mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 09:37:52 +01:00
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
/*
|
|
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "rtc_base/logging_mac.h"
|
|
|
|
#import <Foundation/Foundation.h>
|
|
|
|
namespace rtc {
|
|
std::string DescriptionFromOSStatus(OSStatus err) {
|
|
NSError* error =
|
|
[NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
|
|
return error.description.UTF8String;
|
|
}
|
|
|
|
} // namespace rtc
|