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Modify WebRTC gaincontroller2, remove level estimation, remove VAD
This is necessary to update from r4332 to r6818. Modifies WebRTCAudioPreprocessor::gaincontroller2: struct { TTBOOL bEnable; float fInitialSaturationMarginDB; float fExtraSaturationMarginDB; float fMaxGainChangeDBPerSecond; float fMaxOutputNoiseLevelDBFS; } adaptivedigital; By: struct { TTBOOL bEnable; float fHeadRoomDB; float fMaxGainDB; float fInitialGainDB; float fMaxGainChangeDBPerSecond; float fMaxOutputNoiseLevelDBFS; } adaptivedigital;
1 parent 2287dc7 commit 2dd2acc

26 files changed

+160
-362
lines changed

‎Client/TeamTalkClassic/TeamTalkBase.cpp

+2-11
Original file line numberDiff line numberDiff line change
@@ -92,16 +92,7 @@ AudioPreprocessor InitDefaultAudioPreprocessor(AudioPreprocessorType preprocesso
9292
preprocessor.ttpreprocessor.bMuteLeftSpeaker = preprocessor.ttpreprocessor.bMuteRightSpeaker = FALSE;
9393
break;
9494
case WEBRTC_AUDIOPREPROCESSOR:
95-
preprocessor.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE;
96-
preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB;
97-
preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE;
98-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB;
99-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB;
100-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC;
101-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE;
102-
preprocessor.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE;
103-
preprocessor.webrtc.noisesuppression.nLevel = DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL;
104-
preprocessor.webrtc.echocanceller.bEnable = DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE;
95+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
10596
break;
10697
}
10798
return preprocessor;
@@ -338,4 +329,4 @@ void UserCached::Sync(TTInstance* ttInst, const User& user)
338329
TT_PumpMessage(ttInst, CLIENTEVENT_USER_STATECHANGE, user.nUserID);
339330

340331
TRACE(_T("Restored ") + UserCacheID(user) + _T("\n"));
341-
}
332+
}

‎Client/iTeamTalk/iTeamTalk/UtilTT.swift

+7-9
Original file line numberDiff line numberDiff line change
@@ -181,14 +181,13 @@ let DEFAULT_SPEEXDSP_ECHO_SUPPRESSACTIVE = INT32(-15)
181181

182182
let DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE = FALSE
183183
let DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR = Float(1)
184-
let DEFAULT_WEBRTC_VAD_ENABLE = FALSE
185-
let DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE = FALSE
186184
let DEFAULT_WEBRTC_GAINCTL_ENABLE = DEFAULT_AGC_ENABLE
187185
let DEFAULT_WEBRTC_GAINDB = Float(15)
188186
let DEFAULT_WEBRTC_SAT_PROT_ENABLE = TRUE
189-
let DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB = Float(20)
190-
let DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB = Float(2)
191-
let DEFAULT_WEBRTC_MAXGAIN_DBSEC = Float(3)
187+
let DEFAULT_WEBRTC_HEADROOM_DB = Float(5)
188+
let DEFAULT_WEBRTC_MAXGAIN_DB = Float(50)
189+
let DEFAULT_WEBRTC_INITIAL_GAIN_DB = Float(15)
190+
let DEFAULT_WEBRTC_MAXGAIN_DBSEC = Float(6)
192191
let DEFAULT_WEBRTC_MAX_OUT_NOISE = Float(-50)
193192
let DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE = DEFAULT_DENOISE_ENABLE
194193
let DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL = INT32(2)
@@ -295,15 +294,14 @@ func newAudioPreprocessor(preprocessor: AudioPreprocessorType) -> AudioPreproces
295294
ap.webrtc.echocanceller.bEnable = DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE
296295
ap.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE
297296
ap.webrtc.noisesuppression.nLevel = DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL
298-
ap.webrtc.voicedetection.bEnable = DEFAULT_WEBRTC_VAD_ENABLE
299297
ap.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE
300298
ap.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB
301299
ap.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE
302-
ap.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB
303-
ap.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB
300+
ap.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = DEFAULT_WEBRTC_HEADROOM_DB
301+
ap.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = DEFAULT_WEBRTC_MAXGAIN_DB
302+
ap.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = DEFAULT_WEBRTC_INITIAL_GAIN_DB
304303
ap.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC
305304
ap.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE
306-
ap.webrtc.levelestimation.bEnable = DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE
307305
case NO_AUDIOPREPROCESSOR :
308306
fallthrough
309307
default :

