dce40cf804
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
81 lines
2.9 KiB
C
81 lines
2.9 KiB
C
/*
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* Copyright (c) 2011 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_AGC_LEGACY_DIGITAL_AGC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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#include <stdio.h>
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#endif
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/typedefs.h"
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// the 32 most significant bits of A(19) * B(26) >> 13
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#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
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// C + the 32 most significant bits of A * B
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#define AGC_SCALEDIFF32(A, B, C) ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
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typedef struct
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{
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int32_t downState[8];
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int16_t HPstate;
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int16_t counter;
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int16_t logRatio; // log( P(active) / P(inactive) ) (Q10)
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int16_t meanLongTerm; // Q10
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int32_t varianceLongTerm; // Q8
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int16_t stdLongTerm; // Q10
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int16_t meanShortTerm; // Q10
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int32_t varianceShortTerm; // Q8
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int16_t stdShortTerm; // Q10
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} AgcVad; // total = 54 bytes
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typedef struct
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{
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int32_t capacitorSlow;
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int32_t capacitorFast;
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int32_t gain;
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int32_t gainTable[32];
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int16_t gatePrevious;
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int16_t agcMode;
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AgcVad vadNearend;
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AgcVad vadFarend;
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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FILE* logFile;
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int frameCounter;
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#endif
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} DigitalAgc;
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int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
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int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
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const int16_t* const* inNear,
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size_t num_bands,
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int16_t* const* out,
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uint32_t FS,
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int16_t lowLevelSignal);
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int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
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const int16_t* inFar,
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size_t nrSamples);
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void WebRtcAgc_InitVad(AgcVad* vadInst);
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int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst, // (i) VAD state
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const int16_t* in, // (i) Speech signal
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size_t nrSamples); // (i) number of samples
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int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
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int16_t compressionGaindB, // Q0 (in dB)
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int16_t targetLevelDbfs,// Q0 (in dB)
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uint8_t limiterEnable,
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int16_t analogTarget);
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
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