VACore.ini.proto

#
#  -------------------------------------------------------------------------------------------
#
#    VVV        VVV A
#     VVV      VVV AAA        Virtual Acoustics (VA)
#      VVV    VVV   AAA       Real-time auralization for virtual reality    
#       VVV  VVV     AAA
#        VVVVVV       AAA     (c) Copyright Institute of Technical Acoustics (ITA)
#         VVVV         AAA        RWTH Aachen University (http://www.akustik.rwth-aachen.de)
#
#  -------------------------------------------------------------------------------------------
#
#  WARNING: This configuration file for the VACore has been automatically created by CMake.
# 			You can move this file and change the argument path accordingly, otherwise
#			your changes will be overwritten. Alternatively, you can directly modify  
#			the prototype file 'VACore.ini.proto' and re-run CMake.
#


[Paths]

# Any entry value will be added to the search path list, but only if existing! They will also be made available as macros (see below).
# If a file can not be found during runtime by local name, the core will try to locate it using the paths in ascending name order.
# Macros are not substituted, here. However, it is recommended to use AddSearchPath during runtime, if you have individual directories
# you want to add.

# Relative configuration path (with some more hardware setup files)
conf_dir = conf
conf_dir_dev = @CMAKE_CURRENT_SOURCE_DIR@/conf

# Relative data path (with some common files)
data_dir = data
data_dir_dev = @CMAKE_CURRENT_SOURCE_DIR@/data
raven_data = @ITA_RAVEN_DATA_PATH@

# Absolute data path with further files (big data, not shipped with VA)
big_data_dir = @ITA_VACORE_BIG_DATA_DIR@

# Inside scene data directory
InsideSceneData = @INSIDE_SCENE_DATA@

# Path for TTS Voices of CereVoice
voices_dir = data/Voices


[Files]

# List further config files to be appended to this configuration (also looking at paths)

#VRLabSetup = VASetup.VRLab.ini


[Macros]

# Macros can be defined and will be replaced with given value by the core. Usage: "$(MyMacroName)/file.abc" -> "MyValue/file.abc"
# Macros are substituted forwardly by key name order (use with care), otherwise stay untouched: A = B; C = $(A) -> $(C) is B

DemoSound = WelcomeToVA.wav
DefaultHRIR = ITA_Artificial_Head_5x5_44kHz_128.v17.ir.daff
HumanDir = Singer.v17.ms.daff
Trumpet = Trumpet1.v17.ms.daff

# Define some other macros (examples)
ProjectName = MyVirtualAcousticsProject


[Debug]

# Record device input and store to hard drive (will record every input channel)
InputRecordEnabled = false
InputRecordFilePath = $(ProjectName)_in.wav

# Record device output and store to hard drive (will record every output channel)
OutputRecordEnabled = false
OutputRecordFilePath = $(ProjectName)_out.wav

# Set log level: 0 = quiet; 1 = errors; 2 = warnings (default); 3 = info; 4 = verbose; 5 = trace;
LogLevel = @ITA_VACORE_DEFAULT_DEBUG_LEVEL@

# Controls the core update rate for pushed events (like level meters)
TriggerUpdateMilliseconds = 100

[Audio driver]

# MANDATORY: Audio driver backend (ASIO|Portaudio|Virtual)
#Driver = Virtual
#Driver = ASIO
Driver = Portaudio

# MANDATORY: Audio device ( e.g. ASIO4ALL v2, ASIO Hammerfall DSP, Portaudio 'default', 0,1,2,3,..., virtual user 'Trigger' )
#Device = ASIO4ALL v2
#Device = ASIO Hammerfall DSP
#Device = ASIO Fireface USB
#Device = ASIO MADIface USB
#Device = Focusrite USB 2.0 Audio Driver
#Device = M-Audio Fast Track Ultra ASIO
#Device = Yamaha Steinberg USB ASIO
Device = default
#Device = Trigger

# MANDATORY: Sampling rate [Hz]
Samplerate = 44100

# OPTIONAL: Buffer size / block length used for audio streaming [Samples]
# AUTO will determine the buffer size from the audio device automatically [recommended, set in sound card driver configuration]
# Note: Adjust this to your latency requirements
BufferSize = AUTO

# Define number of output channels manually (e.g. for virtual device)
OutputChannels = AUTO


[Calibration]

# The amplitude calibration mode either sets the internal conversion from
# sound pressure to an electrical or digital amplitude signal (audio stream)
# to 94dB (default) or to 124dB. The rendering modules will use this calibration
# mode to calculate from physical values to an amplitude that can be forwarded
# to the reproduction modules. If a reproduction module operates in calibrated
# mode, the resulting physical sound pressure at receiver location can be maintained.
DefaultAmplitudeCalibrationMode = 94dB

