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/*
* Copyright (c) 2014 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_COMMON_AUDIO_FIR_FILTER_H_
#define WEBRTC_COMMON_AUDIO_FIR_FILTER_H_
#include <string.h>
namespace webrtc {
// Finite Impulse Response filter using floating-point arithmetic.
class FIRFilter {
public:
// Creates a filter with the given coefficients. All initial state values will
// be zeros.
// The length of the chunks fed to the filter should never be greater than
// |max_input_length|. This is needed because, when vectorizing it is
// necessary to concatenate the input after the state, and resizing this array
// dynamically is expensive.
static FIRFilter* Create(const float* coefficients,
size_t coefficients_length,
size_t max_input_length);
virtual ~FIRFilter() {}
// Filters the |in| data supplied.
// |out| must be previously allocated and it must be at least of |length|.
virtual void Filter(const float* in, size_t length, float* out) = 0;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_FIR_FILTER_H_
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