‎Client/qtTeamTalk/audiopreprocessordlg.cpp

+2
Original file line numberDiff line numberDiff line change
@@ -46,6 +46,7 @@ void AudioPreprocessorDlg::showSettings()
4646
switch(m_preprocess.nPreprocessor)
4747
{
4848
case NO_AUDIOPREPROCESSOR :
49+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
4950
ui.stackedWidget->setCurrentIndex(0);
5051
setWindowTitle(tr("No Audio Preprocessor"));
5152
break;
@@ -89,6 +90,7 @@ void AudioPreprocessorDlg::slotAccepted()
8990
switch(m_preprocess.nPreprocessor)
9091
{
9192
case NO_AUDIOPREPROCESSOR :
93+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
9294
break;
9395
case SPEEXDSP_AUDIOPREPROCESSOR :
9496
m_preprocess.speexdsp.bEnableAGC = ui.agcCheckBox->isChecked();

‎Client/qtTeamTalk/common.cpp

+14-10
Original file line numberDiff line numberDiff line change
@@ -340,9 +340,10 @@ AudioPreprocessor loadAudioPreprocessor(AudioPreprocessorType preprocessortype)
340340
AudioPreprocessor preprocessor = initDefaultAudioPreprocessor(preprocessortype);
341341
switch (preprocessor.nPreprocessor)
342342
{
343-
case NO_AUDIOPREPROCESSOR:
343+
case NO_AUDIOPREPROCESSOR :
344+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
344345
break;
345-
case SPEEXDSP_AUDIOPREPROCESSOR:
346+
case SPEEXDSP_AUDIOPREPROCESSOR :
346347
preprocessor.speexdsp.bEnableAGC = ttSettings->value(SETTINGS_STREAMMEDIA_SPX_AGC_ENABLE, DEFAULT_SPEEXDSP_AGC_ENABLE).toBool();
347348
preprocessor.speexdsp.nGainLevel = ttSettings->value(SETTINGS_STREAMMEDIA_SPX_AGC_GAINLEVEL, DEFAULT_SPEEXDSP_AGC_GAINLEVEL).toInt();
348349
preprocessor.speexdsp.nMaxIncDBSec = ttSettings->value(SETTINGS_STREAMMEDIA_SPX_AGC_INC_MAXDB, DEFAULT_SPEEXDSP_AGC_INC_MAXDB).toInt();
@@ -354,7 +355,7 @@ AudioPreprocessor loadAudioPreprocessor(AudioPreprocessorType preprocessortype)
354355
preprocessor.speexdsp.nEchoSuppress = DEFAULT_SPEEXDSP_ECHO_SUPPRESS;
355356
preprocessor.speexdsp.nEchoSuppressActive = DEFAULT_SPEEXDSP_ECHO_SUPPRESSACTIVE;
356357
break;
357-
case TEAMTALK_AUDIOPREPROCESSOR:
358+
case TEAMTALK_AUDIOPREPROCESSOR :
358359
preprocessor.ttpreprocessor.bMuteLeftSpeaker = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_MUTELEFT, false).toBool();
359360
preprocessor.ttpreprocessor.bMuteRightSpeaker = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_MUTERIGHT, false).toBool();
360361
preprocessor.ttpreprocessor.nGainLevel = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_GAINLEVEL, SOUND_GAIN_DEFAULT).toInt();
@@ -363,8 +364,9 @@ AudioPreprocessor loadAudioPreprocessor(AudioPreprocessorType preprocessortype)
363364
preprocessor.webrtc.gaincontroller2.bEnable = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE, DEFAULT_WEBRTC_GAINCTL_ENABLE).toBool();
364365
preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_GAINDB, DEFAULT_WEBRTC_GAINDB).toFloat();
365366
preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE, DEFAULT_WEBRTC_SAT_PROT_ENABLE).toBool();
366-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB, DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB).toFloat();
367-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB, DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB).toFloat();
367+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB, DEFAULT_WEBRTC_HEADROOM_DB).toFloat();
368+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB, DEFAULT_WEBRTC_MAXGAIN_DB).toFloat();
369+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB, DEFAULT_WEBRTC_INITIAL_GAIN_DB).toFloat();
368370
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC, DEFAULT_WEBRTC_MAXGAIN_DBSEC).toFloat();
369371
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE, DEFAULT_WEBRTC_MAX_OUT_NOISE).toFloat();
370372
preprocessor.webrtc.noisesuppression.bEnable = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_NOISESUPPRESS_ENABLE, DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE).toBool();
@@ -380,9 +382,10 @@ void saveAudioPreprocessor(const AudioPreprocessor& preprocessor)
380382
ttSettings->setValue(SETTINGS_STREAMMEDIA_AUDIOPREPROCESSOR, preprocessor.nPreprocessor);
381383
switch (preprocessor.nPreprocessor)
382384
{
383-
case NO_AUDIOPREPROCESSOR:
385+
case NO_AUDIOPREPROCESSOR :
386+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
384387
break;
385-
case SPEEXDSP_AUDIOPREPROCESSOR:
388+
case SPEEXDSP_AUDIOPREPROCESSOR :
386389
ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_ENABLE, preprocessor.speexdsp.bEnableAGC);
387390
ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_GAINLEVEL, preprocessor.speexdsp.nGainLevel);
388391
ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_INC_MAXDB, preprocessor.speexdsp.nMaxIncDBSec);
@@ -391,7 +394,7 @@ void saveAudioPreprocessor(const AudioPreprocessor& preprocessor)
391394
ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_DENOISE_ENABLE, preprocessor.speexdsp.bEnableDenoise);
392395
ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_DENOISE_SUPPRESS, preprocessor.speexdsp.nMaxNoiseSuppressDB);
393396
break;
394-
case TEAMTALK_AUDIOPREPROCESSOR:
397+
case TEAMTALK_AUDIOPREPROCESSOR :
395398
ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_MUTELEFT, preprocessor.ttpreprocessor.bMuteLeftSpeaker);
396399
ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_MUTERIGHT, preprocessor.ttpreprocessor.bMuteRightSpeaker);
397400
ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_GAINLEVEL, preprocessor.ttpreprocessor.nGainLevel);
@@ -400,8 +403,9 @@ void saveAudioPreprocessor(const AudioPreprocessor& preprocessor)
400403
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE, preprocessor.webrtc.gaincontroller2.bEnable);
401404
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_GAINDB, preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB);
402405
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE, preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable );
403-
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB);
404-
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB);
406+
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB);
407+
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB);
408+
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB);
405409
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond);
406410
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS);
407411
ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_NOISESUPPRESS_ENABLE, preprocessor.webrtc.noisesuppression.bEnable);

