Replace WebRTC gaincontroller2 r4332 by r6818

Replaces 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

@@ -186,9 +186,10 @@ 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)
@@ -299,8 +300,9 @@ func newAudioPreprocessor(preprocessor: AudioPreprocessorType) -> AudioPreproces
         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

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/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

@@ -200,6 +200,8 @@ 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;
@@ -208,8 +210,9 @@ AudioPreprocessor initDefaultAudioPreprocessor(AudioPreprocessorType preprocesso
         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

@@ -96,13 +96,18 @@ do {                                                            \
 #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 */

Library/TeamTalk.NET/TeamTalk.cs

@@ -1350,14 +1350,16 @@ public struct FixedDigital
             [StructLayout(LayoutKind.Sequential, CharSet = CharSet.Unicode)]
             public struct AdaptiveDigital
             {
-                /* @brief Enable saturation protector where saturation
-                 * margin is 2 dB. */
+                /* @brief Configuration for fine tuning gain level after echo
+                 *  cancellation and noise suppression. */
                 public bool bEnable;
-                /* Range: 0 <= x <= 100. Default: 20 dB */
-                public float fInitialSaturationMarginDB;
-                /* Range: 0 <= x <= 100. Default: 2 dB */
-                public float fExtraSaturationMarginDB;
-                /* Range: 0 < x < infinite. Default: 3 dB/sec */
+                /** @brief Range: 0 <= x < infinite. Default: 5 dB */
+                public float fHeadRoomDB;
+                /** @brief Range: 0 < x < infinite. Default: 50 dB */
+                public float fMaxGainDB;
+                /** @brief Range: 0 <= x < infinite. Default: 15 dB */
+                public float fInitialGainDB;
+                /* Range: 0 < x < infinite. Default: 6 dB/sec */
                 public float fMaxGainChangeDBPerSecond;
                 /* Range: -infinite < x < 0. Default: -50 */
                 public float fMaxOutputNoiseLevelDBFS;
@@ -1397,8 +1399,9 @@ public WebRTCAudioPreprocessor(bool set_defaults)
                 gaincontroller2.fixeddigital.fGainDB = WebRTCConstants.DEFAULT_WEBRTC_GAINDB;
 
                 gaincontroller2.adaptivedigital.bEnable = WebRTCConstants.DEFAULT_WEBRTC_SAT_PROT_ENABLE;
-                gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = WebRTCConstants.DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB;
-                gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = WebRTCConstants.DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB;
+                gaincontroller2.adaptivedigital.fHeadRoomDB = WebRTCConstants.DEFAULT_WEBRTC_HEADROOM_DB;
+                gaincontroller2.adaptivedigital.fMaxGainDB = WebRTCConstants.DEFAULT_WEBRTC_MAXGAIN_DB;
+                gaincontroller2.adaptivedigital.fInitialGainDB = WebRTCConstants.DEFAULT_WEBRTC_INITIAL_GAIN_DB;
                 gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = WebRTCConstants.DEFAULT_WEBRTC_MAXGAIN_DBSEC;
                 gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = WebRTCConstants.DEFAULT_WEBRTC_MAX_OUT_NOISE;
 
@@ -1420,8 +1423,9 @@ public WebRTCAudioPreprocessor(bool set_defaults)
                 gaincontroller2.fixeddigital.fGainDB = 0.0f;
 
                 gaincontroller2.adaptivedigital.bEnable = false;
-                gaincontroller2.adaptivedigital.fInitialSaturationMarginDB = 0.0f;
-                gaincontroller2.adaptivedigital.fExtraSaturationMarginDB = 0.0f;
+                gaincontroller2.adaptivedigital.fHeadRoomDB = 0.0f;
+                gaincontroller2.adaptivedigital.fMaxGainDB = 0.0f;
+                gaincontroller2.adaptivedigital.fInitialGainDB = 0.0f;
                 gaincontroller2.adaptivedigital.fMaxGainChangeDBPerSecond = 0.0f;
                 gaincontroller2.adaptivedigital.fMaxOutputNoiseLevelDBFS = 0.0f;
 
@@ -1440,9 +1444,10 @@ public struct WebRTCConstants
         public const bool DEFAULT_WEBRTC_GAINCTL_ENABLE = false;
         public const float DEFAULT_WEBRTC_GAINDB = 15;
         public const bool DEFAULT_WEBRTC_SAT_PROT_ENABLE = false;
-        public const float DEFAULT_WEBRTC_INIT_SAT_MARGIN_DB = 20;
-        public const float DEFAULT_WEBRTC_EXTRA_SAT_MARGIN_DB = 2;
-        public const float DEFAULT_WEBRTC_MAXGAIN_DBSEC = 3;
+        public const float DEFAULT_WEBRTC_HEADROOM_DB = 5;
+        public const float DEFAULT_WEBRTC_MAXGAIN_DB = 50;
+        public const float DEFAULT_WEBRTC_INITIAL_GAIN_DB = 15;
+        public const float DEFAULT_WEBRTC_MAXGAIN_DBSEC = 6;
         public const float DEFAULT_WEBRTC_MAX_OUT_NOISE = -50;
         public const bool DEFAULT_WEBRTC_NOISESUPPRESS_ENABLE = false;
         public const int DEFAULT_WEBRTC_NOISESUPPRESS_LEVEL = 2;