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webrtc_m130/webrtc/modules/audio_processing/agc/analog_agc.c
andrew@webrtc.org 27c6980239 Move the volume quantization workaround from VoE to AGC. 
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 20:24:56 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/* analog_agc.c
*
* Using a feedback system, determines an appropriate analog volume level
* given an input signal and current volume level. Targets a conservative
* signal level and is intended for use with a digital AGC to apply
* additional gain.
*
*/
#include <assert.h>
#include <stdlib.h>
#ifdef AGC_DEBUG //test log
#include <stdio.h>
#endif
#include "webrtc/modules/audio_processing/agc/analog_agc.h"
/* The slope of in Q13*/
static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
/* The offset in Q14 */
static const int16_t kOffset1[8] = {25395, 23911, 22206, 20737, 19612, 18805, 17951,
17367};
/* The slope of in Q13*/
static const int16_t kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
/* The offset in Q14 */
static const int16_t kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670,
17286};
static const int16_t kMuteGuardTimeMs = 8000;
static const int16_t kInitCheck = 42;
/* Default settings if config is not used */
#define AGC_DEFAULT_TARGET_LEVEL 3
#define AGC_DEFAULT_COMP_GAIN 9
/* This is the target level for the analog part in ENV scale. To convert to RMS scale you
* have to add OFFSET_ENV_TO_RMS.
*/
#define ANALOG_TARGET_LEVEL 11
#define ANALOG_TARGET_LEVEL_2 5 // ANALOG_TARGET_LEVEL / 2
/* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
* varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future replace it with
* a table.
*/
#define OFFSET_ENV_TO_RMS 9
/* The reference input level at which the digital part gives an output of targetLevelDbfs
* (desired level) if we have no compression gain. This level should be set high enough not
* to compress the peaks due to the dynamics.
*/
#define DIGITAL_REF_AT_0_COMP_GAIN 4
/* Speed of reference level decrease.
*/
#define DIFF_REF_TO_ANALOG 5
#ifdef MIC_LEVEL_FEEDBACK
#define NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET 7
#endif
/* Size of analog gain table */
#define GAIN_TBL_LEN 32
/* Matlab code:
* fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
*/
/* Q12 */
static const uint16_t kGainTableAnalog[GAIN_TBL_LEN] = {4096, 4251, 4412, 4579, 4752,
4932, 5118, 5312, 5513, 5722, 5938, 6163, 6396, 6638, 6889, 7150, 7420, 7701, 7992,
8295, 8609, 8934, 9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};
/* Gain/Suppression tables for virtual Mic (in Q10) */
static const uint16_t kGainTableVirtualMic[128] = {1052, 1081, 1110, 1141, 1172, 1204,
1237, 1271, 1305, 1341, 1378, 1416, 1454, 1494, 1535, 1577, 1620, 1664, 1710, 1757,
1805, 1854, 1905, 1957, 2010, 2065, 2122, 2180, 2239, 2301, 2364, 2428, 2495, 2563,
2633, 2705, 2779, 2855, 2933, 3013, 3096, 3180, 3267, 3357, 3449, 3543, 3640, 3739,
3842, 3947, 4055, 4166, 4280, 4397, 4517, 4640, 4767, 4898, 5032, 5169, 5311, 5456,
5605, 5758, 5916, 6078, 6244, 6415, 6590, 6770, 6956, 7146, 7341, 7542, 7748, 7960,
8178, 8402, 8631, 8867, 9110, 9359, 9615, 9878, 10148, 10426, 10711, 11004, 11305,
11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807, 15212, 15628,
16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923, 20468, 21028, 21603,
22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808, 27541, 28295, 29069, 29864,
30681, 31520, 32382};
static const uint16_t kSuppressionTableVirtualMic[128] = {1024, 1006, 988, 970, 952,
935, 918, 902, 886, 870, 854, 839, 824, 809, 794, 780, 766, 752, 739, 726, 713, 700,
687, 675, 663, 651, 639, 628, 616, 605, 594, 584, 573, 563, 553, 543, 533, 524, 514,
505, 496, 487, 478, 470, 461, 453, 445, 437, 429, 421, 414, 406, 399, 392, 385, 378,
371, 364, 358, 351, 345, 339, 333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278,
273, 268, 263, 258, 254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
200, 197, 193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155, 153, 150,
147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118, 116, 114, 112, 110,
108, 106, 104, 102};
/* Table for target energy levels. Values in Q(-7)
* Matlab code
* targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n', round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
static const int32_t kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67264106,
53429779, 42440782, 33711911, 26778323, 21270778, 16895980, 13420954, 10660642,
8468049, 6726411, 5342978, 4244078, 3371191, 2677832, 2127078, 1689598, 1342095,
1066064, 846805, 672641, 534298, 424408, 337119, 267783, 212708, 168960, 134210,
106606, 84680, 67264, 53430, 42441, 33712, 26778, 21271, 16896, 13421, 10661, 8468,
6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
213, 169, 134, 107, 85, 67};
int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
