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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.
27 lines
1.2 KiB
C
27 lines
1.2 KiB
C
/*
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* Copyright (c) 2012 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_VAD_PITCH_INTERNAL_H_
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#define MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
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// TODO(turajs): Write a description of this function. Also be consistent with
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// usage of |sampling_rate_hz| vs |kSamplingFreqHz|.
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void GetSubframesPitchParameters(int sampling_rate_hz,
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double* gains,
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double* lags,
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int num_in_frames,
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int num_out_frames,
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double* log_old_gain,
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double* old_lag,
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double* log_pitch_gain,
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double* pitch_lag_hz);
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#endif // MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
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