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.
23 lines
706 B
Plaintext
23 lines
706 B
Plaintext
/*
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* Copyright 2016 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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#include "rtc_base/logging_mac.h"
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#import <Foundation/Foundation.h>
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namespace rtc {
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std::string DescriptionFromOSStatus(OSStatus err) {
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NSError* error =
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[NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
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return error.description.UTF8String;
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}
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} // namespace rtc
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