int16_t samples)
{
int32_t nrg, max_nrg, sample, tmp32;
int32_t *ptr;
uint16_t targetGainIdx, gain;
int16_t i, n, L, M, subFrames, tmp16, tmp_speech[16];
Agc_t *stt;
stt = (Agc_t *)state;
//default/initial values corresponding to 10ms for wb and swb
M = 10;
L = 16;
subFrames = 160;
if (stt->fs == 8000)
{
if (samples == 80)
{
subFrames = 80;
M = 10;
L = 8;
} else if (samples == 160)
{
subFrames = 80;
M = 20;
L = 8;
} else
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid number of samples\n\n",
(stt->fcount + 1));
#endif
return -1;
}
} else if (stt->fs == 16000)
{
if (samples == 160)
{
subFrames = 160;
M = 10;
L = 16;
} else if (samples == 320)
{
subFrames = 160;
M = 20;
L = 16;
} else
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid number of samples\n\n",
(stt->fcount + 1));
#endif
return -1;
}
} else if (stt->fs == 32000)
{
/* SWB is processed as 160 sample for L and H bands */
if (samples == 160)
{
subFrames = 160;
M = 10;
L = 16;
} else
{
#ifdef AGC_DEBUG
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid sample rate\n\n",
(stt->fcount + 1));
#endif
return -1;
}
}
/* Check for valid pointers based on sampling rate */
if ((stt->fs == 32000) && (in_mic_H == NULL))
{
return -1;
}
/* Check for valid pointer for low band */
if (in_mic == NULL)
{
return -1;
}
/* apply slowly varying digital gain */
if (stt->micVol > stt->maxAnalog)
{
/* |maxLevel| is strictly >= |micVol|, so this condition should be
* satisfied here, ensuring there is no divide-by-zero. */
assert(stt->maxLevel > stt->maxAnalog);
/* Q1 */
tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
targetGainIdx = (uint16_t)WEBRTC_SPL_DIV(tmp32, tmp16);
assert(targetGainIdx < GAIN_TBL_LEN);
/* Increment through the table towards the target gain.
* If micVol drops below maxAnalog, we allow the gain
* to be dropped immediately. */
if (stt->gainTableIdx < targetGainIdx)
{
stt->gainTableIdx++;
} else if (stt->gainTableIdx > targetGainIdx)
{
stt->gainTableIdx--;
}
/* Q12 */
gain = kGainTableAnalog[stt->gainTableIdx];
for (i = 0; i < samples; i++)
{
// For lower band
tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic[i], gain);
sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
if (sample > 32767)
{
in_mic[i] = 32767;
} else if (sample < -32768)
{
in_mic[i] = -32768;
} else
{
in_mic[i] = (int16_t)sample;
}
// For higher band
if (stt->fs == 32000)
{
tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic_H[i], gain);
sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
if (sample > 32767)
{
in_mic_H[i] = 32767;
} else if (sample < -32768)
{
in_mic_H[i] = -32768;
} else
{
in_mic_H[i] = (int16_t)sample;
}
}
}
} else
{
stt->gainTableIdx = 0;
}
/* compute envelope */
if ((M == 10) && (stt->inQueue > 0))
{
ptr = stt->env[1];
} else
{
ptr = stt->env[0];
}
for (i = 0; i < M; i++)
{
/* iterate over samples */
max_nrg = 0;
for (n = 0; n < L; n++)
{
nrg = WEBRTC_SPL_MUL_16_16(in_mic[i * L + n], in_mic[i * L + n]);
if (nrg > max_nrg)
{
max_nrg = nrg;
}
}
ptr[i] = max_nrg;
}
/* compute energy */
if ((M == 10) && (stt->inQueue > 0))
{
ptr = stt->Rxx16w32_array[1];
} else
{
ptr = stt->Rxx16w32_array[0];
}
for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++)
{
if (stt->fs == 16000)
{
WebRtcSpl_DownsampleBy2(&in_mic[i * 32], 32, tmp_speech, stt->filterState);
} else
{
memcpy(tmp_speech, &in_mic[i * 16], 16 * sizeof(short));
}
/* Compute energy in blocks of 16 samples */
ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
}
/* update queue information */
if ((stt->inQueue == 0) && (M == 10))
{
stt->inQueue = 1;
} else
{
stt->inQueue = 2;
}
/* call VAD (use low band only) */
for (i = 0; i < samples; i += subFrames)
{
WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames);
}
return 0;
}
int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
{
int32_t errHandle = 0;
int16_t i, subFrames;
Agc_t *stt;
stt = (Agc_t *)state;
if (stt == NULL)
{
return -1;
}
if (stt->fs == 8000)
{
if ((samples != 80) && (samples != 160))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 80;
} else if (stt->fs == 16000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else if (stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid sample rate\n\n",
stt->fcount + 1);
#endif
return -1;
}
for (i = 0; i < samples; i += subFrames)
{
errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames);
}
return errHandle;
}
int WebRtcAgc_VirtualMic(void *agcInst, int16_t *in_near, int16_t *in_near_H,
int16_t samples, int32_t micLevelIn,
int32_t *micLevelOut)
{
int32_t tmpFlt, micLevelTmp, gainIdx;
uint16_t gain;
int16_t ii;
Agc_t *stt;
uint32_t nrg;
int16_t sampleCntr;
uint32_t frameNrg = 0;
uint32_t frameNrgLimit = 5500;
int16_t numZeroCrossing = 0;
const int16_t kZeroCrossingLowLim = 15;
const int16_t kZeroCrossingHighLim = 20;
stt = (Agc_t *)agcInst;
/*
* Before applying gain decide if this is a low-level signal.
