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.
32 lines
827 B
C
32 lines
827 B
C
/*
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* Copyright (c) 2018 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_SYSTEM_INLINE_H_
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#define RTC_BASE_SYSTEM_INLINE_H_
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#if defined(_MSC_VER)
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#define RTC_FORCE_INLINE __forceinline
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#define RTC_NO_INLINE __declspec(noinline)
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#elif defined(__GNUC__)
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#define RTC_FORCE_INLINE __attribute__((__always_inline__))
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#define RTC_NO_INLINE __attribute__((__noinline__))
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#else
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#define RTC_FORCE_INLINE
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#define RTC_NO_INLINE
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#endif
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#endif // RTC_BASE_SYSTEM_INLINE_H_
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