| 5th element |
Graphically that program looks identical to the way the KXdrivers integrate. Its just that the KX is very limited in its application.
There is one problem with these setups. How do you deisgn them around something like LspCAD? If I export the filter gain as a frd file from my filter design, can I import this into the programs you are using so the FIR or IIR will do exactly what I say?
I know this is possible because this is basically what LspCADs emulation tool does using IIR's or what it calls circular convolution filters? What is a circular convolution filter? I own the pro version of LspCAD and have tried the digital filters/emulation on a few occasions, just to check the xover I have designed sounds good. The results using this are really nice and actually sounds fantastic even using the analogue out on a soundblaster live card.
One easy way of doing this (I can see), although maybe not optimal, is just to use the 6 channels of digital out (on SPDIF) on the soundblaster card into three stereo offboard DAC's. I can easily get my hands on two other identical dacs to the ones im using (DIY ofcourse!) for not much money.
Digital filters like this would be brilliant for me as I dont stick with one thing, I am forever changing this that or another so the flexability they would bring is a good bonus.
Can you also get something which will do this in hardware? The KX drivers use the EMU10k chips on the soundblaster which has one main advantage; I can collect together the sound outputs from every application used within windows and all the inputs on the soundcard (line in etc) and THEN put them through the DSP, in otherwords the DSP comes last in the signal chain before exiting the soundcard. Can you do this with the ways you are describing? It seems to me that the digital filters decribed are just another program that would work in the same way as the LspCAD emulator. Although this would add flexability (in the form of digital xovers) it would also have unacceptable disadvantages. This is something my analogue active xovers get around quite nicely but are much more limited in their application. |
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| Vil |
>>>Can you also get something which will do this in hardware? The KX drivers use the EMU10k chips on the soundblaster which has one main advantage; I can collect together the sound outputs from every application used within windows and all the inputs on the soundcard (line in etc) and THEN put them through the DSP, in otherwords the DSP comes last in the signal chain before exiting the soundcard. Can you do this with the ways you are describing?
I can mix at same time any signal from wave drivers (they are multiclient) and one physical card's input (line , mic , aux )and also digital s/pdif in . |
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| 5th element |
| So if I am set up correctly can I just run the xover program and do ANYTHING within windows, including playing games and having their sound effects decoded, and the sound be filtered correctly? |
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| Vil |
| yes you can do this , but you must have enough cpu power for all tasks :) |
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| RyanC |
Hey 5th-
There are a number of "pro" cards that will typically offer better performance in some ways then soundblasters. The primary thing about the pro cards to me is ASIO 2 drivers. ASIO drivers were developed by steinberg and they are a kernel level audio protocol- which means that there will be less CPU overhead at lower latiencies. WDM is also good- but only asio 2 offers dedicated reference Clock channels via the driver protocol to essentially slave the application to the reference clock (I don't know that soundblaster doesn't offer this, but ASIO does not = asio 2). Otherwise the programs (via asio1 or WDM) derive the clock info from the audio stream, which introduces jitter and I believe has caused many to throw out the baby with the bathwater when it comes to computer based XO's (and computer based recording). I would recomend searching for asio with spdif or aes/ebu outs. Off of the top of my head the lynx AES would make a great choice (altough a bit overkill).
As for measurements/tweeking these setups are extreemly flexable, these EQ's were all designed for mixing/mastering applications, so typically you get full parametric EQ's shelving HP LP all usually with variable Q. The Q on digital filters will typically go much lower then any analog component filters you will find (or make). You can also get plugins with up to 192dB/oct slopes (not fir) using elliptic or other "no pass band ripple" math. Also because there are no component inconsitancies they will always be more accurate in terms of exact pole frequecies etc. The only thing that would hinder this is the sample rate and basic nyquest theory, so higher sample rates are better for this (even if you have to upsample).
I would still recomend having a dedicated box over the all in one setup. You can distroy tweeters if you comp glitches, but like I said you can certainly check it out with 1 comp and if you like it go from there.
