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