| harvardian |
Hi,
I am looking into creating an audiophile digital crossover. As co-developer of the APOX volume control, I think there may be merit in a unit designed for best audio performance.
Instead of just allowing Nth order slopes, I am hoping to allow arbitrary mag/phase plots to be downloaded and the appropriate filter coefficients calculated. Of course, standard butterworth and other types would be supported.
I am not sure that a DSP is really needed for the calculations. I would rather use a couple of high speed microcontrollers or even a low end X86 type chip (built in FPU).
Could people list a few high level specs that they would like to be included...
In general, I was anticipating:
SPDIF or I2S inputs
Analog Inputs: 24/96 ADC
six or eight channels 24/96 DAC (3 or 4 way)
Digital output?
Volume control via PGA43XX or PGA23XX in analog domain
Etc...
Dale |
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| JasonL |
Great idea. Got any sujestons and planns..
j' |
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| jwb |
You'll be a lot happier with a SHARC than with some x86 CPU. The performance per unit heat produced will be much much higher, and the pin count will be lower.
I've been playing with digital x-overs in software, and I'd like to see phase, slope, 3dB frequency, and passband attenuation all controlled in software.
I guess it really depends on how far toward the high end you want to go. Each channel could have digital outputs, single-ended and balanced analog outputs, etc. The main concern quickly becomes having enough room for all the connectors :) |
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| pinkmouse |
Great idea Dale
Ok, my suggestions-
Good to have
2 ins, and 3+ sub, or 4 +sub outs.
Option of balanced or unbalanced inputs and outputs
Programable/controlable via Usb and a Unix program, as users of both Macs and Pcs can get Linux/Unix running on their machines.
Adjustable delay for time alignment of drivers.
If possible
Linkwitz transform or other EQ on the sub or bass outputs
Maybe two or three bands of adjustable eq for room correction
Volume controls optional for those that don't need them. |
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| roddyama |
Hi Dale,
Now you’re talkin’! I’ll think about this some more, but here are my first thoughts.
Digital time delay per channel
An extra channel per channel for notch filters, or baffle step comp., or FR tailoring.
Channel by-pass
Level per channel
System configuration memory and/or programmable pre-sets
Rodd Yamashita |
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| slowmotion |
Hi Dale ,
I would be interested too ;)
8 analog outputs would be nice ;)
cheers |
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| pinkmouse |
Rodd
Channel mutes perhaps, channel bypass could lead to blown tweeters:bawling:
I think level per channel might be easier to handle in the analogue domain rather than digital, to save processing power, but hey, what do I know;)
At least two presets is definitely a good idea, then you could have one for music and one for HT for instance.
More general thoughts-
A pink noise generator for system setup would be handy... |
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| roddyama |
| quote: | Originally posted by pinkmouse
Rodd
Channel mutes perhaps, channel bypass could lead to blown tweeters:bawling:
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Hi Al,
Yeah, I guess I could wire straight through channels externally. I thought about that because my Heathkit xover has a by-pass, but it is for the bass range only, so in general, you're probably right.
Rodd Yamashita |
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| JasonL |
Is this going to be balanced or non-balanced.
Will it be adjustable.?
will there be one main in and 4 or so out's..
My idea.
Main in
Sub out
hi out
mid out
low out
All adjustable..
Desplay ? yes no.? |
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| pinkmouse |
| Another quick thought, a fixed high Q high pass filter at say 10Hz, for those of us that still use vinyl would be nice:) |
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| hancock |
I want:
2x 1394 input with HDCP (audio and video)
1x DVI input with HDCP
1x HDMI input
4x 1394 output
1x DVI output
and enough MACs and memory to do 1000 taps on each output channel at 96k and a delay on the video.
I know getting the product approved for HDCP on the 1394 costs $10000. Who knows how much the HDMI certification will cost. I'm afraid the RIAA and MPAA have ruined digital for us DIYers.
The good news is, though, that you should be able to do all this with a Windows Media Player Plugin on Windows Longhorn when it comes out in 2005. If Microsoft makes proper use of hyperthreading it should be close to hard real time too. I'm using Windows now to do my DSP with a program called Sounds Logical and an M-Audio Delta 1010. I've given up on using speciallized DSP chips. It's just so much more convenient to do your DSP on a PC. It gives you a lot more time to play around with different filters, which in my book is what DIY DSP is all about.
If you're living in the here and now and just want i2s or AES/EBU ins and outs and don't like PCs, get yourself one of the multitudes of cheap EVM modules from TI, Motorola, etc. You will never be able to get close to those prices with a DIY product. You could, however, add some value by making a kit to reroute the i2s outputs going to the DACS into AES/EBU outputs and use outboard DACs. You could apply the AES/EBU kits to CD and DVD-Audio players as well--foil those damn RIAA bastards.
Good luck on the project, John |
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| pinkmouse |
New idea;)
Very basic cut down circuit with no external controls and just one analogue/ digital in and 3/4 analogue outputs, programmable by remote that could be built into an active speaker. |
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| tiroth |
Like Pinkmouse, I'd hope the modular route was possible; I'd like to see the basic core be as low cost as possible, especially as I'd probably be bypassing a lot of the frills anyway.
I'd like to see some low-cost core that had digital I/O only. ADC/DACs would certainly add a lot to the cost, and many of us might want to do both on our own.
I did like the idea of L+R sub-out; programmable bass management would be a godsend. |
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| ojg |
The simplest solution, and a good place to start out, might be to use a pair of TI TAS3103 processors since these use the i2c interface and could therefore plug directly into the APOX-bus. However they have only biquads and so will not allow you to do long FIR's.
If this is too little, then a DSP is the only solution. A microcontroller or x86 will be too slow and/or too difficult to interface to i2s signals. You could choose between TI C67xx, SHARC 21161 or Motorola 563xx. None of these are dip-chip though :)
I would leave the ADC, DAC and PGA off the board and let people use their own designs here. Too many different opinions on DAC design to satisfy everyone. SPDIF input would be nice though.
ojg |
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| harvardian |
Hi,
Some great ideas. Thanks!