Typically the pro sound cards offer even more routing options- Rme gives you the ability to create different mixes (not just patches) with different inputs and most offer multiclient functionality. The lynx cards are probably the best- many pro's swear that in double blind AB comps they actually sound better even with an all digital path and the same external clock source and converters. There are many factors to doing computer digital audio right, and I don't believe (personally) that soundblaster is seeking the optimum performance, unless you are talking about a nasdaq index.
The most simple solution is also to use a short spdif cable and loop 1 stereo out back in. The lynx cards have a built in function for this (like vil's little patch bay window).
I would say that you can expect better performance than LSP cad-both in terms of CPU effency and sound quality and flexablity. Plugins are a highly competative market, and when you use them to mix you learn what they are really made of (as far as the math goes).
The waves lineq package is somwhat limited in terms of q and the types of FIR slopes, which as I understand it is part due to the nature of FIR filters.
Anyway check it out! - I doubt you will be disapointed-
RC |
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| 5th element |
Thanks for that.
Two things one a question one a bit of information.
First the question - one thing that was left unanswered (I think) is this. Can I export a filters gain, in otherwords what the filter does (see pic) as a frd file, then import it into a "filter plug-in" and the digital filter will copy EXACTLY what I have designed? This would seem like a very useful feature and rather necessary in order to get the best results from the digital filters.
Two - About tweeter destruction if the computer messes up. My active xover creates horrible turn off thumps, so naturally you power down the power amp before turning off the active xover. Sometimes you forget, my tweeters (scan97s) live to tell the tale. They have a 70uf cap on them to stop DC reaching them, so I guess they would survive computer glitches. |
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| RyanC |
Oh and 5th-
You cannot play games with an FIR XO- you will need latency below 20ms total in order to percieve that there is no lag. AFAIK there is no way to do FIR filters below around 100ms total, so see the punch THEN hear the punch. You can however use an IIR filter XO and a card with good kernel level drivers and have 1 realtime XO and one non-realtime FIR setup for different applications. The IIR setup will also eat up less CPU which is good if you are trying to play a game.
Also i was thinking about the tweaking and meausreing thing more. Here is what I do, I have a XO setup with no or little corrective EQ (drivers or room, so just HP LP or HP + LP=BP filters). Then I use the voxengo curve EQ for corrective filtering. I analyze pink noise (actually I prefer a slight HF recession so I use dark pink noise but not so dark as brown). Then I pump pink noise (not the dark pink but just regular pink) through the system, I walk around listening position with a measurement mic and use the curve EQ's match functionality. It will match the mic analyzed spectrum to the reference one using 60 bands of FIR filters.
This will correct for a fair amount of driver and room problems. If you have say a seas driver you can just throw some notch filters in the chain if you would like to correct it's break up modes. Or use a couple of LP filters at the XO point to create a very steep slope. This process will make a hobbled together system sound pretty damn good, let alone a well designed one.
RC |
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| RyanC |
Oh ok-
"First the question - one thing that was left unanswered (I think) is this. Can I export a filters gain, in otherwords what the filter does (see pic) as a frd file, then import it into a "filter plug-in" and the digital filter will copy EXACTLY what I have designed? This would seem like a very useful feature and rather necessary in order to get the best results from the digital filters."
No- not directly like that. But if you know what the filters are you can easily duplicate it. EG parametric notch, -2dB @ 2Khz with a q of .707- you just fill in the numbers. Same with HP LP filters. Just look at the dB/oct. Plugins give you a WYSIWYG FR plot so if you look at it's graph and it says 18dB/oct then that is what it is. Some eq plugins use the Q as feedback resonance- you should avoid those for XO's generally. Reguardless it would not be difficult to duplicate your response graph shown, using a plugin with WYSIWYG graph, assuming it allows for the same filter types (notch, hp, bp, lp shelving, elliptic etc). Of course as the EQ plug is not passive in nature you are not limited to cuts, so I would keep that in mind.
