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.
26 lines
932 B
C
26 lines
932 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 RTC_BASE_COMPILE_ASSERT_C_H_
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#define RTC_BASE_COMPILE_ASSERT_C_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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// RTC_COMPILE_ASSERT(sizeof(foo) < 128);
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// Note: In C++, use static_assert instead!
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#define RTC_COMPILE_ASSERT(expression) \
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switch (0) { \
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case 0: \
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case expression:; \
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}
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#endif // RTC_BASE_COMPILE_ASSERT_C_H_
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