* The idea is that digital AGC will not adapt to low-level
* signals.
*/
if (stt->fs != 8000)
{
frameNrgLimit = frameNrgLimit << 1;
}
frameNrg = WEBRTC_SPL_MUL_16_16(in_near[0], in_near[0]);
for (sampleCntr = 1; sampleCntr < samples; sampleCntr++)
{
// increment frame energy if it is less than the limit
// the correct value of the energy is not important
if (frameNrg < frameNrgLimit)
{
nrg = WEBRTC_SPL_MUL_16_16(in_near[sampleCntr], in_near[sampleCntr]);
frameNrg += nrg;
}
// Count the zero crossings
numZeroCrossing += ((in_near[sampleCntr] ^ in_near[sampleCntr - 1]) < 0);
}
if ((frameNrg < 500) || (numZeroCrossing <= 5))
{
stt->lowLevelSignal = 1;
} else if (numZeroCrossing <= kZeroCrossingLowLim)
{
stt->lowLevelSignal = 0;
} else if (frameNrg <= frameNrgLimit)
{
stt->lowLevelSignal = 1;
} else if (numZeroCrossing >= kZeroCrossingHighLim)
{
stt->lowLevelSignal = 1;
} else
{
stt->lowLevelSignal = 0;
}
micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
/* Set desired level */
gainIdx = stt->micVol;
if (stt->micVol > stt->maxAnalog)
{
gainIdx = stt->maxAnalog;
}
if (micLevelTmp != stt->micRef)
{
/* Something has happened with the physical level, restart. */
stt->micRef = micLevelTmp;
stt->micVol = 127;
*micLevelOut = 127;
stt->micGainIdx = 127;
gainIdx = 127;
}
/* Pre-process the signal to emulate the microphone level. */
/* Take one step at a time in the gain table. */
if (gainIdx > 127)
{
gain = kGainTableVirtualMic[gainIdx - 128];
} else
{
gain = kSuppressionTableVirtualMic[127 - gainIdx];
}
for (ii = 0; ii < samples; ii++)
{
tmpFlt = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_U16(in_near[ii], gain), 10);
if (tmpFlt > 32767)
{
tmpFlt = 32767;
gainIdx--;
if (gainIdx >= 127)
{
gain = kGainTableVirtualMic[gainIdx - 127];
} else
{
gain = kSuppressionTableVirtualMic[127 - gainIdx];
}
}
if (tmpFlt < -32768)
{
tmpFlt = -32768;
gainIdx--;
if (gainIdx >= 127)
{
gain = kGainTableVirtualMic[gainIdx - 127];
} else
{
gain = kSuppressionTableVirtualMic[127 - gainIdx];
}
}
in_near[ii] = (int16_t)tmpFlt;
if (stt->fs == 32000)
{
tmpFlt = WEBRTC_SPL_MUL_16_U16(in_near_H[ii], gain);
tmpFlt = WEBRTC_SPL_RSHIFT_W32(tmpFlt, 10);
if (tmpFlt > 32767)
{
tmpFlt = 32767;
}
if (tmpFlt < -32768)
{
tmpFlt = -32768;
}
in_near_H[ii] = (int16_t)tmpFlt;
}
}
/* Set the level we (finally) used */
stt->micGainIdx = gainIdx;
// *micLevelOut = stt->micGainIdx;
*micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale);
/* Add to Mic as if it was the output from a true microphone */
if (WebRtcAgc_AddMic(agcInst, in_near, in_near_H, samples) != 0)
{
return -1;
}
return 0;
}
void WebRtcAgc_UpdateAgcThresholds(Agc_t *stt)
{
int16_t tmp16;
#ifdef MIC_LEVEL_FEEDBACK
int zeros;
if (stt->micLvlSat)
{
/* Lower the analog target level since we have reached its maximum */
zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
stt->targetIdxOffset = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2);
}
#endif
/* Set analog target level in envelope dBOv scale */
tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
tmp16 = WebRtcSpl_DivW32W16ResW16((int32_t)tmp16, ANALOG_TARGET_LEVEL);
stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN)
{
stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
}
if (stt->agcMode == kAgcModeFixedDigital)
{
/* Adjust for different parameter interpretation in FixedDigital mode */
stt->analogTarget = stt->compressionGaindB;
}
#ifdef MIC_LEVEL_FEEDBACK
stt->analogTarget += stt->targetIdxOffset;
#endif
/* Since the offset between RMS and ENV is not constant, we should make this into a
* table, but for now, we'll stick with a constant, tuned for the chosen analog
* target level.