Right now, I am chossing the DSP based on development costs. The Sharc devices have GNU support.
The Motorola CodeWarrior is now on sale at $499, which is not too bad.
The TI devices seem to have the most expensive support costs.
I was planning on a daughter card arrangement with the analog circuits on it.
Has anyone used some of these tool sets?
The SHARC ADSP-21065L is my first choice, with the Motorola
DSP56367 as second...
Dale |
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| JasonL |
| what is 499 the chip or the whole unit. or a programmer. ? |
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| harvardian |
Hi Jason,
$499 is the cost for the CodeWarrior Compiler/linker. I would also need to buy a development kit (board), so that I can play.
The SHARC ADSP-21065 is about $40.00, but there would be a fair bit of glue chips to support it. If you want SPDIF, I would probably add a CS8515 or CS8516.
Dale |
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| pinkmouse |
Hi Dale
I'm not sure if it's a recommendation or not, but most of the pro audio kit I have looked inside recently seems to use the Sharc chipset;)
Obviously this is a little premature, but another thing comes to mind. Would it be possible, as the board will be using mostly SMT, to get the boards partially prefabbed, as I, and I suspect many others, am a bit wary of that aspect of the DIY. |
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| harvardian |
You are correct. I would not attempt to hand solder a 200 pin PQFP package. We would have the SMT components assembled and tested...
Yes, the SHARC seems to be a good choice, except for the lack of built-in flash memory.
I would probably have a second microprocessor to handle the user interface and external communications.
Would people be opposed to using a PC for configuration. I know many use MACs, but ....
There are too many parameters to do a simple UI. The PC could also do the coefficient generation. That would make the actual DSP code much simpler...
Dale |
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| pinkmouse |
| quote: | Originally posted by harvardian
Y
Would people be opposed to using a PC for configuration. I know many use MACs, but ....
There are too many parameters to do a simple UI. The PC could also do the coefficient generation. That would make the actual DSP code much simpler...
|
Pah, you hate us, you really hate us!:D:D
Seriously, I'm sure we could all cope if we had to, but I still think USB would be a nice touch, and probably faster as well if the setup is going to invove a lot of data shifting. Hmm, but then that probably involves writing drivers and such like...Oh well...;) |
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| ojg |
I have programmed both TI and Motorola DSPs and I have found that the Motorolas are not well suited for C-compilers since they have few general-purpose registers. The Motorolas are easy to program in assembly though once you know how :cool:
The TI is well suited for C-compilers and the CodeComposer that comes with the eval board is only limited to 32kB application size, which is plenty for this. So TIs cost is not very high, actually lower than Motorolas.
I think the gcc port to the SHARC is very much out of date. It might be painful to get it to work, and I don't know if AD's debugger is free. The SHARC is a neat chip though, and should fit well in this application.
I am almost ready to sign up 'cause this will be a great project! If you decide to go with TI or Motorola then I'd be more than willing to help. |
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| tiroth |
| Much much cheaper to implement a serial or parallel interface. |
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| JasonL |
But i dont have parallel on a mac LMAO was there ever a parallel on a mac..
This is getting interesting too.
What is spif for.? |
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| harvardian |
Hi Guys,
I have no trouble with a USB interface. My problem is that I would need to create a program that would work on either type of machine to configure the APOX-DFX.
Perhaps, JAVA would work...
Dale
P.S. I just purchased this board
Here is the eval board that I just purchased.
http://www.analog.com/UploadedFiles...ADSP-21161N.pdf
Has everything that I need to do development:
DSP, SPDIF (CS8414) recvr, 24 bit ADC, 8 channels of DACS, flash, ram, USB interface... |
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| JasonL |
HTTP/1.1 400 Bad Request
you mean that one. REally hehe :cool: |
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| harvardian |
Hi Jason,
Link fixed...
Sorry...
SPDIF is:
S/PDIF
S/PDIF (Sony/Philips Digital Interface) is a standard audio transfer file format. It is usually found on digital audio equipment such as a DAT machine or audio processing device. It allows the transfer of audio from one file to another without the conversion to and from an analog format, which could degrade the signal quality.
The most common connector used with an S/PDIF interface is the RCA connector, the same one used for consumer audio products. An optical connector is also sometimes used.
When you hook a CD player to an external DAC, the format used is SPDIF... |
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| JasonL |
| example on back of dvd players you'll see the one single rca a certian colour some times too right. |
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| harvardian |
| quote: | Originally posted by JasonL
example on back of dvd players you'll see the one single rca a certian colour some times too right. |
I hate to be blunt, but huh?
Yes, the single RCA connection is usually SPDIF even for DVD players. Multi-channel data can also be output in variations of this format. |
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| JasonL |
BLUNT where. LOL.
i was saying a example.
: O ) |
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| JasonL |
| This project is going to be expencive isn't it. ? |
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| harvardian |
Hi Jason,
Craig and I were just discussing the costs. We are hoping to be able to compete with the Behringer unit in terms of costs...
Based on low volume, the price may be higher, but hopefully, the performance will also be higher...
Dale |
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| JasonL |
| Well when you get one done you could start a wikki and people would buy one then... |
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| harvardian |
Hi Guys,
I hate to disappoint, but we will not do any form of multi-channel decoding. There is too much competition and the development licenses are way too expensive…
Dale |
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| Da5id4Vz |
I expected this would be the case but felt compelled to ask.
Is an OEM relationship with an existing licensed manufacture also out of the question? Im not sure, however, that the double D license allows for this much flexibility. |
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| ojg |
The SHARC 21161 is a great chip! I'm glad you went with it instead of the 21065. More horsepower!!!
Good luck! |
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| Petter |
Features:
1. Input sample rate converter. Why not use 192/32 into the chip and 64 bit internal resolution?
2. Non-resampled input as well
3. High quality clock with PLL to enable various output frequencies to clock external devices (reads sources to more or less ELIMINATE jitter).
4. 64 bit FP DSP with ample processing power!!! Don't skimp here!
