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;

Client/TeamTalkClassic/TeamTalkBase.cpp

@@ -92,16 +92,7 @@ AudioPreprocessor InitDefaultAudioPreprocessor(AudioPreprocessorType preprocesso
         preprocessor.ttpreprocessor.bMuteLeftSpeaker = preprocessor.ttpreprocessor.bMuteRightSpeaker = FALSE;
         break;
     case WEBRTC_AUDIOPREPROCESSOR:
-        preprocessor.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE;
-        preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE;
-        preprocessor.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE;
-        preprocessor.webrtc.noisesuppression.nLevel = DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL;
-        preprocessor.webrtc.echocanceller.bEnable = DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE;
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         break;
     }
     return preprocessor;
@@ -338,4 +329,4 @@ void UserCached::Sync(TTInstance* ttInst, const User& user)
     TT_PumpMessage(ttInst, CLIENTEVENT_USER_STATECHANGE, user.nUserID);
 
     TRACE(_T("Restored ") + UserCacheID(user) + _T("\n"));
-}
+}

Client/iTeamTalk/iTeamTalk/UtilTT.swift

@@ -181,14 +181,13 @@ let DEFAULT_SPEEXDSP_ECHO_SUPPRESSACTIVE = INT32(-15)
 
 let DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE = FALSE
 let DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR = Float(1)
-let DEFAULT_WEBRTC_VAD_ENABLE = FALSE
-let DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE = FALSE
 let DEFAULT_WEBRTC_GAINCTL_ENABLE = DEFAULT_AGC_ENABLE
 let DEFAULT_WEBRTC_GAINDB = Float(15)
 let DEFAULT_WEBRTC_SAT_PROT_ENABLE = TRUE
-let DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB = Float(20)
-let DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB = Float(2)
-let DEFAULT_WEBRTC_MAXGAIN_DBSEC = Float(3)
+let DEFAULT_WEBRTC_HEADROOM_DB = Float(5)
+let DEFAULT_WEBRTC_MAXGAIN_DB = Float(50)
+let DEFAULT_WEBRTC_INITIAL_GAIN_DB = Float(15)
+let DEFAULT_WEBRTC_MAXGAIN_DBSEC = Float(6)
 let DEFAULT_WEBRTC_MAX_OUT_NOISE = Float(-50)
 let DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE = DEFAULT_DENOISE_ENABLE
 let DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL = INT32(2)
@@ -295,15 +294,14 @@ func newAudioPreprocessor(preprocessor: AudioPreprocessorType) -> AudioPreproces
         ap.webrtc.echocanceller.bEnable = DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE
         ap.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE
         ap.webrtc.noisesuppression.nLevel = DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL
-        ap.webrtc.voicedetection.bEnable = DEFAULT_WEBRTC_VAD_ENABLE
         ap.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE
         ap.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB
         ap.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE
-        ap.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB
-        ap.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB
+        ap.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = DEFAULT_WEBRTC_HEADROOM_DB
+        ap.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = DEFAULT_WEBRTC_MAXGAIN_DB
+        ap.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = DEFAULT_WEBRTC_INITIAL_GAIN_DB
         ap.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC
         ap.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE
-        ap.webrtc.levelestimation.bEnable = DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE
     case NO_AUDIOPREPROCESSOR :
         fallthrough
     default :

Client/qtTeamTalk/audiopreprocessordlg.cpp

@@ -46,6 +46,7 @@ void AudioPreprocessorDlg::showSettings()
     switch(m_preprocess.nPreprocessor)
     {
     case NO_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         ui.stackedWidget->setCurrentIndex(0);
         setWindowTitle(tr("No Audio Preprocessor"));
         break;
@@ -89,6 +90,7 @@ void AudioPreprocessorDlg::slotAccepted()
     switch(m_preprocess.nPreprocessor)
     {
     case NO_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         break;
     case SPEEXDSP_AUDIOPREPROCESSOR :
         m_preprocess.speexdsp.bEnableAGC = ui.agcCheckBox->isChecked();

