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libtgvoip/webrtc_dsp/modules/audio_processing/transient/dyadic_decimator.h

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/*
* Copyright (c) 2013 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 MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#define MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
#include <cstdlib>
// Provides a set of static methods to perform dyadic decimations.
namespace webrtc {
// Returns the proper length of the output buffer that you should use for the
// given |in_length| and decimation |odd_sequence|.
// Return -1 on error.
inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
bool odd_sequence) {
size_t out_length = in_length / 2;
if (in_length % 2 == 1 && !odd_sequence) {
++out_length;
}
return out_length;
}
// Performs a dyadic decimation: removes every odd/even member of a sequence
// halving its overall length.
// Arguments:
// in: array of |in_length|.
// odd_sequence: If false, the odd members will be removed (1, 3, 5, ...);
// if true, the even members will be removed (0, 2, 4, ...).
// out: array of |out_length|. |out_length| must be large enough to
// hold the decimated output. The necessary length can be provided by
// GetOutLengthToDyadicDecimate().
// Must be previously allocated.
// Returns the number of output samples, -1 on error.
template <typename T>
static size_t DyadicDecimate(const T* in,
size_t in_length,
bool odd_sequence,
T* out,
size_t out_length) {
size_t half_length = GetOutLengthToDyadicDecimate(in_length, odd_sequence);
if (!in || !out || in_length <= 0 || out_length < half_length) {
return 0;
}
size_t output_samples = 0;
size_t index_adjustment = odd_sequence ? 1 : 0;
for (output_samples = 0; output_samples < half_length; ++output_samples) {
out[output_samples] = in[output_samples * 2 + index_adjustment];
}
return output_samples;
}
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_