Otherwise just do the testing/tweaking together- meaning setup your basic HP BP LP filter and then pull out the mic. I prefer to look for actual listening position trends then 1m (semi) anachoic issues- if your system tends to over exentuate ~85hz and your room tends to attenuate it (hello 8ft cielings) then it is NOT a problem, right? Unless of course the bandwidth is different of the null/boost. And of course if you are in a REAL anachoic chamber your mic at 1m will be in a null at 343hz and all direct multiples thereof.
If you have a perfect room (this does not exist by the way- including anachoic rooms) then you want perfectly flat speakers (except I would still prefer a slight non-lumpy recess of the HF). A room will typically take speakers that are +-1dB anachoic and give you +6, -30dB. If you have 8ft cielings with no bass treatment you will have a -15-40dB null around 80-100hz especially if you sit with your ears at the 1/2 wavelength of the vertical node- (which is about where everybody sits). Then if you sit in the center of the L-R walls you are another node! And if you are in the center front to back add a 3rd one and those are just the fundamental room modes (parrallel walls suck).
I guess the point to me is that a dB here or there will not make or break anything- overall design of the room and the speakers that are tailor made/tuned will give you better real world results then what lspcad says in your treated booth (or wherever you do your measureing). But anyway set it up in LSP- do the test, change the filter, redo the test until it, A- sounds right, and B looks decent. Or don't look at it because as Joe Meek said- "if it sounds right, it's right". Then again I say "If it sounds right, it's probably overcompressed". :smash:
For the cap that should be fine- The computer glitch will tend to be a click or pop or stutter type thing. They do not contain an incredable amount of low end (usually). I would still leave the cap and for me if my intended XO freq was 2k I would set it up for 1st order at 1k or 1.5k. I tested my setup with and without the caps 1 oct below the active XO point and I didn't feel/measure that there was a difference. Not really an accurate test tho as It takes a minute to swap them in/out (on the subjective side).
Ryan |
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| mbutzkies |
Shin: It seems you have come back to a PC based Digital Crossover, if you are going to use it both in Home Theater and Music, you need to do a few things.
Lip sync: Video and audio have different latencies, HDTV depending on processor could have up to 130ms of delay, LCD screens have 10-20ms, and Plasmas can have up to 60ms of delay. Typical HDTV processors add delay to the audio stream to compensate. If you have a PC based FIR crossover and DRC you may have to delay the video, or you may get lucky and have enough delay already. Lip sync adjustment will probably be necessary. I have not looked into time shifting software yet.
The Waves Filter has a fixed number of taps, hence a fixed amount of delay. The filter alone creates 60ms of delay exclusive of processing. The Waves filter is for studio environments not for live sound. If you create your own filter, you could do much better. The min delay for a FIR filter depends on frequency, XO slope, and sampling frequency. Because you are using a dome, your crossover will be probably 500Hz+, which will help a lot. A 500 Hz, 60db/oct, 44100 kHz FIR filter would give you about 5ms of windowing delay. You must use the same amount of taps for your W-M cross as your M-T cross. Because Waves has fixed taps, you will be okay there; if you make your own you need to be aware of it. The lower in frequency, the steeper the slope, the higher the windowing delay.
Of course if you only use it only for music, no worries.
Summing Flat: FIR filters are not all linear phase (0 Group delay), nor do they automatically sum flat. You must know which methods they use. Windowing FIR filters will sum flat. The Waves module is really an EQ module and they present you with 3 methods. I believe the low ripple method sums flat while the accurate method does not. Please consult with Waves as I am not 100% sure on that.
Stop Band ripple: FIR crossovers have ripples in the stop band. The Waves filter has ripples 60db down in accurate mode, 100db is low ripple mode. A poorly written FIR filter might have ripples only 20db down so be careful.
For further reading; LinearX has a good write-up on their website, and Waves includes a small PDF.
FIR filters for Home theater use is pretty state of the art, it is not as well documented as more traditional crossovers, but if you are aware of those three things you should be okay.