Client/qtTeamTalk/common.cpp

@@ -340,9 +340,10 @@ AudioPreprocessor loadAudioPreprocessor(AudioPreprocessorType preprocessortype)
     AudioPreprocessor preprocessor = initDefaultAudioPreprocessor(preprocessortype);
     switch (preprocessor.nPreprocessor)
     {
-    case NO_AUDIOPREPROCESSOR:
+    case NO_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         break;
-    case SPEEXDSP_AUDIOPREPROCESSOR:
+    case SPEEXDSP_AUDIOPREPROCESSOR :
         preprocessor.speexdsp.bEnableAGC = ttSettings->value(SETTINGS_STREAMMEDIA_SPX_AGC_ENABLE, DEFAULT_SPEEXDSP_AGC_ENABLE).toBool();
         preprocessor.speexdsp.nGainLevel = ttSettings->value(SETTINGS_STREAMMEDIA_SPX_AGC_GAINLEVEL, DEFAULT_SPEEXDSP_AGC_GAINLEVEL).toInt();
         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)
         preprocessor.speexdsp.nEchoSuppress = DEFAULT_SPEEXDSP_ECHO_SUPPRESS;
         preprocessor.speexdsp.nEchoSuppressActive = DEFAULT_SPEEXDSP_ECHO_SUPPRESSACTIVE;
         break;
-    case TEAMTALK_AUDIOPREPROCESSOR:
+    case TEAMTALK_AUDIOPREPROCESSOR :
         preprocessor.ttpreprocessor.bMuteLeftSpeaker = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_MUTELEFT, false).toBool();
         preprocessor.ttpreprocessor.bMuteRightSpeaker = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_MUTERIGHT, false).toBool();
         preprocessor.ttpreprocessor.nGainLevel = ttSettings->value(SETTINGS_STREAMMEDIA_TTAP_GAINLEVEL, SOUND_GAIN_DEFAULT).toInt();
@@ -363,8 +364,9 @@ AudioPreprocessor loadAudioPreprocessor(AudioPreprocessorType preprocessortype)
         preprocessor.webrtc.gaincontroller2.bEnable = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE, DEFAULT_WEBRTC_GAINCTL_ENABLE).toBool();
         preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_GAINDB, DEFAULT_WEBRTC_GAINDB).toFloat();
         preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE, DEFAULT_WEBRTC_SAT_PROT_ENABLE).toBool();
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB, DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB).toFloat();
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB, DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB).toFloat();
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB, DEFAULT_WEBRTC_HEADROOM_DB).toFloat();
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB, DEFAULT_WEBRTC_MAXGAIN_DB).toFloat();
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB, DEFAULT_WEBRTC_INITIAL_GAIN_DB).toFloat();
         preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC, DEFAULT_WEBRTC_MAXGAIN_DBSEC).toFloat();
         preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = ttSettings->value(SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE, DEFAULT_WEBRTC_MAX_OUT_NOISE).toFloat();
         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)
     ttSettings->setValue(SETTINGS_STREAMMEDIA_AUDIOPREPROCESSOR, preprocessor.nPreprocessor);
     switch (preprocessor.nPreprocessor)
     {
-    case NO_AUDIOPREPROCESSOR:
+    case NO_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         break;
-    case SPEEXDSP_AUDIOPREPROCESSOR:
+    case SPEEXDSP_AUDIOPREPROCESSOR :
         ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_ENABLE, preprocessor.speexdsp.bEnableAGC);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_GAINLEVEL, preprocessor.speexdsp.nGainLevel);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_AGC_INC_MAXDB, preprocessor.speexdsp.nMaxIncDBSec);
@@ -391,7 +394,7 @@ void saveAudioPreprocessor(const AudioPreprocessor& preprocessor)
         ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_DENOISE_ENABLE, preprocessor.speexdsp.bEnableDenoise);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_SPX_DENOISE_SUPPRESS, preprocessor.speexdsp.nMaxNoiseSuppressDB);
         break;
-    case TEAMTALK_AUDIOPREPROCESSOR:
+    case TEAMTALK_AUDIOPREPROCESSOR :
         ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_MUTELEFT, preprocessor.ttpreprocessor.bMuteLeftSpeaker);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_MUTERIGHT, preprocessor.ttpreprocessor.bMuteRightSpeaker);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_TTAP_GAINLEVEL, preprocessor.ttpreprocessor.nGainLevel);
@@ -400,8 +403,9 @@ void saveAudioPreprocessor(const AudioPreprocessor& preprocessor)
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE, preprocessor.webrtc.gaincontroller2.bEnable);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_GAINDB, preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE, preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable );
-        ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB);
-        ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB);
+        ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB);
+        ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB);
+        ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB, preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE, preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS);
         ttSettings->setValue(SETTINGS_STREAMMEDIA_WEBRTC_NOISESUPPRESS_ENABLE, preprocessor.webrtc.noisesuppression.bEnable);

