Arun Raghavan 753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30

53 lines
1.9 KiB
C++

/*
* 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 WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
#include <queue>
#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
// Calculates the first and second moments for each value of a buffer taking
// into account a given number of previous values.
// It preserves its state, so it can be multiple-called.
// TODO(chadan): Implement a function that takes a buffer of first moments and a
// buffer of second moments; and calculates the variances. When needed.
// TODO(chadan): Add functionality to update with a buffer but only output are
// the last values of the moments. When needed.
class MovingMoments {
public:
// Creates a Moving Moments object, that uses the last |length| values
// (including the new value introduced in every new calculation).
explicit MovingMoments(size_t length);
~MovingMoments();
// Calculates the new values using |in|. Results will be in the out buffers.
// |first| and |second| must be allocated with at least |in_length|.
void CalculateMoments(const float* in, size_t in_length,
float* first, float* second);
private:
size_t length_;
// A queue holding the |length_| latest input values.
std::queue<float> queue_;
// Sum of the values of the queue.
float sum_;
// Sum of the squares of the values of the queue.
float sum_of_squares_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_