As I mentioned before, a guy named AEDIO created a FIR crossover named divider for Foobar, you should check those out. A quick search on Hydrogenaudio.org will locate the dlls for you. Waves are considered some of best pro audio plugins so those are good as well. There are some problems with AEDIO’s plugins-cannot use with home theater(Foobar only) and they don’t work with kernel streaming or ASIO, only Direct Sound or waveout. But they are Kaman windowing FIR filters with 100db ripple attenuation, adjustable taps(delay), and they sum flat.
Eventually I plan to write my own FIR crossover filter, I haven’t found a FIR crossover filter that exactly fits my needs. |
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| 5th element |
Using a complicated FIR and CP hungry filter for music is fine. Using a much more economical and less complicated IIR for games would be no problem either. Im actually near field (less then 1m from each speaker) when playing games so maybe a different setup might be a good idea anyway.
Ah I see what you are saying about those filters. LspCAD lets you use up to 8th order i think 48dB/octave is plenty. This would let me see what a 1.5k xover sounds like with the seas (like you mentioned) W15CY and the scan97. I know the scan can handle 2.2khz with a 3rd order acoustic but I dont know about 1,5khz. A steep slope would suit the W15 even more by reducing the additional harmonics induced by the resonance at 8.2k.
I would still design using LspCAD mind you to make sure I get the phase response as perfect as I can. You method sounds ok but with one flaw. -Does the curve EQ thingy in vox tailor for phase like the Tact audio stuff does?-. If it does then its not a flaw. You can have a flat response when the drivers are in phase and a really **** notch when polarity is reversed, meaning the drivers are not in correct phase. Ofcourse adding in delay circuits on your tweeter will solve this, but it will take measuring to get correct.
Cheers for you responses. |
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| RyanC |
Ok a couple of things-
In cubase or Nuendo the delay incured by the process (plugin) is compensated for automatically- So if you have a steep filter on one driver and a soft one on another the difference in delay incured by the plugin "should be" handled for you. I do not believe this is possible with art teknika but either way it does not matter because-
I just setup a sine at the -6dB point and use the (free) voxengo sample delay to allign the drivers at the XO freq. I am alligning them at the listening position (That is what vil is doing too it looks like). I'm not really ready to drop the $ on a LSP so I don't have anything now that will give me a full phase allignment test, so this is all I have right now. But you could use my technique and then measure the overall allignment with lsp. Reguardless you are left with only the phase inconsistancies of the drivers/power amps if they are alligned at the XO f6. I do this by recording the signal back in to the computer from each driver- this way you can assure that you are not delaying it by a full wavlength or more to achieve allignment (and that you are doing it to the right driver).
This is as I see it the primary advantage of digital XO's you can delay any band (easily) to achieve the best inter driver temporal allignment.
I am not familier with the tact stuff- but AFAIK only an FIR process could tailor phase independant of FR within a given driver (It makes sense because only FIR can do one independant of the other in reverse). I have been begging for a parametric phase adjuster though, who knows, maybe soon. The curve EQ is phase linear so it does not effect phase of the signal. As is the case with the Lineq's. You could create an FIR XO within an XO to shift a certain part of the signal later in time- it's hard to say what that would do for you though as they would comb filter when mixed back together (so you would need steep slopes).
A couple of thoughts- you might be suprised how much better it sounds to use either a steeper filter or a lowQ parametric centered well above the XO freq to completely eliminate those modes from the seas drivers. At least on my sub drivers, TC2+'s, the difference was quite audible, but suprisingly not very measureable. This may not effect overall FR that much (due to the hi q nature of break up modes?? I dunno) but reguardless your driver does not sound good at 8k (I would guess by lookin at the FR plot) So it would be better (to me at least) to have it not produce any 8k.
Also my personal feeling is that dome tweeters (or any tweeters really) are rated for what they can hadle heat/xmax wise, not where they sound good + are low in distortion. I discoverd this by listening to just tweeters and adjusting the XO freq- I find that tweets in general tend to get very harsh at the bottum of their range especially in higher SPL situations.