*/
stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
#ifdef MIC_LEVEL_FEEDBACK
stt->targetIdx += stt->targetIdxOffset;
#endif
/* Analog adaptation limits */
/* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
stt->analogTargetLevel = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
stt->startUpperLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 1];/* -19 dBov */
stt->startLowerLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 1];/* -21 dBov */
stt->upperPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 2];/* -18 dBov */
stt->lowerPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 2];/* -22 dBov */
stt->upperSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 5];/* -15 dBov */
stt->lowerSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 5];/* -25 dBov */
stt->upperLimit = stt->startUpperLimit;
stt->lowerLimit = stt->startLowerLimit;
}
void WebRtcAgc_SaturationCtrl(Agc_t *stt, uint8_t *saturated, int32_t *env)
{
int16_t i, tmpW16;
/* Check if the signal is saturated */
for (i = 0; i < 10; i++)
{
tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(env[i], 20);
if (tmpW16 > 875)
{
stt->envSum += tmpW16;
}
}
if (stt->envSum > 25000)
{
*saturated = 1;
stt->envSum = 0;
}
/* stt->envSum *= 0.99; */
stt->envSum = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum,
(int16_t)32440, 15);
}
void WebRtcAgc_ZeroCtrl(Agc_t *stt, int32_t *inMicLevel, int32_t *env)
{
int16_t i;
int32_t tmp32 = 0;
int32_t midVal;
/* Is the input signal zero? */
for (i = 0; i < 10; i++)
{
tmp32 += env[i];
}
/* Each block is allowed to have a few non-zero
* samples.
*/
if (tmp32 < 500)
{
stt->msZero += 10;
} else
{
stt->msZero = 0;
}
if (stt->muteGuardMs > 0)
{
stt->muteGuardMs -= 10;
}
if (stt->msZero > 500)
{
stt->msZero = 0;
/* Increase microphone level only if it's less than 50% */
midVal = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog + stt->minLevel + 1, 1);
if (*inMicLevel < midVal)
{
/* *inMicLevel *= 1.1; */
tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel);
*inMicLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
/* Reduces risk of a muted mic repeatedly triggering excessive levels due
* to zero signal detection. */
*inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
stt->micVol = *inMicLevel;
}
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
stt->fcount, stt->micVol);
#endif
stt->activeSpeech = 0;
stt->Rxx16_LPw32Max = 0;
/* The AGC has a tendency (due to problems with the VAD parameters), to
* vastly increase the volume after a muting event. This timer prevents
* upwards adaptation for a short period. */
stt->muteGuardMs = kMuteGuardTimeMs;
}
}
void WebRtcAgc_SpeakerInactiveCtrl(Agc_t *stt)
{
/* Check if the near end speaker is inactive.
* If that is the case the VAD threshold is
* increased since the VAD speech model gets
* more sensitive to any sound after a long
* silence.
*/
int32_t tmp32;
int16_t vadThresh;
if (stt->vadMic.stdLongTerm < 2500)
{
stt->vadThreshold = 1500;
} else
{
vadThresh = kNormalVadThreshold;
if (stt->vadMic.stdLongTerm < 4500)
{
/* Scale between min and max threshold */
vadThresh += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1);
}
/* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
tmp32 = (int32_t)vadThresh;
tmp32 += WEBRTC_SPL_MUL_16_16((int16_t)31, stt->vadThreshold);
stt->vadThreshold = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 5);
}
}
void WebRtcAgc_ExpCurve(int16_t volume, int16_t *index)
{
// volume in Q14
// index in [0-7]
/* 8 different curves */
if (volume > 5243)
{
if (volume > 7864)
{
if (volume > 12124)
{
*index = 7;
} else
{
*index = 6;
}
} else
{
if (volume > 6554)
{
*index = 5;
} else
{
*index = 4;
}
}
} else
{
if (volume > 2621)
{
if (volume > 3932)
{
*index = 3;
} else
{
*index = 2;
}
} else
{
if (volume > 1311)
{
*index = 1;
} else
{
*index = 0;
}
}
}
}
int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
int32_t *outMicLevel,
int16_t vadLogRatio,
int16_t echo, uint8_t *saturationWarning)
{
uint32_t tmpU32;
int32_t Rxx16w32, tmp32;
int32_t inMicLevelTmp, lastMicVol;
int16_t i;
uint8_t saturated = 0;
Agc_t *stt;
stt = (Agc_t *)state;
inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
if (inMicLevelTmp > stt->maxAnalog)
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
#endif
return -1;
} else if (inMicLevelTmp < stt->minLevel)
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
#endif
return -1;
}
if (stt->firstCall == 0)
{
int32_t tmpVol;
stt->firstCall = 1;
tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)51, 9);
tmpVol = (stt->minLevel + tmp32);
/* If the mic level is very low at start, increase it! */
if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog))
{
inMicLevelTmp = tmpVol;
}
stt->micVol = inMicLevelTmp;
}
/* Set the mic level to the previous output value if there is digital input gain */
if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog))
{
inMicLevelTmp = stt->micVol;
}
/* If the mic level was manually changed to a very low value raise it! */
if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput))
{
tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)51, 9);
inMicLevelTmp = (stt->minLevel + tmp32);
stt->micVol = inMicLevelTmp;
#ifdef MIC_LEVEL_FEEDBACK
//stt->numBlocksMicLvlSat = 0;
#endif
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
stt->fcount);
#endif
}
if (inMicLevelTmp != stt->micVol)
{
if (inMicLevel == stt->lastInMicLevel) {
// We requested a volume adjustment, but it didn't occur. This is
// probably due to a coarse quantization of the volume slider.
