Andre Yew wrote:
|| "Rusty Boudreaux" wrote in message
|| ...
||| However, your argument using general processors is not
||| valid. In the realm of DSPs and ASICs this level of processing
||| power is certainly achievable at modest cost.
||
|| I disagree. If we're using brute-force FIR techniques, name me an
|| affordable computation system that can achieve 16 billion 32-bit
|| integer MACs per second. That's for a two-channel, 96 kHz system for
|| 24-bit audio, and assumes 32 bits are enough accumulation precision
|| for such a long filter. The newly announced TI C6412 claims to do 2
|| billion 16-bit MACs per second, so if we generously assume things
|| scale linearly, you'd need at least 16 of them to achieve the
|| required
|| computational throughput. They're about $40 in quantity, so parts
|| cost for the CPU alone is $40*16=$640, which translates into
|| approximately 6*$640=$3840 MSRP difference, not counting support
|| circuitry, the case, power supply, R&D time, etc. to support the
|| extra
|| CPUs. Now if we want to do it for a typical home theatre system with
|| 6 or 8 channels, multiply that by 3 or 4. I don't think that's a
|| modest cost.
||
|| One solution, as Denis points out, is smarter algorithms that require
|| less computation load. Optimized FIRs, or decimation are two ways to
|| getting there. The Lake DSP stuff was what I was thinking about
|| actually for reducing latency.

Andre,
maybe a FIR-approach is not the right way here, at least not a very long
FIR.
Basically, as the correction signal is radiated through the speakers as
well, it will be reflected by the room the same way as the original, and so
will be the signal to correct the reflections of the correction signal and
so on. This seems to be more the work of an IIR-filter, which is
computionally very very cheap.
So the FIR is only needed for a limited period, maybe 100ms or even less
which is a lot easier to compute.
I have been using Angelo Farina's Aurora plugins for CEP(32bit float) to get
the FIR-part and then Matlab to model the IIR-part, but I couldn't get a
satisfactory result. Everything sounded a bit dull and distant, but I might
have made mistakes.

Maybe the whole correction is conceptionally a wrong approach. I would
rather try something else, by positioning additional speakers on the first
reflection points on the wall/ceiling to cancel the first reflections to get
a longer ITD(initial time delay gap).
In the moment I'm doing this and the result is much more promising. I have
also placed active bass "suckers" in the room corners to cancel the standing
waves creation.
Unfortunately I do not have enough hardware to create all the signals needed
in real time. This is also a bit much of a task for a single person.
--
ciao Ban
Bordighera, Italy
electronic hardware designer
http://www.bansuri.my-page.ms/