Is a virtual room
Finite Impulse Response (FIR) generator designed to
create high quality natural-sounding reverberberation to
be processed with:
- MultiVolver™ (Classic,
VST or WCP)
- Lake
Huron™ or CP4™ (more of a historical interest)
- other convolvers, such as
VST-based, that can read WAV-format FIRs
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Is a fast and general off-line
multi-channel reverb/mix (classic) convolver:
general design
convolves and mixes up to
8 x 8 channels in x out in
any combination
allowing for up to an 8 channel
reverb/mix e.g. to 7.1
Oct 2010 two new
convolvers were added to the 5th edition of The Suite:
- MultiVolver VST™ an 8x8 VST plugin convolver (plus separate 1x1, 1x2 and
2x1 versions)
- MultiVolver WCP™ an NxM offline convolver
Both using a common
simple but flexible text file for the FIR setup (for MATLAB,
WAV or Lake SIM FIRs), an example:
[Setup]
InChannels
= 1
OutChannels
= 2
BlockSize
= 4096
Gain
= 0
[FIRs]
Format
= WAV
01->01
= MVTST_L.WAV
01->02
= MVTST_R.WAV
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MultiVolver™ Classic
MultiVolver WCP™
MultiVolver VST™
(in AudioMulch VST host)
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- creates natural-sounding,
high-density, full frequency range 32-bit FIR reverberation
using virtual microphones:
- mono (omni or cardioid)
- coincident stereo (cardioids)
- AB-stereo (variable spacing,
also "wide AB")
- B-format (WXYZ)
- 5-channel (L, C, R, Ls,
Rs e.g. for natural-sounding stereo to 5-channel upmix,
interactive mic setup)
- the 5-channel implementation
allows for individual microphone selection of:
- location
- aim
- type (cardioid, super-cardioid,
hyper-cardioid, figure-of-8, special figure-of-8+ or
omni-directional)
- setups can be saved and
reused
- uses virtual mono or stereo
sources. By creating several sets of mono or stereo source
responses up to 8 sources can be used together with the MultiVolver™
since it will handle the relative calibration/scaling between
responses automatically even for filters created at different
times
- uses an FIR late reverberation
algorithm similar to the one used in CATT-Acoustic™:
- is rather a virtual room
impulse response creator than just a tool giving reverberation
- is based on physical principles
and creates responses with a theoretically correct, with time
increasing, reflection density avoiding the granularity
in the tail created by many Infinite Impulse Response (IIR)
algorithms
- is based on the experience
from ten years of computer modeling and auralization
of real rooms
- is used by Deutsche Grammophon
together with a Lake Huron™ to enhance too dry recordings
- was used together with a
Lake CP4™ to add reverberation to the
music score at the World Cup in Football (Soccer...) final at
Le Stade de France, 1998 (actually CATT-VRoom, the PureVerb™
predecessor)
- creates reverb for VR
audio applications (together with Lake hardware, PureVerb™
can e.g. create all files required to run Lake AniScape™)
- supports several samplerates:
- 44100 Hz
- 48000 Hz
- 88200 Hz
- 96000 Hz
- 16 000 Hz (for Lake
AniScape™ reverb)
- has a straight-forward
interface with settings (*in octave-bands 125 to 16k Hz) for:
- room size
- virtual source location
- source directivity index*
- virtual microphone location
- reverberation time*
- diffusion*
- absorption distribution
(uniform, audience or detailed*)
- reflection incidence distribution
- reflection density
- optional use of the Image
Source Model for 1st order reflections
- optionally creates responses
with the direct sound excluded and the original signal
can be bypassed when mixing in the reverb
- has a large set of presets
to promote realistic reverb settings and give good starting
points for further fine-tuning:
- user expandable "Generic
Halls" presets where average size and RT values can be
taken from actual well know halls
- suggests reverberation time
based on the volume and type of music (chamber music,
opera, choir etc.) using curves of optimal ratios from literature
(see fig below)
- suggests a volume based
on the reverberation time and type of music (chamber music,
opera, choir etc.) using curves of optimal ratios from literature
(see fig above)
- suggests listener (or virtual
microphone) location based on the reverberation radius
of the selected hall (Close, Medium, Far)
- creates CATT
PLT-files (2D plans, shaded 3D models and RT/Volume graphs)
that can be viewed, printed and exported either via the included
stand-alone CATT PLT-viewer or with
CATT-Acoustic™ (from v7.2, also
the demo version)
- creates a new PureVerb eXtended (PVX) single-file,
multi-input, multi-output, multi-microphone impulse response
format that also stores a PureVerb settings-file.
A PVX-file can be loaded by File|Open for examining the parameters
used when creating the impulse responses, or to re-calculate
the same basic responses after some changes in parameters.
The PVX-format simplifies use of PureVerb responses in
MultiVolver since only one file needs
to selected for a complete set of filters. The PvxViewer shows the
contents of a PVX-file, impulse responses, octave-band filtered
impulse responses and decays.