// Restore the requested value to prevent getting stuck.
inMicLevelTmp = stt->micVol;
}
else {
// As long as the value changed, update to match.
stt->micVol = inMicLevelTmp;
}
}
if (inMicLevelTmp > stt->maxLevel)
{
// Always allow the user to raise the volume above the maxLevel.
stt->maxLevel = inMicLevelTmp;
}
// Store last value here, after we've taken care of manual updates etc.
stt->lastInMicLevel = inMicLevel;
lastMicVol = stt->micVol;
/* Checks if the signal is saturated. Also a check if individual samples
* are larger than 12000 is done. If they are the counter for increasing
* the volume level is set to -100ms
*/
WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);
/* The AGC is always allowed to lower the level if the signal is saturated */
if (saturated == 1)
{
/* Lower the recording level
* Rxx160_LP is adjusted down because it is so slow it could
* cause the AGC to make wrong decisions. */
/* stt->Rxx160_LPw32 *= 0.875; */
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7);
stt->zeroCtrlMax = stt->micVol;
/* stt->micVol *= 0.903; */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(29591, (uint32_t)(tmp32));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 2)
{
stt->micVol = lastMicVol - 2;
}
inMicLevelTmp = stt->micVol;
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
stt->fcount, stt->micVol);
#endif
if (stt->micVol < stt->minOutput)
{
*saturationWarning = 1;
}
/* Reset counter for decrease of volume level to avoid
* decreasing too much. The saturation control can still
* lower the level if needed. */
stt->msTooHigh = -100;
/* Enable the control mechanism to ensure that our measure,
* Rxx160_LP, is in the correct range. This must be done since
* the measure is very slow. */
stt->activeSpeech = 0;
stt->Rxx16_LPw32Max = 0;
/* Reset to initial values */
stt->msecSpeechInnerChange = kMsecSpeechInner;
stt->msecSpeechOuterChange = kMsecSpeechOuter;
stt->changeToSlowMode = 0;
stt->muteGuardMs = 0;
stt->upperLimit = stt->startUpperLimit;
stt->lowerLimit = stt->startLowerLimit;
#ifdef MIC_LEVEL_FEEDBACK
//stt->numBlocksMicLvlSat = 0;
#endif
}
/* Check if the input speech is zero. If so the mic volume
* is increased. On some computers the input is zero up as high
* level as 17% */
WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);
/* Check if the near end speaker is inactive.
* If that is the case the VAD threshold is
* increased since the VAD speech model gets
* more sensitive to any sound after a long
* silence.
*/
WebRtcAgc_SpeakerInactiveCtrl(stt);
for (i = 0; i < 5; i++)
{
/* Computed on blocks of 16 samples */
Rxx16w32 = stt->Rxx16w32_array[0][i];
/* Rxx160w32 in Q(-7) */
tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos], 3);
stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
/* Circular buffer */
stt->Rxx16pos++;
if (stt->Rxx16pos == RXX_BUFFER_LEN)
{
stt->Rxx16pos = 0;
}
/* Rxx16_LPw32 in Q(-4) */
tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_LPw32, kAlphaShortTerm);
stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;
if (vadLogRatio > stt->vadThreshold)
{
/* Speech detected! */
/* Check if Rxx160_LP is in the correct range. If
* it is too high/low then we set it to the maximum of
* Rxx16_LPw32 during the first 200ms of speech.
*/
if (stt->activeSpeech < 250)
{
stt->activeSpeech += 2;
if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max)
{
stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
}
} else if (stt->activeSpeech == 250)
{
stt->activeSpeech += 2;
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx16_LPw32Max, 3);
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN);
}
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160w32 - stt->Rxx160_LPw32, kAlphaLongTerm);
stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;
if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit)
{
stt->msTooHigh += 2;
stt->msTooLow = 0;
stt->changeToSlowMode = 0;
if (stt->msTooHigh > stt->msecSpeechOuterChange)
{
stt->msTooHigh = 0;
/* Lower the recording level */
/* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
/* Reduce the max gain to avoid excessive oscillation
* (but never drop below the maximum analog level).
* stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
*/
tmp32 = (15 * stt->maxLevel) + stt->micVol;
stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
stt->zeroCtrlMax = stt->micVol;
/* 0.95 in Q15 */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(31130, (uint32_t)(tmp32));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 1)
{
stt->micVol = lastMicVol - 1;
}
inMicLevelTmp = stt->micVol;
/* Enable the control mechanism to ensure that our measure,
* Rxx160_LP, is in the correct range.