5. Frequency domain filtering
6. Phase domain filtering
7. Time delays
8. Don't need SPDIF out bckl, l/rclk and mclk should be enough. S/PDIF is a troublesome format which increases parts count + adds jitter by default since it uses an embedded clock (unless you are willing to transmit MCLK next to it as well).
PC based units will be more convenient for me. Be warned of costs involved with manufacturing. I have a 2 part Audio Amateur article about a guy who did something similar (yet different :)) who had total cost runaway based on in part mask manufacturing for solder application with complex chips. I can probably get you this to review if you are interested.
I believe Crystal offer multi-channel decoding hard-coded into some of their chips. I am not sure how that works in terms of licensing.
The easiest and cheapes is the integrated TI DAC chips with biquads proposef before me. They would meld nicely into the current Apox line and enable saled of more Apox and digital crossover at low cost. I have considered them, but the silly programming (which you guys are good at) through SPI or whatever killed it for me.
I seriosly doubdt that you will be able to compete pricewise with Behringer et al., but hey! Why not!!!
Petter |
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| dwk123 |
I've been interested in DSP xovers for quite a while - 8-10 years or so since first hearing about the Meridian speakers. I've seen a couple dozen discussions like this one, about how to come up with the ultimate audio dsp box etc. None of them as far as I know ever amounted to anything, for a variety of reasons.
The biggest problem IMHO is that they start out looking at the hardware first, and this isn't where the interesting problem is. The challenges are really in the software, both the choice of algorithms/filters used as well as the 'user interface' into the unit - how do you upload/download configurtions? measure/massage/generate FIR filters? Channel mixing/routing?. As we can see already in this thread, getting anyone to even agree on what the feature set should be is hard enough, forget the problems ultimately defining it down to an implementation level and implementing it.
Furthermore, the hardware engineering is problematic in many ways - usually more expensive than originally expected, longer design cycle than expected which means that new chips are available before the board goes 'production', rather difficult to build decent expansion capabilities in to appeal to a broad enough audience for critical mass (ie 1 dsp and 4 channels for 'cheap' entry level, 2/3/4 dsp's and 8/10/12 channels for high-end theater application), funding the prototype cycles and initial production runs etc.
I went through a couple motorola eval boards (56007 and 56362) without ever really getting a system that worked, before bailing out on the whole custom hardware idea.
IMHO the only sane way to approach this problem as a part-timer/hobbiest is to implement the design on a PC with *good* multi-channel soundcard first - get the filter topologies pinned down, identify the control parameters/mechanisms, determine the actual horsepower/memory needed etc. Modern PC's have enough floating-pt capability for virtually anything you can throw at them, and the probramming environment is far far more productive for prototyping/investagative work (unless you're already a DSP expert). Furthermore, a simple linux-based pc w/ say an M-Audio Revo soundcard can be thrown together for maybe $350 or so, and is more than enough for a prototyping platform. (RME or higher-end M-Audio cards are also supported under linux if you want something better)
Once you have the system working on the PC, THEN you start looking at the custom hardware since now you have a firm functional basis on which to design. It may add a few months to the overall design cycle, but the payoff is that you *know* the system will meet the design goals before you start.
For me, I now use Jack under Linux, and really it's incredibly easy to do this - BruteFIR has Jack support for long FIR filters, there are a bunch of LADSPA hosts for simple biquad implementations, you get signal routing and switching for free etc. Input-to-output latency can be as low as a 3-5 ms for biquad based setups - long FIR's like bigger buffers for better efficiency, but I still run 32k tap FIR's in a 3-way xover w/ 16ms latency on a 1.6GHz P4, with plenty of cpu left over. |
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| tbla |
| regarding the biggest newest top of the line meridians.....a friend of mine went for a "listening session" he drove for 3 hours and listened for less than 10 minuts - extremely disappointing.......i still believe that very few people are able to design a good loudspeaker....!!! :( |
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| Petter |
Also, it is possible to run National Instruments Labview directly on FPGA's -- which might prove to be the optimally performant system based on a simple GUI development method. I will look into this tomorrow.
Still, even remembering the long discussion about PC versus DSP about a year ago (which turned quite ugly) that the PC based method is probably the simplest. However, this means one effectively enters as a software vendor rather than a hardware vendor.
Petter |
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| harvardian |
Hi Petter,
Great suggestion. We did not know these exist. From 5-16 biquad filter stages.
1) Are the ones with built-in ADC/DAC good enough?
2) Should we use the all digital ones and use better ADC/DACS?
Dale |
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| scraggles |
| quote: | Hi Guys,
I hate to disappoint, but we will not do any form of multi-channel decoding. There is too much competition and the development licenses are way too expensive…
Dale |
Out of interest, what are you considering as competition for multi-channel decoding? Is there a DIY solution for this, whether in the form of a board that can be integrated or a total kit?
I'm contemplating building both a DAC and a Preamp, but see it being obsolete in no time. I could build multi-channel units, but how do I get the (DVD) signal into multi-channel? |
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| harvardian |
Hi Scraggles,
I guess that I was referring to all of the inexpensive HT preamps/receivers on the market. The licenses are geared towards shipping lots of units. Dolby Labs wants a bundle just to play.
There are no DIY solutions, and I fear that this is beyond most people, including me!
Dale |
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| dipchip |
What do people think about a system like this?
First there is a small PCB board: Call it BOARD A
BOARD A features:
(SPDIF and/or analog input)-> (I2S)->(differential output I2S (For long cable lengths) onto ethernet type cable)
a microcontroller to act as CAN Master node and have a USB PC interface.
The CAN bus is also placed on the ethernet cable.
There would probably be three RJ-45 jacks on this board. (one for each
speaker)
Then embedded inside each one of your speakers (maybe?) is BOARD B.
BOARD B: features
(differential input I2S )->(I2S)->(DSP digital filters with three
outputs)->(3 x 24 bit audio DAC outputs)->(volume controller)->(your power amp here)
There would be some sort of switch on the board where you would select L, R,or L+R (for example, even though the digital input would contain both L and R data.
