// Sets the parameters |best_correlation| and |peak_index| to suitable
// values when the signal contains no active speech.
void SetParametersForPassiveSpeech(size_t len,
int16_t* best_correlation,
int* peak_index) const override;
// Checks the criteria for performing the time-stretching operation and,
// if possible, performs the time-stretching.
ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
size_t input_length,
size_t peak_index,
int16_t best_correlation,
bool active_speech,
AudioMultiVector* output) const override;
private:
DISALLOW_COPY_AND_ASSIGN(Accelerate);
};
struct AccelerateFactory {
AccelerateFactory() {}
virtual ~AccelerateFactory() {}
virtual Accelerate* Create(int sample_rate_hz,
size_t num_channels,
const BackgroundNoise& background_noise) const;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_