Client/qtTeamTalk/mainwindow.cpp

@@ -5522,6 +5522,7 @@ void MainWindow::changeMediaFileVolume(int pos)
         m_mfp.audioPreprocessor.speexdsp.nGainLevel = pos;
         break;
     case WEBRTC_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
     case NO_AUDIOPREPROCESSOR :
         return;
     }
@@ -6452,6 +6453,7 @@ void MainWindow::slotUpdateMediaTabUI()
         break;
     case NO_AUDIOPREPROCESSOR :
     case WEBRTC_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         ui.mediaVolumeSlider->setEnabled(false);
         ui.mediaVolumeSlider->setValue(0);
         ui.mediaVolumeLabel->setText(tr("%1 %").arg(100));
@@ -7457,6 +7459,7 @@ void MainWindow::slotMicrophoneGainChanged(int value)
     switch (preprocessor.nPreprocessor)
     {
     case NO_AUDIOPREPROCESSOR :
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
         preprocessor = initDefaultAudioPreprocessor(NO_AUDIOPREPROCESSOR);
         TT_SetSoundInputPreprocessEx(ttInst, &preprocessor);
         TT_SetSoundInputGainLevel(ttInst, refGain(value));

Client/qtTeamTalk/settings.h

@@ -507,11 +507,12 @@
 #define SETTINGS_STREAMMEDIA_SPX_AGC_GAINMAXDB      "stream-media/spxaudiopreprocessor/agc-gainmaxdb"
 #define SETTINGS_STREAMMEDIA_SPX_DENOISE_ENABLE     "stream-media/spxaudiopreprocessor/denoise-enable"
 #define SETTINGS_STREAMMEDIA_SPX_DENOISE_SUPPRESS   "stream-media/spxaudiopreprocessor/denoise-suppress"
-#define SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE          "stream-media/webrtcaudiopreprocessor/gain-enable"
-#define SETTINGS_STREAMMEDIA_WEBRTC_GAINDB                  "stream-media/webrtcaudiopreprocessor/gain-db"
-#define SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE         "stream-media/webrtcaudiopreprocessor/sat-protection-enable"
-#define SETTINGS_STREAMMEDIA_WEBRTC_INIT_SAT_MARGIN_DB      "stream-media/webrtcaudiopreprocessor/init-sat-margin-db"
-#define SETTINGS_STREAMMEDIA_WEBRTC_EXTRA_SAT_MARGIN_DB     "stream-media/webrtcaudiopreprocessor/extra-sat-margin-db"
+#define SETTINGS_STREAMMEDIA_WEBRTC_GAINCTL_ENABLE  "stream-media/webrtcaudiopreprocessor/gain-enable"
+#define SETTINGS_STREAMMEDIA_WEBRTC_GAINDB          "stream-media/webrtcaudiopreprocessor/gain-db"
+#define SETTINGS_STREAMMEDIA_WEBRTC_SAT_PROT_ENABLE "stream-media/webrtcaudiopreprocessor/sat-protection-enable"
+#define SETTINGS_STREAMMEDIA_WEBRTC_HEADROOM_DB     "stream-media/webrtcaudiopreprocessor/headroom-db"
+#define SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DB      "stream-media/webrtcaudiopreprocessor/maxgain-db"
+#define SETTINGS_STREAMMEDIA_WEBRTC_INITGAIN_DB      "stream-media/webrtcaudiopreprocessor/initial-gain-db"
 #define SETTINGS_STREAMMEDIA_WEBRTC_MAXGAIN_DBSEC           "stream-media/webrtcaudiopreprocessor/maxgain-dbsec"
 #define SETTINGS_STREAMMEDIA_WEBRTC_MAX_OUT_NOISE           "stream-media/webrtcaudiopreprocessor/max-out-noise"
 #define SETTINGS_STREAMMEDIA_WEBRTC_NOISESUPPRESS_ENABLE    "stream-media/webrtcaudiopreprocessor/noise-suppress-enable"