# Set the minimum allowed distance (m) to a sound source (point source can get infinitely loud).
# This can also be used if sound sources appear too loud near-by, but in the limiting range this
# rendering will not be physically correct.
DefaultMinimumDistance = 0.25

# The default distance is used when spherical spreading is deactivated
DefaultDistance = 2.0


[HomogeneousMedium]

DefaultSoundSpeed = 344.0 # m/s
DefaultStaticPressure = 101125.0 # [Pa]
DefaultTemperature = 20.0 # [Degree centigrade]
DefaultRelativeHumidity = 20.0 # [Percent]
DefaultShiftSpeed = 0.0, 0.0, 0.0 # 3D vector in m/s


[Renderer:MyBinauralFreeField]
Enabled = true
Class = BinauralFreeField
Reproductions = MyTalkthroughHeadphones
OutputDetectorEnabled = false
RecordOutputEnabled = false
RecordOutputFilePath = $(ProjectName)_Renderer_MyBinauralFreeField_Output.wav
HRIRFilterLength = 256
MotionModelNumHistoryKeys = 10000
MotionModelWindowSize = 0.1
MotionModelWindowDelay = 0.1
MotionModelLogInputSources = false
MotionModelLogEstimatedOutputSources = false
MotionModelLogInputListener = false
MotionModelLogEstimatedOutputListener = false
SwitchingAlgorithm = linear

[Renderer:MyAmbientMixer]
class = AmbientMixer
Description = Low-cost renderer to make sound audible without spatializations
Enabled = false
OutputGroup = MyDesktopHP
Reproductions = MyTalkthroughHeadphones

[Renderer:MyBinauralArtificialReverb]
Class = BinauralArtificialReverb
Enabled = false
Reproductions = MyTalkthroughHeadphones
ReverberationTime = 0.71
RoomVolume = 200
RoomSurfaceArea = 88
MaxReverbFilterLengthSamples = 88200
PositionThreshold = 1.0
AngleThresholdDegree = 30
SoundPowerCorrectionFactor = 0.05
TimeSlotResolution = 0.005
MaxReflectionDensity = 12000.0
ScatteringCoefficient = 0.1

[Renderer:MyBinauralRoomAcoustics]
Class = BinauralRoomAcoustics
Enabled = false
Reproductions = MyTalkthroughHeadphones
# Setup options: Local, Remote, Hybrid
Setup = Local
ServerIP = PC-SEACEN
HybridLocalTasks = DS
HybridRemoteTasks = ER_IS, DD_RT
RavenDataBasePath = $(raven_data)
# Task processing (Timeout = with desired update rate, for resource efficient processing; EventSync = process on request (for sporadic updates); Continuous = update as often as possible, for standalone server)
TaskProcessing = Timeout
# Desired update rates in Hz, may lead to resource problems
UpdateRateDS = 12.0
UpdateRateER = 4.2
UpdateRateDD = 0.08
DirectSoundPowerCorrectionFactor = 0.05

[Renderer:MyPrototypeFreeField]
Class = PrototypeFreeField
Enabled = false
Reproductions = MyTalkthroughHeadphones
MotionModelNumHistoryKeys = 10000
MotionModelWindowSize = 0.2
MotionModelWindowDelay = 0.1
MotionModelLogInputSources = false
MotionModelLogEstimatedOutputSources = false
DumpListeners = false
DumpListenersGain = 1.0
SwitchingAlgorithm = linear

[Renderer:MyHearingAidRenderer]
Class = PrototypeHearingAid
Enabled = false
Reproductions = HearingAidTalkthrough
#Reproductions = MyTalkthroughHeadphones
UpdateRateDS = 120.0
UpdateRateIS = 20.0
UpdateRateRT = 1.0
RenderingGain = 1.0
RenderingDelayInMs = 10.0
HRIRFilterLength = 256
# Dump individual listener signals
DumpListeners = false
# Gain for listener dumping [Factor]
DumpListenersGain = 10

[Renderer:MyGenericRenderer]
Class = PrototypeGenericPath
Enabled = false
Reproductions = MyTalkthroughHeadphones
NumChannels = 2
IRFilterLengthSamples = 88200
OutputMonitoring = true
RecordOutputEnabled = false
RecordOutputFilePath = $(ProjectName)_Renderer_MyGenericRenderer_Output.wav

[Renderer:MyAirTrafficNoiseRenderer]
Class = BinauralAirTrafficNoise
Enabled = false
Reproductions = MyTalkthroughHeadphones


[Renderer:MyVBAPFreefield]
Class = VBAPFreeField
Enabled = false
Output = VRLab_Horizontal_LS
Reproductions = MixdownHeadphones


[Renderer:MyDummyRenderer]
class = PrototypeDummy
Description = Dummy renderer for testing and benchmarking
Enabled = false
OutputGroup = MyDesktopHP
Reproductions = MyTalkthroughHeadphones


# --= Audio reproduction =--

# Here you write down the audio reproduction modules that are created after VA starts.
# Each section has the format [Reproduction:<ID>] (<ID> must be unique and is freely chooseable).