The board could ignore the data it didn't need it for that particular speaker. or could mix L&R for bass managment etc..)
a microcontroller would act as a CAN Slave node and would send all received commands to the DSP.
BOARD A: could then receive all of the filter parameters via the USB bus.
BOARD A would then pass all of this filter parameters over the CAN bus
to the microcontroller on BOARD B, where they would be passed to the DSP
and to volume controller. That way you have digital going all the way to
your speaker.
The microcontroller would then store these parameters into EEPROM for later
use.
Does this seem like a good idea to anyone, or am I nuts! :yes:
Thanks,
Craig Beiferman |
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| pinkmouse |
Sounds good Craig, but do we really need another volume stage? It would only really be of any use if you just had one source, such as a cd player, and used the crossover as a pre as well. And you will also lose sales of the Apox kits!
Maybe you could also make the B board a basic 2 channel system, with extra add on boards each with another two channels worth of processing/DAC, for those with greater crossover needs. This might make it cheaper for those with lesser requirements, or the financially challenged;) |
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| harvardian |
The crossover is really geared towards DIY'ers that want to build "active" loudspeakers. For them, there would be a fully digital link from the source (CD) to the speaker. There will also be a ADC for those with analog inputs.
For this application, the volume control should be after the crossover in the analog domain.
We envision this system as just another choice in the APOX system. You could, of course, bypass the volume control on the crossover board and use one of the other APOX boards.
Of course, one could put all boards in a chassis and run amp outputs to the speakers. The break up of the boards lets the user choose either method.
Dale |
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| Peter Daniel |
| You definitely need volume control in crossover modules to adjust volume for individual drivers. But the question remains if it should be as elaborate as APOX, or a simple trimpot which later can be replaced with fixed resistor would be enough? As you gear it towards DIY'ers, the latter could be cheaper and probably better sounding solution;) |
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| harvardian |
Hi Peter,
Here is our thought on the subject.
If you are actually doing an active speaker and would like the crossover/amps at the speaker, you would need to control the volume at the speaker. The crossover is acting like a preamp in this case.
Dale |
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| Peter Daniel |
| This makes sense, but I guess you need a source switching unit in case you want more than one source. |
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| harvardian |
The APOX-IS1 should work ;)
We may have a digital switching as well |
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| pinkmouse |
| quote: | Originally posted by harvardian
The crossover is really geared towards DIY'ers that want to build "active" loudspeakers. For them, there would be a fully digital link from the source (CD) to the speaker. There will also be a ADC for those with analog inputs.
For this application, the volume control should be after the crossover in the analog domain.
We envision this system as just another choice in the APOX system. You could, of course, bypass the volume control on the crossover board and use one of the other APOX boards.
|
I understand completely that you want to make a killer system, and I see how the on board volume controls fit in with that idea. However, I think you are limiting your potential market by having this as a built in expense that I suspect most users will not use, as they already have pre amps and multi channel decoder systems.
How about just having a plug in option that will just transfer data to one of the standard Apox system boards if people want this facility, and can be bypassed by those that have existing system components to integrate.
BTW, I can't wait for the IS1, as I am in need of a simple way to remote my GCs for my bedroom system;) |
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| harvardian |
I guess that I am either very confused or not explaining myself correctly.
Here is one use of the system:
1) User has CD/DVD player with SPDIF (digital) output. User connects to our board A (input/driver board). This is a raw digital signal. How does one control volume? Some CD's can control in the digital domain, but this may reduce resolution.
For the other analog inputs, you could feed the raw analog or a pre-amped signal.
Of course, one could just not install the volume chips and bypass the signals. They are expensive!
Dale |
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| pinkmouse |
| quote: | Originally posted by harvardian
IOf course, one could just not install the volume chips and bypass the signals. They are expensive! |
That was my point! I think we were going round in circles:D |
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| relder |
Just FYI, have you all seen the SHARC XO project in AudioExpress May-July 2001? (June has the meat of the design) Uses 2 ADSP21061 DSPs.
Looks like the author's web site no longer functions so code might be difficult to obtain. |
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| Peter Daniel |
| So the volume in crossovers is adjusted in digital or analog domain? |
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| Petter |
Surround decoding is EASY on the PC, because there are several vendors who offer such functionality at a reasonable cost.
I would personally like to see a PC based solution, perhaps with some external stuff, ideally all controlled from the PC (or an Apox).
I have played with the Behringer GUI some, and it is probably very labour intensive to create such a GUI, particularly the GUI on the device itself. Download the PC software and judge for yourselves.
Petter |
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| tcpip |
| quote: | Originally posted by Petter
Surround decoding is EASY on the PC, because there are several vendors who offer such functionality at a reasonable cost.
I would personally like to see a PC based solution, perhaps with some external stuff, ideally all controlled from the PC (or an Apox).
|
I agree. I too believe a PC-based solution makes more sense. In this context, dwk123's post about first building a prototype on a PC makes total sense. And I feel that if the aim is just to build an xo, BruteFIR and Friends seem to prove that one modern PC can handle a 4-way stereo xo + eq quite well.
In addition, I want to suggest that the software interface should interact with this box using an "open" standard. I feel the "right" way to visualise the digital xo box is as a server, and the control program on the PC/Mac/whatever is a client which connects to the server, reconfigures it, sends parameters, gets status, etc. That way, the protocol between server and client will be a clean dividing point, allowing independent DIY paths of development for the two. If you build the digital xo box on a PC, you can actually embed a small HTTP server on it, and control the box over an Ethernet cable from anywhere.
I feel that if HTTP is not an option, then USB could be used. USB devices come in various classes, and the simplest class is "memory" or "storage" devices. Such a device just appears as an array of bytes to the computer. If your digital xo box can pretend to be a "storage USB device", then people can write programs on any OS, on any type of hardware, to control it. All commands from the control program to the xo box can be transmitted by writing specific messages to specific locations in this "storage" area, and return values can be communicated from the box to the computer by messages in other parts of the "storage" area. In essence, a chunk of memory on the xo box becomes a message passing area. If the designers adopt this design and publish the message protocols first, then software design and hardware design can proceed in parallel. And then, we don't have to worry about USB support on this or that OS. It will always be possible to "mount" the USB device as a "storage" device in any modern OS, and then any program can be written in any programming language to communicate by reading and writing messages in that storage area. Issues of shared data structure locking will not arise, because all communication will be initiated by the "client" (i.e. the computer) and the "server" (the xo box) will only respond, using a kind of remote procedure call mechanism.