- optionally directly creates
WAV-file format and includes a file-format conversion
tool that converts Lake SIM-files to 16-, 24-
or 32-bit PCM integer WAVs for use with other convolvers. Optionally creates WAVs
in WAVE-EX format
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sample
PureVerb™ screen-shots |
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sample
5-channel responses |
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- examples of uses (all
of these can be done in one step with auto-scaling):
- mono
mono reverb (1 x 1)
- mono
stereo reverb/up-mix (1 x
2)
- mono
B-format reverb/upmix (1 x 4)
- mono
5-channel reverb/up-mix (1 x 5)
- stereo
stereo reverb/re-mix (2 x 2)
- stereo
B-format reverb/up-mix (2 x 4)
- 8-channel
5-channel reverb/mix (8 x 5)
- cross-talk cancellation
of a binaurally recorded stereo file (2x2, CATT-Acoustic™
>= v7 can create cross-talk cancellation filters)
- 5-channel
binaural down-mix (5x2 or 6x2 for
5.1)*)
- n-channel Ambisonic
binaural down-mix (e.g. 4x2, 6x2
or 8x2)*)
- B-format
Ambisonic decode
for up to 8 loudspeakers
*) CATT-Acoustic™
>= v7 can create binaural room filters for various
listening room configurations
- each of the 8 inputs can
have a gain to balance the mix
- Classic: each of
the 8 inputs can be any of:
- a mono WAV
- the left of a stereo WAV
- the right of a stereo WAV
- stereo WAV (left/right
mix)
- WCP: each of the inputs can
be any of:
- a mono WAV
- a selected channel of a
multi-channel WAV
- Classic: the output
channels are either one mono WAV per channel or stereo
WAVs assigned to successive pairs of outputs. WCP: output channels are either
one mono WAV per channel or multi.channel WAV
- automatically handles
calibration/scaling/overflow:
- separate mono
5-channel filters for each source
location can be created in PureVerb™
and they will be relative calibrated when the processing is made,
no wild trial runs required to check for overflow
- estimates filter gains
and adjusts so that the output WAVs will not overflow. If it still happens
in some odd cases (e.g. if the input WAVs have very low margins), it will
be discovered and a safe second pass will be used
- gives a min "margin" value
on the output WAV-files (similar to that of pro DATs
and similar). If the resulting margin is too high, an automatic
remake can be performed
- automatically handles
delays from sources to microphones (if PureVerb™ or
CATT-Acoustic™ responses are used, for imported
responses initial delays should be included as initial zeros)
- utilizes Lake's
32-bit integer SIM-file format or PureVerb PVX for for the FIR filters:
- allows off-line tests to
be made and subsequent processing using the
same filters in a Lake Huron™ processor
(or the other way around)
- each set of FIR-filters
can have a base-name so that fast and safe filter changes can be
made (requires only one name selection)
- can as well use measured
FIRs via file conversion to SIM-format
- PVX load dialog:
- uses/creates 16-,24- or
32-bit PCM WAVs, optionally in WAVE-EX format.
- uses no temporary hard
disk space at all, can thus convolve any length WAVs (as
long as the resulting WAVs can be fit on the output disk)
- performs floating point
convolution of long filters typically at 8x speed on an
old 500 MHz PIII PC. Sample processing times with 88,000 tap
FIRs (i.e. 2 sec reverb at 44.1 kHz) and 8
minutes of music:
Type of processing
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Number of
convolutions
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Processing
time ; speed
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mono
mono reverb
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1
(1x1)
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1
min ; 8x
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mono
stereo reverb
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2
(1x2)
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2
min ; 4x
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stereo
stereo reverb/re-mix
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4
(2x2)
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4
min ; 2x
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mono
5-ch reverb/up-mix
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5
(1x5)
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5
min ; 1.6x
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stereo
5-ch reverb/up-mix
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10
(2x5)
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10
min ; 0.8x
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maximum
8 x 8 matrix
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64
(8x8)
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64
min ; 0.125x
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- real-time 2x5 zero- or
low-latency processing in hardware at 44.1 or 48 kHz , as outlined
above, requires:
- a Lake Huron™
with a minimum of 3 DSP-boards (12 Motorola
DSPs) giving a latency of 160 ms (2 ms latency requires 5 DSP-boards,
20 DSPs)
- or (for 2x4 only
actually):
- two
Sony DRE-S777: 2x(11+11)=44 Sony
custom DSPs total (assuming the required second
DSP boards also contains 11 DSPs)
- two
Yamaha SREV1: 2x32=64 Yamaha
custom DSPs total
- has an integrated Ambisonic
decode filter creator:
- optional user-supplied
shelf- and inverse speaker filters
- energy and velocity vector
graphs
- save/load rigs or complete
filter setups
- includes a WAV-file player
with single, A/B and play-list options
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sample
MultiVolver VST/WCP™ screen-shots |
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sample
MultiVolver™ classic screen-shots |
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signal flow graphs, principles |
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Ambisonic decoder details |
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