mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +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.
35 lines
1014 B
C++
35 lines
1014 B
C++
/*
|
|
* Copyright (c) 2014 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_COMMON_H_
|
|
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
|
|
switch (layout) {
|
|
case AudioProcessing::kMono:
|
|
case AudioProcessing::kMonoAndKeyboard:
|
|
return 1;
|
|
case AudioProcessing::kStereo:
|
|
case AudioProcessing::kStereoAndKeyboard:
|
|
return 2;
|
|
}
|
|
RTC_NOTREACHED();
|
|
return 0;
|
|
}
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_COMMON_H_
|