TeamTalk 5 C-API DLL
Version 5.11A
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WebRTC's audio preprocessor. More...
#include <TeamTalk.h>
Public Attributes | |
struct { | |
TTBOOL bEnable | |
Enable pre-amplifier. Replacement for TT_SetSoundInputGainLevel() More... | |
float fFixedGainFactor | |
Gain factor. Default: 1. More... | |
} | preamplifier |
Configuration of WebRTC pre-amplifier. More... | |
struct { | |
TTBOOL bEnable | |
Enable WebRTC echo canceller. The WebRTC echo canceller requires sound input and output devices are initialized using TT_InitSoundDuplexDevices(). This is because both input and output device must use the same sample rate. More... | |
} | echocanceller |
Configuration of WebRTC's echo canceller. See also TT_SetSoundDeviceEffects() More... | |
struct { | |
TTBOOL bEnable | |
Enable WebRTC noise suppression. More... | |
INT32 nLevel | |
Noise suppression level. 0 = Low, 1 = Moderate, 2 = High, 3 = VeryHigh. Default: 1. More... | |
} | noisesuppression |
Configuration of WebRTC's noise suppression. See also SpeexDSP. More... | |
struct { | |
TTBOOL bEnable | |
Use WebRTC's voice detection to trigger CLIENTEVENT_VOICE_ACTIVATION. More... | |
} | voicedetection |
Configuration of WebRTC's voice detection. More... | |
struct { | |
TTBOOL bEnable | |
Enable WebRTC's fixed digital gain. WebRTC's automatic gain control (AGC) More... | |
struct { | |
float fGainDB | |
Gain level in dB. Range: 0 <= x < 50. Default: 0. More... | |
} fixeddigital | |
Gain level for AGC. Only active when bEnable is true. More... | |
struct { | |
TTBOOL bEnable | |
float fInitialSaturationMarginDB | |
float fExtraSaturationMarginDB | |
float fMaxGainChangeDBPerSecond | |
float fMaxOutputNoiseLevelDBFS | |
} adaptivedigital | |
Configuration for fine tuning gain level. More... | |
} | gaincontroller2 |
Configuration of WebRTC's gain controller 2 for AGC. More... | |
struct { | |
TTBOOL bEnable | |
Enable level estimater. When enabled TT_GetSoundInputLevel() will return a value based on WebRTC's level estimater. A WebRTC level estimater value of 0 will result in SOUND_VU_MAX and level estimater value of 127 will return SOUND_VU_MIN. More... | |
} | levelestimation |
Configuration of WebRTC's level estimater. More... | |
WebRTC's audio preprocessor.
Use WebRTC's audio preprocessor, https://webrtc.org
Note that WebRTC's can only operate on 10 msec audio frame, so nTxIntervalMSec
in AudioCodec must a multiple of 10.
WebRTCAudioPreprocessor is recommended to TT_SetSoundDeviceEffects() on desktop platforms.
Activate WebRTCAudioPreprocessor by calling TT_SetSoundInputPreprocessEx().
Definition at line 1311 of file TeamTalk.h.
TTBOOL WebRTCAudioPreprocessor::bEnable |
Enable pre-amplifier. Replacement for TT_SetSoundInputGainLevel()
Enable level estimater. When enabled TT_GetSoundInputLevel() will return a value based on WebRTC's level estimater. A WebRTC level estimater value of 0 will result in SOUND_VU_MAX and level estimater value of 127 will return SOUND_VU_MIN.
Enable WebRTC's fixed digital gain. WebRTC's automatic gain control (AGC)
Use WebRTC's voice detection to trigger CLIENTEVENT_VOICE_ACTIVATION.
Enable WebRTC noise suppression.
Enable WebRTC echo canceller. The WebRTC echo canceller requires sound input and output devices are initialized using TT_InitSoundDuplexDevices(). This is because both input and output device must use the same sample rate.
TT_EnableVoiceActivation() must still be called to activate voice detection event CLIENTEVENT_VOICE_ACTIVATION.
Enabling WebRTC's voice detection invalidates use of TT_SetVoiceActivationLevel()
Definition at line 1318 of file TeamTalk.h.
float WebRTCAudioPreprocessor::fFixedGainFactor |
Gain factor. Default: 1.
Definition at line 1320 of file TeamTalk.h.
struct { ... } WebRTCAudioPreprocessor::preamplifier |
Configuration of WebRTC pre-amplifier.
struct { ... } WebRTCAudioPreprocessor::echocanceller |
Configuration of WebRTC's echo canceller. See also TT_SetSoundDeviceEffects()
INT32 WebRTCAudioPreprocessor::nLevel |
Noise suppression level. 0 = Low, 1 = Moderate, 2 = High, 3 = VeryHigh. Default: 1.
Definition at line 1341 of file TeamTalk.h.
struct { ... } WebRTCAudioPreprocessor::noisesuppression |
Configuration of WebRTC's noise suppression. See also SpeexDSP.
struct { ... } WebRTCAudioPreprocessor::voicedetection |
Configuration of WebRTC's voice detection.
float WebRTCAudioPreprocessor::fGainDB |
Gain level in dB. Range: 0 <= x < 50. Default: 0.
Definition at line 1370 of file TeamTalk.h.
struct { ... } WebRTCAudioPreprocessor::fixeddigital |
Gain level for AGC. Only active when bEnable
is true.
float WebRTCAudioPreprocessor::fInitialSaturationMarginDB |
Definition at line 1379 of file TeamTalk.h.
float WebRTCAudioPreprocessor::fExtraSaturationMarginDB |
Definition at line 1381 of file TeamTalk.h.
float WebRTCAudioPreprocessor::fMaxGainChangeDBPerSecond |
Definition at line 1383 of file TeamTalk.h.
float WebRTCAudioPreprocessor::fMaxOutputNoiseLevelDBFS |
Definition at line 1385 of file TeamTalk.h.
struct { ... } WebRTCAudioPreprocessor::adaptivedigital |
Configuration for fine tuning gain level.
struct { ... } WebRTCAudioPreprocessor::gaincontroller2 |
Configuration of WebRTC's gain controller 2 for AGC.
struct { ... } WebRTCAudioPreprocessor::levelestimation |
Configuration of WebRTC's level estimater.