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.
44 lines
1.1 KiB
C
44 lines
1.1 KiB
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_RTC_EXPORT_H_
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#define RTC_BASE_SYSTEM_RTC_EXPORT_H_
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// RTC_EXPORT is used to mark symbols as exported or imported when WebRTC is
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// built or used as a shared library.
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// When WebRTC is built as a static library the RTC_EXPORT macro expands to
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// nothing.
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#ifdef WEBRTC_ENABLE_SYMBOL_EXPORT
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#ifdef WEBRTC_WIN
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#ifdef WEBRTC_LIBRARY_IMPL
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#define RTC_EXPORT __declspec(dllexport)
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#else
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#define RTC_EXPORT __declspec(dllimport)
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#endif
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#else // WEBRTC_WIN
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#if __has_attribute(visibility) && defined(WEBRTC_LIBRARY_IMPL)
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#define RTC_EXPORT __attribute__((visibility("default")))
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#endif
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#endif // WEBRTC_WIN
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#endif // WEBRTC_ENABLE_SYMBOL_EXPORT
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#ifndef RTC_EXPORT
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#define RTC_EXPORT
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#endif
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#endif // RTC_BASE_SYSTEM_RTC_EXPORT_H_
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