tg2sip/libtgvoip/webrtc_dsp/modules/audio_processing/common.h

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/*
* 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_