alex stone wrote:
Ok, this might be a bit of curly question, and as i
don't know if this
is possible, either valid or not.
The subject is placement, and pertains to orchestral recording. (My own
work composed within the box with linuxsampler, from midi in RG, and
recorded in Ardour.)
I'd like to place my instruments as close as possible to an orchestral
setup, in terms of recorded sound. That is, once i've recorded, i'd like
to use convolution and other tools to 'correctly' place instruments
within the overall soundscape.
example:
With the listener sitting 10 metres back from the stage, and facing the
conductor (central) my 1st violins are on the listener's left. Those
first violins occupy a portion of the overall soundscape from a point
approximately 2 metres to the left of the conductor, to an outside left
position, approximately 10 metres from the conductor, and with 8 desks
(2 players per desk) about 4 metres deep at the section's deepest
point, in the shape of a wedge, more or less. That's the pan width of
the section.
Now as i understand it, a metre represents approximately 3ms, so
calculating the leading edge of the section across the stage as 'zero',
the first violin players the furthest in depth from the front of the
stage, should, in theory, (and i know this is approximate only, as i sat
as a player in orchestras for some years, and understand the instinctive
timing compensation that goes on) play about 12ms later than those at
the front. Using the ears, and experimenting, this actually translates
as about 6ms, before the sound becomes unrealistic, using layered violin
samples, both small section and solo. (highly subjective i know, but i
only have my own experience as a player and composer to fall back on here.)
make sure that you are using different samples for each desk if you use
individual delays, otherwise you will introduce comb filtering artefacts.
but i doubt these delays will have any perceptible benefit.
A violin has it's own unique characteristics in
distribution of sound
emanating from the instrument. The player sits facing the conductor, and
the bulk of the overall sound goes up, at an angle, at more or less
30degrees towards the ceiling to a 'point' equivalent to almost directly
over the listener's right shoulder. Naturally the listener 'hears' the
direct sound most prominently, (both with ears, and the 'visual
perception' he gains from listening with his eyes.) Secondly, the violin
also sounds, to a lesser degree, downwards, and in varying proportions,
in a reasonably 'spherical' sound creation model, with the possible
exception of the sound hitting the player's body, and those in his
immediate vicinity. (and other objects, like stands, sheet music, etc,
all playing a part too.)
I've experimented with this quite a bit, and the best result seems to
come from a somewhat inadequate, but acceptable, computational model
based on using, you guessed it, the orchestral experience ears.
So i take one 'hall' impulse, and apply it to varying degrees, mixed
with as precise a pan model as possible (and i use multiple desks to
layer with,more or less, so there's a reasonably accurate depiction of a
pan placed section, instead of the usual pan sample model of either
shifting the section with a stereo pan, or the inadequate right channel
down, left channel up method.)
phew! ambitious!
to make this more complicated (not by intent, i assure
you), i'm
attempting to add a degree of pseudo mike bleed, from my 1st violins,
into the cellos sitting deeper on the stage, and in reduced amounts to
the violas and second violins sitting on the other side of the digital
stage.
All of this is with the intent of getting as as lifelike a sound as
possible from my digital orchestra.
why simulate mike bleed? i thought you were after creating a "true"
orchestra sound, not one including all unwanted multi-miking
artefacts... i'd rather concentrate on instruments and room.
The questions:
In terms of convolution, , can i 'split' a convolution impulse with some
sort of software device, as to emulate the varying degrees of spherical
sound from instruments as described above?
you could get a b-format response from every place in the orchestra
(with all other musicians sitting there, for damping), and then convolve
it with the violin (which would also have to be shoehorned to b-format,
simulating the desired radiation pattern).
but if you have the room and the orchestra, you might as well let them
play your stuff ;)
So, one impulse (I use Jconv by default, as it does a
great job, far
better than most gui bloated offerings in the commercial world) that can
be, by way of sends, and returns, be 'split' or manipulated not only in
terms of length of impulse, but fed as 'panned' so as to put more
impulse 'up', less impulse 'down' and just a twitch of impulse
'forward'
of the player, with near enough to none on the sound going back into the
player.
i'm not sure i understand 100%, but you might want to look into
ambisonics for that. ardour can do it just fine, all you need to do is
bypass the panners and use fons' AMB plugins instead. as to target
format, you could use UHJ stereo. if you desire 5.1, you might want to
consider working in second order ambisonics.
I've written this rather clumsily, but i hope some
of you experts may
understand what i'm trying to achieve here.
Can the impulse be split down it's middle, separating left from right,
aurally speaking, and if this is possible, can i split the impulse into
'wedges' emulating that sphere i wrote of, more or less?
no, i don't think so. you will need a spatial impulse response. the
simplest way to obtain one is to use a soundfield microphone (or a
tetramic, for that matter).
if there's a way to do this, then i'm all
ears, as my mike bleed
experiments suffer from a 'generic' impulse per section affecting
everything to the same degree, including the instruments bled in. I
should note here, this is not about gain, but a wedge of impulse, cut
out of the overall chunk, that represents a 'window' or pan section of
the whole.
i still don't understand why you're after "mike bleed".
I suppose an analogy for the chunk of impulse idea
would be to stretch a
ribbon across a stage, and cut a metre out of the middle. That metre
would be the bit i'd use, as a portion of the whole, in an aural
soundscape, to manipulate, or place, instruments, to a finer degree, in
the attempt to create a more realistic '3d' effect for the listener.
That metre along with other cut out sections of the impulse soundscape
could help me introduce a more....'human' element to a layered
instrument section.
yeah, well, *if* we had a way of capturing a sound field completely over
such a vast area, we would all be very happy indeed. it can be recreated
using wave field synthesis or very high order ambisonics, but currently
there is no way of capturing it, other than measuring a set of
individual points in that sound field.
hth,
jörn