‎Client/qtTeamTalk/mainwindow.cpp

+3
Original file line numberDiff line numberDiff line change
@@ -5522,6 +5522,7 @@ void MainWindow::changeMediaFileVolume(int pos)
55225522
m_mfp.audioPreprocessor.speexdsp.nGainLevel = pos;
55235523
break;
55245524
case WEBRTC_AUDIOPREPROCESSOR :
5525+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
55255526
case NO_AUDIOPREPROCESSOR :
55265527
return;
55275528
}
@@ -6452,6 +6453,7 @@ void MainWindow::slotUpdateMediaTabUI()
64526453
break;
64536454
case NO_AUDIOPREPROCESSOR :
64546455
case WEBRTC_AUDIOPREPROCESSOR :
6456+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
64556457
ui.mediaVolumeSlider->setEnabled(false);
64566458
ui.mediaVolumeSlider->setValue(0);
64576459
ui.mediaVolumeLabel->setText(tr("%1 %").arg(100));
@@ -7457,6 +7459,7 @@ void MainWindow::slotMicrophoneGainChanged(int value)
74577459
switch (preprocessor.nPreprocessor)
74587460
{
74597461
case NO_AUDIOPREPROCESSOR :
7462+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
74607463
preprocessor = initDefaultAudioPreprocessor(NO_AUDIOPREPROCESSOR);
74617464
TT_SetSoundInputPreprocessEx(ttInst, &preprocessor);
74627465
TT_SetSoundInputGainLevel(ttInst, refGain(value));