*/
stt->activeSpeech = 0;
stt->Rxx16_LPw32Max = 0;
#ifdef MIC_LEVEL_FEEDBACK
//stt->numBlocksMicLvlSat = 0;
#endif
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
stt->fcount, stt->micVol, stt->maxLevel);
#endif
}
} else if (stt->Rxx160_LPw32 > stt->upperLimit)
{
stt->msTooHigh += 2;
stt->msTooLow = 0;
stt->changeToSlowMode = 0;
if (stt->msTooHigh > stt->msecSpeechInnerChange)
{
/* Lower the recording level */
stt->msTooHigh = 0;
/* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
/* Reduce the max gain to avoid excessive oscillation
* (but never drop below the maximum analog level).
* stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
*/
tmp32 = (15 * stt->maxLevel) + stt->micVol;
stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
stt->zeroCtrlMax = stt->micVol;
/* 0.965 in Q15 */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(31621, (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 1)
{
stt->micVol = lastMicVol - 1;
}
inMicLevelTmp = stt->micVol;
#ifdef MIC_LEVEL_FEEDBACK
//stt->numBlocksMicLvlSat = 0;
#endif
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
stt->fcount, stt->micVol, stt->maxLevel);
#endif
}
} else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
{
stt->msTooHigh = 0;
stt->changeToSlowMode = 0;
stt->msTooLow += 2;
if (stt->msTooLow > stt->msecSpeechOuterChange)
{
/* Raise the recording level */
int16_t index, weightFIX;
int16_t volNormFIX = 16384; // =1 in Q14.
stt->msTooLow = 0;
/* Normalize the volume level */
tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
if (stt->maxInit != stt->minLevel)
{
volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
(stt->maxInit - stt->minLevel));
}
/* Find correct curve */
WebRtcAgc_ExpCurve(volNormFIX, &index);
/* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05 */
weightFIX = kOffset1[index]
- (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(kSlope1[index],
volNormFIX, 13);
/* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
if (stt->micVol < lastMicVol + 2)
{
stt->micVol = lastMicVol + 2;
}
inMicLevelTmp = stt->micVol;
#ifdef MIC_LEVEL_FEEDBACK
/* Count ms in level saturation */
//if (stt->micVol > stt->maxAnalog) {
if (stt->micVol > 150)
{
/* mic level is saturated */
stt->numBlocksMicLvlSat++;
fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
}
#endif
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
stt->fcount, stt->micVol);
#endif
}
} else if (stt->Rxx160_LPw32 < stt->lowerLimit)
{
stt->msTooHigh = 0;
stt->changeToSlowMode = 0;
stt->msTooLow += 2;
if (stt->msTooLow > stt->msecSpeechInnerChange)
{
/* Raise the recording level */
int16_t index, weightFIX;
int16_t volNormFIX = 16384; // =1 in Q14.
stt->msTooLow = 0;
/* Normalize the volume level */
tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
if (stt->maxInit != stt->minLevel)
{
volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
(stt->maxInit - stt->minLevel));
}
/* Find correct curve */
WebRtcAgc_ExpCurve(volNormFIX, &index);
/* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1 */
weightFIX = kOffset2[index]
- (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(kSlope2[index],
volNormFIX, 13);
/* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
if (stt->micVol < lastMicVol + 1)
{
stt->micVol = lastMicVol + 1;
}
inMicLevelTmp = stt->micVol;
#ifdef MIC_LEVEL_FEEDBACK
/* Count ms in level saturation */
//if (stt->micVol > stt->maxAnalog) {
if (stt->micVol > 150)
{
/* mic level is saturated */
stt->numBlocksMicLvlSat++;
fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
}
#endif
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
stt->fcount, stt->micVol);
#endif
}
} else
{
/* The signal is inside the desired range which is:
* lowerLimit < Rxx160_LP/640 < upperLimit
*/
if (stt->changeToSlowMode > 4000)
{
stt->msecSpeechInnerChange = 1000;
stt->msecSpeechOuterChange = 500;
stt->upperLimit = stt->upperPrimaryLimit;
stt->lowerLimit = stt->lowerPrimaryLimit;
} else
{
stt->changeToSlowMode += 2; // in milliseconds
}
stt->msTooLow = 0;
stt->msTooHigh = 0;
stt->micVol = inMicLevelTmp;
}
#ifdef MIC_LEVEL_FEEDBACK
if (stt->numBlocksMicLvlSat > NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET)
{
stt->micLvlSat = 1;
fprintf(stderr, "target before = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
WebRtcAgc_UpdateAgcThresholds(stt);
WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]),
stt->compressionGaindB, stt->targetLevelDbfs, stt->limiterEnable,
stt->analogTarget);
stt->numBlocksMicLvlSat = 0;
stt->micLvlSat = 0;
fprintf(stderr, "target offset = %d\n", stt->targetIdxOffset);
fprintf(stderr, "target after = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
}
#endif
}
}
/* Ensure gain is not increased in presence of echo or after a mute event
* (but allow the zeroCtrl() increase on the frame of a mute detection).
