mirror of
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.
22 lines
894 B
C
22 lines
894 B
C
/*
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* Copyright (c) 2012 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_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
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#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
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// Use this macro to verify at compile time that certain restrictions are met.
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// The argument is the boolean expression to evaluate.
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// Example:
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// COMPILE_ASSERT(sizeof(foo) < 128);
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// Note: In C++, use static_assert instead!
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#define COMPILE_ASSERT(expression) switch (0) {case 0: case expression:;}
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#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
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