‎Client/qtTeamTalk/settings.h

+6-5
Original file line numberDiff line numberDiff line change
@@ -507,11 +507,12 @@
507507
#define SETTINGS_STREAMMEDIA_SPX_AGC_GAINMAXDB "stream-media/spxaudiopreprocessor/agc-gainmaxdb"
508508
#define SETTINGS_STREAMMEDIA_SPX_DENOISE_ENABLE "stream-media/spxaudiopreprocessor/denoise-enable"
509509
#define SETTINGS_STREAMMEDIA_SPX_DENOISE_SUPPRESS "stream-media/spxaudiopreprocessor/denoise-suppress"
510-
#define SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE "stream-media/webrtcaudiopreprocessor/gain-enable"
511-
#define SETTINGS_STREAMMEDIA_WEBRTC_GAINDB "stream-media/webrtcaudiopreprocessor/gain-db"
512-
#define SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE "stream-media/webrtcaudiopreprocessor/sat-protection-enable"
513-
#define SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB "stream-media/webrtcaudiopreprocessor/init-sat-margin-db"
514-
#define SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB "stream-media/webrtcaudiopreprocessor/extra-sat-margin-db"
510+
#define SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE "stream-media/webrtcaudiopreprocessor/gain-enable"
511+
#define SETTINGS_STREAMMEDIA_WEBRTC_GAINDB "stream-media/webrtcaudiopreprocessor/gain-db"
512+
#define SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE "stream-media/webrtcaudiopreprocessor/sat-protection-enable"
513+
#define SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB "stream-media/webrtcaudiopreprocessor/headroom-db"
514+
#define SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB "stream-media/webrtcaudiopreprocessor/maxgain-db"
515+
#define SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB "stream-media/webrtcaudiopreprocessor/initial-gain-db"
515516
#define SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC "stream-media/webrtcaudiopreprocessor/maxgain-dbsec"
516517
#define SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE "stream-media/webrtcaudiopreprocessor/max-out-noise"
517518
#define SETTINGS_STREAMMEDIA_WEBRTC_NOISESUPPRESS_ENABLE "stream-media/webrtcaudiopreprocessor/noise-suppress-enable"

‎Client/qtTeamTalk/utiltt.cpp

+5-4
Original file line numberDiff line numberDiff line change
@@ -200,16 +200,17 @@ AudioPreprocessor initDefaultAudioPreprocessor(AudioPreprocessorType preprocesso
200200
preprocessor.ttpreprocessor.bMuteLeftSpeaker = DEFAULT_TEAMTALK_MUTELEFT;
201201
preprocessor.ttpreprocessor.bMuteRightSpeaker = DEFAULT_TEAMTALK_MUTERIGHT;
202202
break;
203+
case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
204+
break;
203205
case WEBRTC_AUDIOPREPROCESSOR :
204206
preprocessor.webrtc.preamplifier.bEnable = DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE;
205207
preprocessor.webrtc.preamplifier.fFixedGainFactor = DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR;
206-
preprocessor.webrtc.levelestimation.bEnable = DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE;
207-
preprocessor.webrtc.voicedetection.bEnable = DEFAULT_WEBRTC_VAD_ENABLE;
208208
preprocessor.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE;
209209
preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB;
210210
preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE;
211-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB;
212-
preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB;
211+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = DEFAULT_WEBRTC_HEADROOM_DB;
212+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = DEFAULT_WEBRTC_INITIAL_GAIN_DB;
213+
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = DEFAULT_WEBRTC_MAXGAIN_DB;
213214
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC;
214215
preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE;
215216
preprocessor.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE;

‎Client/qtTeamTalk/utiltt.h

+10-7
Original file line numberDiff line numberDiff line change
@@ -94,15 +94,18 @@ do { \
9494

9595
#define DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE FALSE
9696
#define DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR 1.0f
97-
#define DEFAULT_WEBRTC_VAD_ENABLE FALSE
98-
#define DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE FALSE
97+
/* gain controller 2 */
9998
#define DEFAULT_WEBRTC_GAINCTL_ENABLE DEFAULT_AGC_ENABLE
100-
#define DEFAULT_WEBRTC_GAINDB 15
99+
/* gain controller 2 - fixed digital */
100+
#define DEFAULT_WEBRTC_GAINDB 0.0f
101+
/* gain controller 2 - adaptive digital */
101102
#define DEFAULT_WEBRTC_SAT_PROT_ENABLE TRUE
102-
#define DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB 20
103-
#define DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB 2
104-
#define DEFAULT_WEBRTC_MAXGAIN_DBSEC 3
105-
#define DEFAULT_WEBRTC_MAX_OUT_NOISE -50
103+
#define DEFAULT_WEBRTC_HEADROOM_DB 5.0f
104+
#define DEFAULT_WEBRTC_MAXGAIN_DB 50.0f
105+
#define DEFAULT_WEBRTC_INITIAL_GAIN_DB 15.0f
106+
#define DEFAULT_WEBRTC_MAXGAIN_DBSEC 6.0f
107+
#define DEFAULT_WEBRTC_MAX_OUT_NOISE -50.0f
108+
/* noise suppression */
106109
#define DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE DEFAULT_DENOISE_ENABLE
107110
#define DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL 2
108111
#define DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE FALSE /* requires duplex mode */

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