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.
29 lines
821 B
C++
29 lines
821 B
C++
/*
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* Copyright 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_LOGGING_MAC_H_
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#define RTC_BASE_LOGGING_MAC_H_
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#if !defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
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#error "Only include this header in macOS builds"
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#endif
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#include <CoreServices/CoreServices.h>
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#include <string>
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namespace rtc {
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// Returns a UTF8 description from an OS X Status error.
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std::string DescriptionFromOSStatus(OSStatus err);
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} // namespace rtc
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#endif // RTC_BASE_LOGGING_MAC_H_
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