session-android/jni/webrtc/modules/audio_coding/neteq/accelerate.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

78 lines
2.9 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.
virtual 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.
virtual 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_