Hope this makes sense. I've seen a disproportionate focus on hardware in other threads sometimes, leading to ignoring some critical software design issues. I understand software better than hardware, so this hurts. :D Let's hope this one will be different.
I don't care if the designers use proprietary hardware or build on a PC. I know that if I had to build it, I'd start with a PC, in order to "stand on the shoulders of" that hardware. However, all of us stand to lose if ease of use, generality or functionality is compromised because "it would be too much work to provide it on our proprietary hardware design" but could have been provided trivially using off-the-shelf hardware on a PC (e.g. USB interface, Ethernet interface, etc.) Today, even Ethernet-ready laser printers have HTTP servers embedded; I'm sure it's easier writing a printer's firmware than the xo box.
Personally, I think it will be a big tragedy if such a lovely xo solution is built, but is built in such a way that only some proprietary software programs on some specific OSs can control it. It'll kill the growth of the use of the box, and I will almost certainly never use it.
Tarun |
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| sfdoddsy |
By the same token I would not be interested in a PC based system. I would be very interested in a simple digital in digital out with 3-4 bands of EQ on the outputs, but it really needs to be plug and play, set and forget, etc etc.
Sure use a PC to configure, as TACT does, but I like my little reliable quiet moveable boxes.
BTW, what timeframe are we talking about?
Steve |
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| Petter |
I just downloaded media player: Foobar - it is quite possible this machine has the required functionality, but since i have moved to other methods (Behringer) have less time pressure to figure this out :)
Petter |
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| tcpip |
| quote: | Originally posted by sfdoddsy
By the same token I would not be interested in a PC based system. I would be very interested in a simple digital in digital out with 3-4 bands of EQ on the outputs, but it really needs to be plug and play, set and forget, etc etc.
Sure use a PC to configure, as TACT does, but I like my little reliable quiet moveable boxes.
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The two approaches don't need to be mutually exclusive. The hardware designer can build using a PC motherboard, and this motherboard can go into a box which looks like a VHS videocassette player. Such cabinets exist. Of course, serious DIYers will probably build their own cabinet. In either case, it'll look just like a set and forget box for the user. Only the hardware designer will know it's a PC inside. No keyboard, mouse, etc needs to remain connected. You'll plug in the USB or Ethernet port whenever you need to configure/monitor it, that's all.
In fact, I too would be totally disinclined to use something which looks and sounds like a PC (noisy fans, keyboard, monitor, etc) next to my audio system. I want a box, 17" wide, a foot deep, and black in colour. :D
A PC is no longer a PC, check the Mini-ITX Website
| quote: | BTW, what timeframe are we talking about?
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You lost me here. Timeframe for?
Tarun |
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| tcpip |
As an example of the kind of hardware and internals that are also called PCs these days, check out the Hoojum Cubit P4. It even has the hunk of thick aluminium to block all them Evil Microphonics which all Real Audiophiles are so concerned about. Of course, for any digital xo work, you'll probably want to add your own sound card hardware... but it's a great starting point.
In fact, if the digital filtering is done in the user state outside the kernel, on Linux, like with BruteFIR, then you can even give a menu of sound cards for the customer to choose from.
Another product range to drool over is Hush Technologies' boxes. More than enough compute power to do a lot of digital filtering, and totally fanless and silent. And looks great on the audio rack too. One of the models, fully equipped with CPU, RAM, HDD, and CD-ROM drive, is available for EUR 666.00. And their boxes are also available in black, and you can show off all them heatsink fins and hint that you're using "Class A digital filtering." :D
Tarun |
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| Da5id4Vz |
In the Mini ITX world the Shuttle SS-51 has been getting very affordable. Outpost (Frie's Electronics), MEI Microcenter and Tiger Direct all carry it. Its worth checking out on the Mini-ITX Website. It uses fan less heat pipe technology to remove heat from the CPU.
Nice little aluminum box to.
-Dave |
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| dc |
I've been looking into a similar project for a few months now. Wanting a Windows-based solution and not knowing how to program, I found a graphical programming language called Max/MSP, which I believe could be used to create a multi-channel convolution engine (to implement DRC, or Digital Room Correction, a freeware program available on freshmeat and discussed on the htpc section of www.avsforum.com and also at the hifi_dsp Yahoo! group (btw, there are people at both of these groups who are interested in a project similar to the one being discussed in this thread). Max/MSP would also allow for the implementation of user-defined high-order, linear phase xo and time delay per driver. I have no idea how much computing power this would take, though.
tcpip, you've written that the hush technologies box (a Via 1 ghz, if memory serves) is more than enough power.... More than enough power for what? I'm looking for enough power to process up to 16 paths (6 channels total - 2x4, 1x3 and 1x2) of high-order, linear phase filters.
Also, several people have mentioned BruteFIR and Linux. I'm a fan of both, but, there are some limitations. For instance, there are no Linux drivers for my Lynx Two-B soundcard. Also, for those who would use the PC as a front-end (music jukebox, DVD player, etc.) Windows programs and surround decoding algorithms are years ahead of what's available on Linux. |
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| MWP |
| quote: | | Windows programs and surround decoding algorithms are years ahead of what's available on Linux. |
Years... i wouldnt be too sure of that...
But we shouldnt be looking at taking the HTPC path here...
This project should be confined to a nice active xover, room eq... thats it.
I say forget ADC/DACs as as said before, youll never please poeple this way... everyone has thier opinions on what DAC sounds better, etc.
So, stick with:
SPDIF In -> EQ -> XOvers -> 6/8 SPDIF Out.