*/
if (echo == 1 || (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs))
{
if (stt->micVol > lastMicVol)
{
stt->micVol = lastMicVol;
}
}
/* limit the gain */
if (stt->micVol > stt->maxLevel)
{
stt->micVol = stt->maxLevel;
} else if (stt->micVol < stt->minOutput)
{
stt->micVol = stt->minOutput;
}
*outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->micVol, stt->scale);
if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale))
{
*outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale);
}
return 0;
}
int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
const int16_t *in_near_H, int16_t samples,
int16_t *out, int16_t *out_H, int32_t inMicLevel,
int32_t *outMicLevel, int16_t echo,
uint8_t *saturationWarning)
{
Agc_t *stt;
int32_t inMicLevelTmp;
int16_t subFrames, i;
uint8_t satWarningTmp = 0;
stt = (Agc_t *)agcInst;
//
if (stt == NULL)
{
return -1;
}
//
if (stt->fs == 8000)
{
if ((samples != 80) && (samples != 160))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
#endif
return -1;
}
subFrames = 80;
} else if (stt->fs == 16000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else if (stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef AGC_DEBUG //test log
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else
{
#ifdef AGC_DEBUG// test log
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
#endif
return -1;
}
/* Check for valid pointers based on sampling rate */
if (stt->fs == 32000 && in_near_H == NULL)
{
return -1;
}
/* Check for valid pointers for low band */
if (in_near == NULL)
{
return -1;
}
*saturationWarning = 0;
//TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS
*outMicLevel = inMicLevel;
inMicLevelTmp = inMicLevel;
// TODO(andrew): clearly we don't need input and output pointers...
// Change the interface to take a shared input/output.
if (in_near != out)
{
// Only needed if they don't already point to the same place.
memcpy(out, in_near, samples * sizeof(int16_t));
}
if (stt->fs == 32000)
{
if (in_near_H != out_H)
{
memcpy(out_H, in_near_H, samples * sizeof(int16_t));
}
}
#ifdef AGC_DEBUG//test log
stt->fcount++;
#endif
for (i = 0; i < samples; i += subFrames)
{
if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
stt->fs, stt->lowLevelSignal) == -1)
{
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
#endif
return -1;
}
if ((stt->agcMode < kAgcModeFixedDigital) && ((stt->lowLevelSignal == 0)
|| (stt->agcMode != kAgcModeAdaptiveDigital)))
{
if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevelTmp, outMicLevel,
stt->vadMic.logRatio, echo, saturationWarning) == -1)
{
return -1;
}
}
#ifdef AGC_DEBUG//test log
fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
#endif
/* update queue */
if (stt->inQueue > 1)
{
memcpy(stt->env[0], stt->env[1], 10 * sizeof(int32_t));
memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(int32_t));
}
if (stt->inQueue > 0)
{
stt->inQueue--;
}
/* If 20ms frames are used the input mic level must be updated so that
* the analog AGC does not think that there has been a manual volume
* change. */
inMicLevelTmp = *outMicLevel;
/* Store a positive saturation warning. */
if (*saturationWarning == 1)
{
satWarningTmp = 1;
}
}
/* Trigger the saturation warning if displayed by any of the frames. */
*saturationWarning = satWarningTmp;
return 0;
}
int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig)
{
Agc_t *stt;
stt = (Agc_t *)agcInst;
if (stt == NULL)
{
return -1;
}
if (stt->initFlag != kInitCheck)
{
stt->lastError = AGC_UNINITIALIZED_ERROR;
return -1;
}
if (agcConfig.limiterEnable != kAgcFalse && agcConfig.limiterEnable != kAgcTrue)
{
stt->lastError = AGC_BAD_PARAMETER_ERROR;
return -1;
}
stt->limiterEnable = agcConfig.limiterEnable;
stt->compressionGaindB = agcConfig.compressionGaindB;
if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31))
{
stt->lastError = AGC_BAD_PARAMETER_ERROR;
return -1;
}
stt->targetLevelDbfs = agcConfig.targetLevelDbfs;
if (stt->agcMode == kAgcModeFixedDigital)
{
/* Adjust for different parameter interpretation in FixedDigital mode */
stt->compressionGaindB += agcConfig.targetLevelDbfs;
}
/* Update threshold levels for analog adaptation */
WebRtcAgc_UpdateAgcThresholds(stt);
/* Recalculate gain table */
if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
{
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
#endif
return -1;
}
/* Store the config in a WebRtcAgc_config_t */
stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;
return 0;
}
int WebRtcAgc_get_config(void *agcInst, WebRtcAgc_config_t *config)
{
Agc_t *stt;
stt = (Agc_t *)agcInst;
if (stt == NULL)
{
return -1;
}
if (config == NULL)
{
stt->lastError = AGC_NULL_POINTER_ERROR;
return -1;
}
if (stt->initFlag != kInitCheck)
{
stt->lastError = AGC_UNINITIALIZED_ERROR;
return -1;
}
config->limiterEnable = stt->usedConfig.limiterEnable;
config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