Client/qtTeamTalk/utiltt.cpp

@@ -200,16 +200,17 @@ AudioPreprocessor initDefaultAudioPreprocessor(AudioPreprocessorType preprocesso
         preprocessor.ttpreprocessor.bMuteLeftSpeaker = DEFAULT_TEAMTALK_MUTELEFT;
         preprocessor.ttpreprocessor.bMuteRightSpeaker = DEFAULT_TEAMTALK_MUTERIGHT;
         break;
+    case WEBRTC_AUDIOPREPROCESSOR_OBSOLETE_R4332 :
+        break;
     case WEBRTC_AUDIOPREPROCESSOR :
         preprocessor.webrtc.preamplifier.bEnable = DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE;
         preprocessor.webrtc.preamplifier.fFixedGainFactor = DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR;
-        preprocessor.webrtc.levelestimation.bEnable = DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE;
-        preprocessor.webrtc.voicedetection.bEnable = DEFAULT_WEBRTC_VAD_ENABLE;
         preprocessor.webrtc.gaincontroller2.bEnable = DEFAULT_WEBRTC_GAINCTL_ENABLE;
         preprocessor.webrtc.gaincontroller2.fixeddigital.fGainDB = DEFAULT_WEBRTC_GAINDB;
         preprocessor.webrtc.gaincontroller2.adaptivedigital.bEnable = DEFAULT_WEBRTC_SAT_PROT_ENABLE;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB;
-        preprocessor.webrtc.gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB;
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fHeadRoomDB = DEFAULT_WEBRTC_HEADROOM_DB;
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fInitialGainDB = DEFAULT_WEBRTC_INITIAL_GAIN_DB;
+        preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainDB = DEFAULT_WEBRTC_MAXGAIN_DB;
         preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = DEFAULT_WEBRTC_MAXGAIN_DBSEC;
         preprocessor.webrtc.gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = DEFAULT_WEBRTC_MAX_OUT_NOISE;
         preprocessor.webrtc.noisesuppression.bEnable = DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE;

Client/qtTeamTalk/utiltt.h

@@ -94,15 +94,18 @@ do {                                                            \
 
 #define DEFAULT_WEBRTC_PREAMPLIFIER_ENABLE      FALSE
 #define DEFAULT_WEBRTC_PREAMPLIFIER_GAINFACTOR  1.0f
-#define DEFAULT_WEBRTC_VAD_ENABLE               FALSE
-#define DEFAULT_WEBRTC_LEVELESTIMATION_ENABLE   FALSE
+/* gain controller 2 */
 #define DEFAULT_WEBRTC_GAINCTL_ENABLE           DEFAULT_AGC_ENABLE
-#define DEFAULT_WEBRTC_GAINDB                   15
+/* gain controller 2 - fixed digital */
+#define DEFAULT_WEBRTC_GAINDB                   0.0f
+/* gain controller 2 - adaptive digital */
 #define DEFAULT_WEBRTC_SAT_PROT_ENABLE          TRUE
-#define DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB       20
-#define DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB      2
-#define DEFAULT_WEBRTC_MAXGAIN_DBSEC            3
-#define DEFAULT_WEBRTC_MAX_OUT_NOISE            -50
+#define DEFAULT_WEBRTC_HEADROOM_DB              5.0f
+#define DEFAULT_WEBRTC_MAXGAIN_DB               50.0f
+#define DEFAULT_WEBRTC_INITIAL_GAIN_DB          15.0f
+#define DEFAULT_WEBRTC_MAXGAIN_DBSEC            6.0f
+#define DEFAULT_WEBRTC_MAX_OUT_NOISE            -50.0f
+/* noise suppression */
 #define DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE     DEFAULT_DENOISE_ENABLE
 #define DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL      2
 #define DEFAULT_WEBRTC_ECHO_CANCEL_ENABLE       FALSE /* requires duplex mode */