This will keep things nice and simple, and also help keep the cost down for us poor guys ;) |
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| dc |
| I guess you can look at my statement two ways... I didn't necessarily mean that it will take years for Linux to catch up to the current state of Windows, but, I did mean that Linux is now where Windows was years ago. More development is being aimed at Windows. If what you're looking for is simple a Linux box with spdif in and out, I think the solution already exists - BruteFIR on a Hush box with the appropriate soundcard (maybe RME makes something? There are Linux drivers for most, if not all, of their cards). Creating a Windows based solution would be much more of a challenge, but would allow use of the Lynx Two-B and AES3 ins and outs (Lynx makes a 16-channel AES3 card - the Lynx AES16). Or, use the DACs on the Lynx, which have repeatedly been positively compared to the best standalone DACs available. |
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| tcpip |
| quote: | Originally posted by dc
tcpip, you've written that the hush technologies box (a Via 1 ghz, if memory serves) is more than enough power.... More than enough power for what? |
I asked questions about computer power in a thread I started and the answers surprised me. I found those responses quite eye-opening.
| quote: | | I'm a fan of both, but, there are some limitations. For instance, there are no Linux drivers for my Lynx Two-B soundcard. |
Personally, I feel that we may not suffer in spite of lack of support for some of the high-end cards for this application. Already, others seem to feel that we need only digital in/out, in which case probably lower-end cards can do. Or else go with some of the supported high-end cards, either supported by ALSA or OSS. Anything special I'm missing about the Lynx Two?
| quote: | | Also, for those who would use the PC as a front-end (music jukebox, DVD player, etc.) Windows programs and surround decoding algorithms are years ahead of what's available on Linux. |
DVD playback yes, because of the conflict of open source and CSS. But for just audio playback, I don't see any constraints at all. And for this box, we are not talking any front-end. This is supposed to be a "standard black box in the audio rack", which is programmed over USB (or Ethernet?) from another PC. That PC can run any OS it wants. So for this box, what's the constraint?
| quote: | Originally posted by MWP
But we shouldnt be looking at taking the HTPC path here...
This project should be confined to a nice active xover, room eq... thats it.
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For whatever it's worth, I agree. :)
Tarun |
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| dmfraser |
I have used this device to implement a 2 and 3 way digital crossover and had great results. TI's software is a little clumsy to use but the device has preformed as well as the A/D and D/A converters and analog ground scheme allow it to. Just make sure to run it at 96 KHz.
The advantage is that TI's software supplied will allow you to implement all the functions, with no programming even though it is clumsy and there is a bit of a learning curve.
Also, their software, on the CD I got with the evan board does not support burning EAPROMS. However, I nagged them and they sent me a build that does support EAROMS to run the thing is master mode.
If anyone writes any additional software for operating this DSP, contact me at dmf@renkus-heinz.com. We might be willing to buy it. |
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| netgeek |
I've been playing around with a "paper design" using the TAS3103 for a short while and it appears to be perfect for my application (a 3-way powered speaker with provision for room EQ and other features). I don't have an eval board (yet) but I'm wondering just how dismal the software/development support might be for this device (?)...
Compared to a pcb stuffed full of op-amps and many, many configuration options (e.g. phase alignment, notch filters, etc.) this device might make the ideal "black box". It would take care of the immediate xover, slope, phase, and EQ-fixits required for a given enclosure/driver design and (even better) would allow for later room-EQ and other tweaks in real installations. And all this without the need for a multitude of high-tolerance matched components made of "unobtainium" and the endless fiddling that real world "tuning" involves.
The real challenge, if it's to be useful in the DIY world, would be in creating a decent software package (and GUI) which would allow the user to easily take advantage of all the device has to offer and to then download and test the results. Not easy! But worth it, if it can be done.
Count me in if anyone wants to pursue a widget based on this approach - otherwise I'm off to bash on the thing on my own ('cause I think the potential is too great to ignore)....:spin:
Regards,
Bill |
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| WTS |
| Hi; Great Idea. Although I'm not a programer, I'd love to have something like this in my system. Analog based xovers just doesn't cut it, personally, I think any opamps in the signal path doesn't cut it. I was on Ti's site yesterday and I noticed this device and I was thinking the same thing, will it work. I'm not sure, but does it have what it takes to be in the audiophile class though. |
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| Fred Dieckmann |
"Analog based xovers just doesn't cut it, personally, I think any opamps in the signal path doesn't cut it."
"SPDIF In -> EQ -> XOvers -> 6/8 SPDIF Out.
This will keep things nice and simple, and also help keep the cost down for us poor guys."
You don't have to use op amps. A really good sounding SPDIF interface is much more difficult to design than an analog stage. A simple Spice model is much easier to write than software. You can model the driver rolloff and crossover slopes pretty easily as a starting point for crossover adjustment. There are also programs for hobbiest to model speaker response curves and passive networks to get the desired crossover slopes. Copying the transfer fuction with an active nework after that is not as difficult as you would think.
Which do you think has the easier learning curve and is the path of least resistance? For a decent design the question of which will sound better is a given...... analog, even with some of the better sounding op amps if designing discrete transistor circuits is not your thing. |
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| netgeek |
I'm actually a big fan of op-amps and planned to use lots of them in an active crossover...:) The devices are so good these days that I'm sure I personally can't hear anything they might contribute - but others may disagree. However, I'm thinking that something like the TAS3103 might be worth trying as an alternate (because A/D and D/As are also pretty good these days ;) ).
If you take a look at the excellent work done by Linkwitz (specifically his active design for the Orion, et al) he has obviously done an amazing amount of testing, tweeking and optimising. He's also explained it all very well and if you take it a section at a time it's perfectly logical and clear.
With a device like the TAS3103 you can implement the same functional blocks (ala Linkwitz) with the added advantage that you can very quickly make changes (e.g. throw in a notch filter at the last minute if needed) based on your test results. What's more, you can also add delay, amplitude compensation curves, etc., etc. - all without touching a soldering iron. Instead, you merely update the parameter table to be loaded into the TAS3103 to reflect your changes and you're done (at least until you change your mind again :rolleyes: )...
