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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
78 lines
2.9 KiB
C++
78 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
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#include <assert.h>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declarations.
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class BackgroundNoise;
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// This class implements the Accelerate operation. Most of the work is done
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// in the base class TimeStretch, which is shared with the PreemptiveExpand
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// operation. In the Accelerate class, the operations that are specific to
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// Accelerate are implemented.
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class Accelerate : public TimeStretch {
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public:
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Accelerate(int sample_rate_hz, size_t num_channels,
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const BackgroundNoise& background_noise)
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: TimeStretch(sample_rate_hz, num_channels, background_noise) {
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}
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virtual ~Accelerate() {}
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// This method performs the actual Accelerate operation. The samples are
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// read from |input|, of length |input_length| elements, and are written to
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// |output|. The number of samples removed through time-stretching is
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// is provided in the output |length_change_samples|. The method returns
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// the outcome of the operation as an enumerator value.
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ReturnCodes Process(const int16_t* input,
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size_t input_length,
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AudioMultiVector* output,
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int16_t* length_change_samples);
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protected:
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// Sets the parameters |best_correlation| and |peak_index| to suitable
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// values when the signal contains no active speech.
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virtual void SetParametersForPassiveSpeech(size_t len,
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int16_t* best_correlation,
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int* peak_index) const OVERRIDE;
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// Checks the criteria for performing the time-stretching operation and,
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// if possible, performs the time-stretching.
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virtual ReturnCodes CheckCriteriaAndStretch(
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const int16_t* input, size_t input_length, size_t peak_index,
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int16_t best_correlation, bool active_speech,
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AudioMultiVector* output) const OVERRIDE;
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private:
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DISALLOW_COPY_AND_ASSIGN(Accelerate);
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};
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struct AccelerateFactory {
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AccelerateFactory() {}
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virtual ~AccelerateFactory() {}
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virtual Accelerate* Create(int sample_rate_hz,
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size_t num_channels,
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const BackgroundNoise& background_noise) const;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
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