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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
33 lines
1.2 KiB
C++
33 lines
1.2 KiB
C++
/*
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* Copyright (c) 2015 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 RTC_BASE_ARRAYSIZE_H_
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#define RTC_BASE_ARRAYSIZE_H_
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#include <stddef.h>
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// This file defines the arraysize() macro and is derived from Chromium's
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// base/macros.h.
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// The arraysize(arr) macro returns the # of elements in an array arr.
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// The expression is a compile-time constant, and therefore can be
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// used in defining new arrays, for example. If you use arraysize on
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// a pointer by mistake, you will get a compile-time error.
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// This template function declaration is used in defining arraysize.
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// Note that the function doesn't need an implementation, as we only
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// use its type.
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template <typename T, size_t N>
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char (&ArraySizeHelper(T (&array)[N]))[N];
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#define arraysize(array) (sizeof(ArraySizeHelper(array)))
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#endif // RTC_BASE_ARRAYSIZE_H_
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