config->compressionGaindB = stt->usedConfig.compressionGaindB;
return 0;
}
int WebRtcAgc_Create(void **agcInst)
{
Agc_t *stt;
if (agcInst == NULL)
{
return -1;
}
stt = (Agc_t *)malloc(sizeof(Agc_t));
*agcInst = stt;
if (stt == NULL)
{
return -1;
}
#ifdef AGC_DEBUG
stt->fpt = fopen("./agc_test_log.txt", "wt");
stt->agcLog = fopen("./agc_debug_log.txt", "wt");
stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
#endif
stt->initFlag = 0;
stt->lastError = 0;
return 0;
}
int WebRtcAgc_Free(void *state)
{
Agc_t *stt;
stt = (Agc_t *)state;
#ifdef AGC_DEBUG
fclose(stt->fpt);
fclose(stt->agcLog);
fclose(stt->digitalAgc.logFile);
#endif
free(stt);
return 0;
}
/* minLevel - Minimum volume level
* maxLevel - Maximum volume level
*/
int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel,
int16_t agcMode, uint32_t fs)
{
int32_t max_add, tmp32;
int16_t i;
int tmpNorm;
Agc_t *stt;
/* typecast state pointer */
stt = (Agc_t *)agcInst;
if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0)
{
stt->lastError = AGC_UNINITIALIZED_ERROR;
return -1;
}
/* Analog AGC variables */
stt->envSum = 0;
/* mode = 0 - Only saturation protection
* 1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
* 2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
* 3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
*/
#ifdef AGC_DEBUG//test log
stt->fcount = 0;
fprintf(stt->fpt, "AGC->Init\n");
#endif
if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
{
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
#endif
return -1;
}
stt->agcMode = agcMode;
stt->fs = fs;
/* initialize input VAD */
WebRtcAgc_InitVad(&stt->vadMic);
/* If the volume range is smaller than 0-256 then
* the levels are shifted up to Q8-domain */
tmpNorm = WebRtcSpl_NormU32((uint32_t)maxLevel);
stt->scale = tmpNorm - 23;
if (stt->scale < 0)
{
stt->scale = 0;
}
// TODO(bjornv): Investigate if we really need to scale up a small range now when we have
// a guard against zero-increments. For now, we do not support scale up (scale = 0).
stt->scale = 0;
maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale);
minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale);
/* Make minLevel and maxLevel static in AdaptiveDigital */
if (stt->agcMode == kAgcModeAdaptiveDigital)
{
minLevel = 0;
maxLevel = 255;
stt->scale = 0;
}
/* The maximum supplemental volume range is based on a vague idea
* of how much lower the gain will be than the real analog gain. */
max_add = WEBRTC_SPL_RSHIFT_W32(maxLevel - minLevel, 2);
/* Minimum/maximum volume level that can be set */
stt->minLevel = minLevel;
stt->maxAnalog = maxLevel;
stt->maxLevel = maxLevel + max_add;
stt->maxInit = stt->maxLevel;
stt->zeroCtrlMax = stt->maxAnalog;
stt->lastInMicLevel = 0;
/* Initialize micVol parameter */
stt->micVol = stt->maxAnalog;
if (stt->agcMode == kAgcModeAdaptiveDigital)
{
stt->micVol = 127; /* Mid-point of mic level */
}
stt->micRef = stt->micVol;
stt->micGainIdx = 127;
#ifdef MIC_LEVEL_FEEDBACK
stt->numBlocksMicLvlSat = 0;
stt->micLvlSat = 0;
#endif
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt,
"AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
stt->minLevel, stt->maxAnalog, stt->maxLevel);
#endif
/* Minimum output volume is 4% higher than the available lowest volume level */
tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (int32_t)10, 8);
stt->minOutput = (stt->minLevel + tmp32);
stt->msTooLow = 0;
stt->msTooHigh = 0;
stt->changeToSlowMode = 0;
stt->firstCall = 0;
stt->msZero = 0;
stt->muteGuardMs = 0;
stt->gainTableIdx = 0;
stt->msecSpeechInnerChange = kMsecSpeechInner;
stt->msecSpeechOuterChange = kMsecSpeechOuter;
stt->activeSpeech = 0;
stt->Rxx16_LPw32Max = 0;
stt->vadThreshold = kNormalVadThreshold;
stt->inActive = 0;
for (i = 0; i < RXX_BUFFER_LEN; i++)
{
stt->Rxx16_vectorw32[i] = (int32_t)1000; /* -54dBm0 */
}
stt->Rxx160w32 = 125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */
stt->Rxx16pos = 0;
stt->Rxx16_LPw32 = (int32_t)16284; /* Q(-4) */
for (i = 0; i < 5; i++)
{
stt->Rxx16w32_array[0][i] = 0;
}
for (i = 0; i < 10; i++)
{
stt->env[0][i] = 0;
stt->env[1][i] = 0;
}
stt->inQueue = 0;
#ifdef MIC_LEVEL_FEEDBACK
stt->targetIdxOffset = 0;
#endif
WebRtcSpl_MemSetW32(stt->filterState, 0, 8);
stt->initFlag = kInitCheck;
// Default config settings.
stt->defaultConfig.limiterEnable = kAgcTrue;
stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;
if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1)
{
stt->lastError = AGC_UNSPECIFIED_ERROR;
return -1;
}
stt->Rxx160_LPw32 = stt->analogTargetLevel; // Initialize rms value
stt->lowLevelSignal = 0;
/* Only positive values are allowed that are not too large */
if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
{
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
#endif
return -1;
} else
{
#ifdef AGC_DEBUG//test log
fprintf(stt->fpt, "\n");
#endif
return 0;
}
}
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