# Warum diese Schicht mit Eingngen und Ausgngen?
#- Vernnftige Gruppierung fr Meters
#- Abstraktion der Renderer und Reproductions von physikalischen Kanlen
#- Zusatzfunktionalitt unbh. von Renderer und Reproductions: Equalization
#
#- Diese Inputs mssen als Signalquellen im Core erscheinen


[Reproduction:MyTalkthroughHeadphones]
Class = Talkthrough
Enabled = true
Name = Generic talkthrough to output group
Outputs = MyDesktopHP
InputDetectorEnabled = false
OutputDetectorEnabled = false
RecordInputEnabled = false
RecordInputFilePath = $(ProjectName)_Reproduction_MyTalkthroughHeadphones_Input.wav
RecordOutputEnabled = false
RecordOutputFilePath = $(ProjectName)_Reproduction_MyTalkthroughHeadphones_Output.wav

[Reproduction:MySubwooferMixer]
Class = LowFrequencyMixer 
Enabled = false
Name = Generic low frequency (subwoofer) loudspeaker mixer
Outputs = Cave_SW
MixingChannels = ALL

[Reproduction:MyHD600]
Class = Headphones
Enabled = false
# Headphone impulse response inverse file path (normalized)
HpIRInvFile = HPEQ/HD600_all_eq_128_stereo.wav
#HpIRInvFilterLength = 22050
# Headphone impulse response inverse gain for calibration ( HpIR * HpIRInv == 0dB )
HpIRInvCalibrationGainDecibel = 0.1
Name = Equalized Sennheiser HD600 headphones
Outputs = MyDesktopHP

[Reproduction:MyNCTC]
Class = NCTC
Enabled = false
Name = Crosstalk cancellation for N loudspeaker
Outputs = MyDesktopLS
TrackedListenerID = 1
# algorithm: reg|...
Algorithm = reg
RegularizationBeta = 0.01
CTCFilterLength = 4096
DelaySamples = 2048
UseTrackedListenerHRIR = false
CTCDefaultHRIR = $(DefaultHRIR)
Optimization = OPTIMIZATION_NONE

[Reproduction:MyAmbisonics]
Class = HOA
Enabled = false
Name = Higher-Order Ambisonics
Enabled = false
Algorithm = HOA
Outputs = VRLab_Horizontal_LS

[Reproduction:MyMixdownHeadphones]
Class = BinauralMixdown
Enabled = false
Name = Binaural mixdown of virtual loudspeaker setup using HRIR techniques
Outputs = MyDesktopHP
VirtualOutput = MyDesktopLS
TrackedListenerID = 1
HRIRFilterLength = 128

[Reproduction:HearingAidTalkthrough]
Class = Talkthrough
Enabled = false
Name = Hearing aid takthrough
Outputs = HearingAid


[Setup]

# hardware setup example for a desktop environment, usually more sophisticated
# settings are outsourced to a VASetup.ini that is included in the 'Files' section

# -- outputs

[Output:MyDesktopHP]
Description = Desktop user with headphones
Devices = MyHP

[Output:MyDesktopLS]
Description = Desktop user with loudspeaker monitors
Devices = MyLSLeft, MyLSRight


# -- inputs

[Input:MyDesktopMike]
Description = Generic desktop microphone, i.e. from WebCam or Line-In
Devices = MyLineIn


# -- hardware devices

[OutputDevice:MyHP]
Type = HP
Description = Headphone hardware device (two-channels)
Channels = 1,2

[OutputDevice:MyLSLeft]
Type = LS
Description = Loudspeaker hardware device
Channels = 1
Position = -0.5, 0.2, -0.4

[OutputDevice:MyLSRight]
Type = LS
Description = Loudspeaker hardware device
Channels = 2
Position = 0.5, 0.2, -0.4

[InputDevice:MyLineIn]
Type = MIC
Description = Table microphone in front of user using line-in
Position = 0, -0.3, 0.2
Channels = 1