So, the question for me at least is what it sounds like and whether or not it's worth the trouble. I'm thinking it may well be worth it although Dan has pointed out above that the tools are fairly "clumsy". The prospect of a single A/D --> TAS3103 --> 3 D/A solution (all fairly cheap!) looks attractive if it plays out as well in reality as it does on paper.
Just my 0.02..........................
Bill
"So, is the universe really analog - or digital-with-infinite-resolution ?" :scratch2: |
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| dmfraser |
Our listening tests does put this device in the audiophile class. The total THD+N, input to output on the DSP I made with the TAS3101 is in the area of 0.012%. Running at 96KHz and 24 bits. Used good AKM A/D and D/A converters with 110db+ dynamic range. For opamps on the output filters, I used OP275s. The input and output on my DSP is fully balanced as it is for pro use.
If anyone is trying this device and wants some help, using my experience, write me at dmfraser@sbcglobal.net.
If you write a better GUI for it, contact me too. |
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| WTS |
I personally don't like opamps, from what I've heard of them in the past. Granted, I have not heard any of the lastest super opamps. I just think the less active circuitry in the path the better. I once built a 3 way active xover(i don't recall off hand what I was using for opamps). I thought it sounded great until I decided to try your basic 6db xovers at the front end of each amp. A world of difference, it just opened up. So what was at fault, the design as a whole, the opamps used, I don't know. But I know of other people who went the same route and they won't go back. But thats just my 2 cents worth.
Now as far as doing it in the digital domain, I'm sure that comes with its own pros and cons. I think I'd rather do it the digital domain. My system now is strictly digital right up to the pwramps(biamp) so it would be kinda hard to implement an active analog based xover.
dmfraser,I'm glad to hear that the TAS3101 rates right up there. There seems to be alot of talk about behringer units, do you think this one could surpase it or equal it. |
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| dmfraser |
| Behringer stuff is decent but not super. They make compromises to get the price so low. More bar band equipment than real [ro. |
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| niconoise |
| has anyone looked at the ad1954? |
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| netgeek |
The Analog Devices widgets look very interesting but according to their data sheet "max 48 kHz sample rate".........:(
Also, they have alot of extraneous features and options which seem geared more towards car audio or lower end 2.1 applications. Still, they're on the right track...sort of......:)
Regards,
Bill |
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| dmfraser |
I found that with a 48KHz sample rate the THD+N is about 3 times higher. On the same test unit I get 0.048% THD+N at 48KHz and 0.016% at 96 KHz.
This translates into a somewhat grittier sound at 48KHz though the extra THD does make to sound seem a little brighter. Adding some dither helps with the distortion at a cost of more noise. I have to design for 100db of more S/N so I have to do 96 KHz. However, for consumer use, where the sound source is 16 bit consumer grade CDs sampling at 44.1 KHz, playing through low powered (under 200W) low efficiency speakers (<95 db 1W/1M) then 48KHz is likely OK. Its just not audiophile.
I'm using, from AKM, the AKM5385A A/D and the AKM4395 D/A. I also use the Cirrus CS4362 in some products. All these are balanced input/output devices as I cannot get over 90db S/N or dynamic range out of single ended devices when I have 1000W Class-D power amplifiers as part of the system.
By going fully balanced and running at 96 KHz, I can get up to 107db dynamic range from the system.
Dan |
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| netgeek |
Well, since it's been a month now since the last post - we must all have had a chance to think about it, right?..........:) :)
I'd still be interested in the TAS3103 or AD1953/4 approach but can't bring myself to hand over $700 or so for an eval board just to play with either device. The alternative would be to glean as much as possible from the published app notes and then lash together a preliminary design to hack at but, unfortunately, the quantity of material available (e.g. not even a published reference design ??!) seems pretty sparse.
Dan, if you have an eval board - did they provide much in the way of support materials with it, or do you pretty much have to wing it from scratch?
Bill |
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| dmfraser |
TI included a GUI with the TAS3004 and TAS3103 eval boards I bought. They also included a piece of software called the Automatic Loudspeaker Equalizer where you import MLSSA data and it will automatically come up with a set of EQs for you to make your speaker match the curve you give it. They are asking $500.00 for that by itself though I don't know if anyone will pay that.
As well, when I found a feature on the GUI was not operative, I complained to TI and they made the feature operative and sent me a new version of the software. However, it is likely what they are shipping to others is sill unable to write to an EAROM for storing setups.
The DSP eval board for the TAS3103 was fixed at a 48 KHz sample rate and I had to change the crystal to make it run at 96K. As well, the A/D and D/A on the eval boards were only consumer grade and I had to use the I2C inputs and outputs on the EVM to go through decent converters.
However, I was able to get a functioning DSP built without having to write any code for it. And I got over 100db S/N out of it.
I just got my prototype PCBs today for a 1 In/3 or 4 out DSP using the TAS3103 and I hope this will work as good as the 1 in/2 out did for me.
Dan |
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| netgeek |
It sounds like the support stuff TI provides is at least somewhat useful but I suspect that if I try the "lone hacker" approach (i.e. without eval board, etc.) I probably won't get too far. And without a "real" (that is - "commercial") application and an equivalent "real" customer I can't realistically expect much help from TI...:) Can't fault them for that really - they need to filter out the "tirekickers" (like ME :) ) from live customers after all. Unfortunately, since I don't have my old day-job to use as a cover story..:rolleyes: .. I can't just get in touch with TI's product manager and ask for any favors.
Anybody want to rent out an eval board?............:whazzat: |
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| netgeek |
P.S.... for Dan:
They didn't just happen to throw in a schematic for the eval board by any chance (free of NDAs, etc.)????????
Bill |
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| dmfraser |
Just received a DEQ2496. I needed something to give me AES/EBU signals from an analog source. This thing blew me away. It measures great (I have 3 Audio Precisions in my lab) and sounded great. It has every feature I could think of in an EQ including Real Time Analyzer, feedback killer, room equalizer, etc. Even delay lines that can be compensated for different air temperatures.
It uses two SHARC processors and has AKM A/D and D/A converters for the main signals. In fact the converters are amongst the best available and better than used on consumer equipment. It runs at 96KHz on the input and can take SPDIF, AES/EBU or analog input and output all the same. Optical in and out too. A great piece of gear for $299.00.
