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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.
38 lines
1.2 KiB
C++
38 lines
1.2 KiB
C++
/*
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* Copyright (c) 2014 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 MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_
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#define MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_
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#ifdef _MSC_VER /* visual c++ */
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#define ALIGN16_BEG __declspec(align(16))
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#define ALIGN16_END
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#else /* gcc or icc */
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#define ALIGN16_BEG
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#define ALIGN16_END __attribute__((aligned(16)))
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#endif
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#ifdef __cplusplus
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namespace webrtc {
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#endif
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extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_sqrtHanning[65];
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extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65];
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extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65];
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extern const float WebRtcAec_kExtendedSmoothingCoefficients[2][2];
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extern const float WebRtcAec_kNormalSmoothingCoefficients[2][2];
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extern const float WebRtcAec_kMinFarendPSD;
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#ifdef __cplusplus
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} // namespace webrtc
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#endif
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#endif // MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_
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