In light of the recent discussion on digital X-overs, I have stumbled across the following device from Texas Instruments. The TAS3001 Digital Audio Processor. This device can be used for eq'ing, phase correction, and digital crossovers. All that would then be needed are the output DACs. I am in the process of desiging an evaluation system using two of these devices and an easy to program micro. Gotta love free samples. I will post my progress and findings.
dac digital crossover
Check out the Accuphase's new product DF-35 which essentially doe what we are all hoping to get as our next holy grail in hi-fi. Their web site has a user manual which gives a full detail. www.accuphase.com (click the what's new option). The reviews in Japanese hi end audio circle can't paint with enough superlatives. It takes no genius to imagine how individual channels of speakers could sound if their cut-offs of unwanted frequencies were done at up to 96db! The Accuphase x-over network controls, in dsp, volume, frequency cut-off and time alignment.
Does the TI chip let you do these? How's your project going? You know Tact Audio www.tactaudio.com has a similar system that is PC controlled.
digital x-over PC
To test two-way digital x-over, I use my PC, soundcard(s) and Labview programming. The Labview program filters any stereo wave file realtime into four channel output (stereo hi and low frequency).
Filtertype, order, x-over frequency, delay and separate volumes are user selectable.
No soldering needed.;)
trouble with soundcard based testing is that because of the noise inside a pc case and the lack of quality sound cards the ability to make useful measurements above 12-15KHz is somewhat limited due to the low level of accuracy @ these frequencies.
where did you get that labview program that you speak of? does it do digital output also (SP/DIF from soundcard?) where can i get it?
Digital X-over made easy
How do you think of using an CPLD or FPGA.
The IIR or FIR algorithms can be implemented in an easy way.
Or probably you can also use the new DSPIC of microchip.
The only disadvantage is that is has only a bit datapath.
has anyone used NeWFIIR or any of it's LINUX based FIR implementation cousins? after reading a lot about it, i am highly interested.
I use Labview (by National Instruments) programming professionally and thus also private.
I think i will make a freeware executable, but not today.
These should also work under Linux as Labview does.
Hmm, While LV is nice, and perhaps
serves as a great repository of
experimental data, I don't have a calibrated
microphone, so I can't be sure that the latest
parameter change is equivalent to the _final
version_ (much as I would like this to be the case).
I think the original poster was on-track with
the TAS3001. This part is available at Digikey
for $3.20, sig. cheaper than a DSP. I guess I
want to go 'blue sky' here and spell out my
The crossover uses two TAS3001 each running
identical coefficients, one for left and one
for right (if I am making a mistake here please let me know). In my design, the left and right channels, as clocked into the chip, would be redefined as 'bass' and 'treble'; the implication being that the chip shall not recieve the sensible SPDIF signal, but rather, a mono version of one channel, thus the requirement of two chips and an additional decoder.
Presentation of the result (L+R) begs a number of taste test questions (I would rather send the bass to my Crown and my teble to my tubes).
To the matter of programming the TAS3001, again I would defer to LabVIEW, and generate an application that somehow communicates a distilled version of a GUI to_the_chip. Of course, once the chip is power_cycled the last known coefficients would be used, i.e. stand alone.
Since I don't have a calibrated mic. I think that this would be the perfect setup to develop a MTM config free standing speaker, coefficients determined by emotional response of the listener, with direct coupling to all speakers, electricity in the room.
I'm new here, and although this thread seems to have gone a bit cold, I'll chime in a bit.
I'm currently running a digital xover on a Linux PC using BruteFir - related to NwFIIR tools that Yoda mentions earlier in the thread. I'm using a Delta 66 soundcard, but will be moving to a Delta 1010 in the forseeable future. IMHO, the PC is by far the best platform for DSP work for the hobbiest, since you get to spend your time doing DSP stuff, not designing circuit boards etc. With a *good* soundcard (Delta 1010, RME Hammerfall etc) the results are bordering on true state-of-the-art. My current work is what I'd call a prototype, and I'm running a 2-way system with 16k tap FIR filters. It takes ~15% of a 1.2GHz Athlon, so there's lots of room to grow. Results on my Carver ribbons are stunning. No real room correction happening yet, but that's next in line (after volume control)
Not having seen the earlier discussions on digital xovers, it's a bit hard to get a read on what people are looking for/thinking. The Accuphase unit for example is interesting, but looks like it only does the xover curves - it does not handle driver eq in any way (presumably so that Accuphase can sell you their mega-$$$$ digital eq). For the price they are likely to be charging, this strikes me as a crippling oversight for anything other than subwoofer xovers. Also, although 'coe' likes the idea of 96dB attenuation, the practical realities of driver integration almost always require you to have much gentler slopes. (Aside - we toured the Dunlavy factory a couple years ago, and I got to talk to John D briefly about his digital speaker prototype. Despite having the flexibility to use any xover slope they wanted, they ended up using ~3rd order 18dB/octave slopes (linear phase), as they sounded the best. )
The TI filter-on-a-chip units look quite interesting, although somewhat limited. The inclusion of parametric eq provides the opportunity for all-in-one xover and driver correction, which is cool (although I'm at a loss to explain why they need '80 mips' for basically 12 biquads). The biggest downsides are a) IIR filters only, no FIR, which means no linear phase xovers b) relatively complex overall system design. The latter would appear to be the biggest problem for the DIY crowd - by the time you get an input receiver, split that out to N/2 channels * Y bands, integrate the microcontroller to handle those chips, route to DACs/transmitters, cobble together some sort of UI and possibly measurement system, this is a non-trivial project. Couple that with the ample evidence that digital circuit design/layout is critical to good DAC/jitter performance, and it's a daunting task. If you have good hardware/microcontroller skills, this might not be an insurmountable problem, though
One option for volume control that should work well in this type of system is the CS3310 chips, or the new Burr-Brown equivalents. They are used in some highly regarded mega-buck gear, and are relatively cheap and easy to work with. Plus, you can run them balanced, which I believe should work well with balanced-output DACs. If you already have a microcontroller, these should be an easy addition, and *should be* better than the digital attenuation on the TI filter chips. The DIP versions of the CS3310's are available in small quantities, for something like $10-$12 each. I forgot the name of the distributer where I got mine, though :-(
|All times are GMT. The time now is 05:14 AM.|
vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2013 DragonByte Technologies Ltd.
Copyright ©1999-2013 diyAudio