The display is great too and there are piles of non-volatile presets. The limiters/expanders tested out great too. If you need an equalizer that does damn near everything, I recommend this unit highly.
I imagine the DCX2496 three way crossover with 3 analog inputs and 6 analog outputs to be just as good. I'd recommend you look up the specs on it before going off to build something. I really doubt you will do a lot better until you get to several times the price.
Dan |
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| CeramicMan |
| quote: | Originally posted by dmfraser
I found that with a 48KHz sample rate the THD+N is about 3 times higher. On the same test unit I get 0.048% THD+N at 48KHz and 0.016% at 96 KHz.
This translates into a somewhat grittier sound at 48KHz though the extra THD does make to sound seem a little brighter. Adding some dither helps with the distortion at a cost of more noise.... |
I doubt the lower quality sound would be due to THD, but instead due to aliasing. For this reason analogue low-pass filters are still required above 20kHz even with 192kHz DACs let alone 48kHz ones. I for one would not buy a DSP XO solution if it did not have built in DACs. For one thing, I'd have to buy another product just to make the first product work :smash: , and any possible benefits would probably be outweighed by problems introduced by having the DACs a large physical distance from the DSP.
I would try to make the product affordable, otherwise you lose most of your potential market and start fighting for attention in the "audiophile" market. Hint: avoid the audiophile market at all costs: some of those people can hear differences where there are none and vice versa; and most of your budget will go down the drain on marketing and selling the product on appearance.
On my wish-list would be a DSP solution for 2 channels in, 6 or maybe even 7 channels out. The box would have some sort of minimal UI so a PC is not required, however it would also have connectivity to a PC for setting up initially, major adjustments and reprogramming....
I think there's a limit to how much the actual crossover slopes can benefit sound quality, and "linear-phase" crossovers are practically just a red herring - a good algorithm should handle that anyway. I would be much more interested in being able to calibrate my speakers with microphones, specifically to eliminate ringing effects in the time domain.
CM |
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| netgeek |
It seems as though this thread (as with many others) is going in multiple directions. Nothing wrong with that - it promotes alot of good discussion and comments that probably benefit many folks - even if their needs/wishes are for drastically differing applications (but at least somewhat related). I guess we need to be more specific up-front as to what we're after. So here's my attempt:
I'm not looking to replace nicely designed, rack-mounted parametric EQs, graphic equalisers, or fancy digital crossovers. It seems that there are some great products out there right now (and as Dan points out - for $299 you just are NOT going to beat that with a DIY).
What I AM looking to do is to replace a passive crossover with a digital version that is low-cost, repeatable, doesn't require high-tolerance passive components, can be tweaked ad-naseum (without a soldering iron), and provides an alternative to very high component-count active crossover/equalizer solutions. Sort of a cheap "black-box" intended to be used in active speakers. Ideally, it's a 1-in (probably balanced) 3-out (single-ended since it will directly feed individual amps) widget that can be embedded in a speaker enclosure along with associated amps, PSU, etc.
Cheap is good for this! For example, if it were cheap enough, someone could construct a "Super-Dayton BR1" if they wanted to (okay, so this is an extreme example :) ) which would allow them to tweak the response curves, etc. quite a bit and apply the outputs directly to (perhaps) some inexpensive "gain-clone" type amps to run the drivers. Expand this idea to include a sub and/or 3-way design with better drivers, more or less power, provision for adding in room-EQ, delay, etc., etc. - well I'm sure you get the idea. In other words, I'm not out to replace Behringers - I'm out to replace passive crossovers with something much more flexible that can be measured, modeled, locked-in and then stuffed into an enclosure. Seems like the TI or Analog Devices widgets might be just the thing. The problem, of course, would be to create a PACKAGE that an end-user could easily configure, tweak and test without having to become an expert on the use of the device at a low level (i.e. the equivalent of Assembly-Language) in order to satisfy a relatively problem (e.g. "How do I set up a simple xover at 1500Hz, attenuate a peak at 3500Hz by "X" amount, and add an LF boost of 6 dB beginning at 35Hz"?)........
Of course, it will never be this simple - but the parts exist already. So how do we use them?...............
:rolleyes:
Bill |
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| WTS |
Hi; Now thats what I'm looking for Netqeek. 1 in 3 or 4 out. No A-D or D-A converters needed for me, strictly I2S. A nice interface(rs232) into windows for programming the xovers, boost, cut etc and it should have a master level(volume) control that can used without the windows programming interface. Is this too much. Would be perfect, that way I wouldn't need to buy a Behringer and only use a faction of its circuitry.
Walter |
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| fcserei |
As the perfect solution is still not available how about this:
An older EV DSP with freely configurable internal signal flow with gain, iir filter, crossover, mixer, delay, dither modules - married through i2s to a Panasonic digital amp board in box with battery supply.
Digital input with clock sync to transport. Analog inputs can be added as modules, but not in use now. The DSP has 8 output channels but I'm using only 5 for the 5*130W output.
PC configurable, but without the pc there are two buttons for volume up/down.
Total cost is ~$300.
A two box system (transport plus this box) for a fully digital multiamp chain. |
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| JasonL |
| So does this thing make toast..? |
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| dmfraser |
D2Audio make a decent 4 channel power amp module, the XS-100 with a good built in DSP. (I was at their party at the Palms in Vegas last Friday night during CES) We are getting a version with FIR filters for making steerable column arrays. The unit does require a fully regulated power supply. The module with a good DSP and 4 x 125W of amplification has an OEM cost of $150.00 (this is the 500+ price I believe) and needs a $150.00 power supply. They said they would sell the eval board with the module and the power supply for $495.00. I don't know for sure because they loaned it to me.
However, the product sounds good and measures good and it is likely we will be buying it from them. They also make a 2 channel version with 375W and 125W and a 3 channel with 250W and 125W x 2. There is also an 8 chnnel coming out.
Overall, while their early products were not up to professional standards, their current products are well worth looking at. |
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