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session-android/jni/webrtc/modules/audio_processing/agc/analog_agc.c
Moxie Marlinspike d83a3d71bc Support for Signal calls. 
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

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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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