Reason for revert: Because of down-stream dependencies, this CL needs to be reverted. The dependencies will be resolved and then the CL will be relanded. Original issue's description: > Revert "Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )" > > This reverts commit c54aad6ae07fe2a44a65be403386bd7d7d865e5b. > > BUG=webrtc:5724 > NOPRESUBMIT=true > > Committed: https://crrev.com/8864fe5e08f8d8711612526dee9a812adfcd3be1 > Cr-Commit-Position: refs/heads/master@{#12247} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5724 Review URL: https://codereview.webrtc.org/1855393004 Cr-Commit-Position: refs/heads/master@{#12248}
72 lines
1.7 KiB
C++
72 lines
1.7 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_AEC_ECHO_CANCELLATION_INTERNAL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
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extern "C" {
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#include "webrtc/common_audio/ring_buffer.h"
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}
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
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namespace webrtc {
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typedef struct {
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int delayCtr;
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int sampFreq;
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int splitSampFreq;
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int scSampFreq;
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float sampFactor; // scSampRate / sampFreq
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short skewMode;
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int bufSizeStart;
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int knownDelay;
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int rate_factor;
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short initFlag; // indicates if AEC has been initialized
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// Variables used for averaging far end buffer size
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short counter;
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int sum;
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short firstVal;
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short checkBufSizeCtr;
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// Variables used for delay shifts
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short msInSndCardBuf;
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short filtDelay; // Filtered delay estimate.
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int timeForDelayChange;
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int startup_phase;
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int checkBuffSize;
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short lastDelayDiff;
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#ifdef WEBRTC_AEC_DEBUG_DUMP
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FILE* bufFile;
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FILE* delayFile;
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FILE* skewFile;
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#endif
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// Structures
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void* resampler;
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int skewFrCtr;
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int resample; // if the skew is small enough we don't resample
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int highSkewCtr;
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float skew;
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RingBuffer* far_pre_buf; // Time domain far-end pre-buffer.
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int farend_started;
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AecCore* aec;
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} Aec;
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
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