- 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
cretes WAVs in WAVE-EX format
 |
sample PureVerb™
screen-shots |
 |
sample 5-channel
responses |
|
- 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, such filters
are
available separatelly)
- each
of
the 8 inputs can have a gain to balance the mix
- 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)
- the
output channels are either one mono WAV per channel or
stereo WAVs assigned to successive pairs of outputs
- 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
|
Number of
convolutions
|
Processing
time ; speed
|
mono mono
reverb
|
1
(1x1)
|
1
min ; 8x
|
mono stereo
reverb
|
2
(1x2)
|
2
min ; 4x
|
stereo stereo
reverb/re-mix
|
4
(2x2)
|
4
min ; 2x
|
mono 5-ch
reverb/up-mix
|
5
(1x5)
|
5
min ; 1.6x
|
stereo 5-ch
reverb/up-mix
|
10
(2x5)
|
10
min ; 0.8x
|
|
maximum
8 x 8 matrix
|
64
(8x8)
|
64
min ; 0.125x
|
- 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
 |
sample MultiVolver™
screen-shots |
 |
signal
flow graphs, principles |
 |
Ambisonic
decoder details |
|