
This change generates bit-exact values when running through audioproc_f than before. This change was originally uploaded here: * https://codereview.webrtc.org/1181933002/ * https://codereview.webrtc.org/1177043017/ And reverted because of an ASAN problem in Chrome here: * https://codereview.webrtc.org/1192863006/ * https://codereview.webrtc.org/1194963003/ TBR=andrew@webrtc.org Review URL: https://codereview.webrtc.org/1211563003 Cr-Commit-Position: refs/heads/master@{#9502}
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 WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
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#define WEBRTC_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 // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
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