Cone mids and mid-bass's produce a much preferable 3k-5k to my ears than tweets in general (now to find one with good off axis response). But then inter driver spacing becomes a problem. I can't speak for your tweets tho as i have not used them- and I don't have a ton of experience with the higher end drivers (yet).
Anyway yea- use lspcad, just don't get caught into the mindset involved in the inherinet limitations of passive filters as many of these are not present on a digi XO.
Good luck-
RC |
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| ShinOBIWAN |
I've had some playtime with both the Behringer and the Cubase setup.
Both offer fantastic quality, I'd give the nod to the cubase implementation regarding quality but TBH you have to sit and really listen to notice the differences without room correction. Turn room correction on and the cubase setup really moves up a gear. Everything is cleaner, clearer and very sorted.
I've got issues with both setups however. The behringer has no pre-amp section and the cubase with 3-way DRC, FIR and time/phase alignment is a massive CPU hog. Games stutter, frames are occassionally dropped in theatertek and it has a steep learning curve at first.
I really don't know which to go with actually. For all its evils the cubase XO is incredible. The behringer XO is good but without the DRC its lagging.
Because I feel the difference is very small between the cubase and behringer WITHOUT DRC. I think I'm going to go with a hybrid approach, let the PC do room correction and the behringer can do the XO. I still haven't tried this yet but I can't see a problem. |
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| mbutzkies |
It looks like you are going to need a quad athlon.;)
RyanC: The problem of delay is not with music, it is syncing video to audio, and the fact that video would get ahead of audio, you would have to timeshift your video for that. Hopefully you could fix with limited rigging. I don't use nuendo, but I guess you can play video through that but using a multi-track mixer would be an extremely cumbersome setup. If its all audio, you can just add additional delay to sync up like you did. The problem with FIR filters there are two delays, a windowing delay that will not change with hardware, and a processing delay which is hardware dependent. For simplicity the Waves LnEq has equal delay whether your Q=1, or 6.5, that is also why the eq is not completely adjustable.
5th Element might have the idea of just using two different crossovers. |
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| RyanC |
Cool yea-
Shinobiwan- The room correction is pretty damn awesome - I think if you are the "set and forget" type then the DCX or somthing like it is probably best. But for the endless tinkerer (me) I like the computer. The other thing to keep in mind is that the DCX runs on its own clock with no sync (it always runs at 96k) therefore it runs free. And some $2 SRC chip upsammples at an non multiple rate (EG it would be better if it ran at 88.2 for CD playback) and meshes the two clocks. Your performance now might only be a bit better- but if you bought a big ben or the antelope box, the difference would be even bigger due to the DCX's lack of proper clocking.
mbutzkies-
Right, but if you are inside the 20ms window you will not precieve the latency- Of course if you sit 10ft from your speakers there is already 8ms of latency due to the fact that sound moves alot slower than light. If you brain didn't allow for this sound would always be out of sync with sight, as light always moves faster then sound.
Reguardless that is why we are talking about 2 XO setups- a realtime IIR setup (I have mine down to about 8ms round trip) and a FIR setup for non realtime applications. This way when i am playing my keyboard through the system I use the IIR setup and when i am mixing/mastering i use FIR.
RC |
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| ShinOBIWAN |
Ryan,
I've got a friend interested in setting up what you did for me with those nuendo files. Problem is he's running a copy Cubase VST 5.
Would it be ok if I passed on your email so you could have a chat with him? |
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| RyanC |
Sure-
But Unfortunatly I don't have a ton of experience with v5. I was into sonar in those days. I don't know what the routing imitations are-
RC |
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| ShinOBIWAN |
Ryan,
He has access to a copy of Magix Samplitude 8 at his workplace.
I've told him that the Cubase 5 is rather old now - 5 years now I think.
Would this be better? |
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| ShinOBIWAN |
Vil thanks for that.
I saw your post earlier in the thread but I dismissed the setup as inferior to the cubase one.
I've had a play with the demo version and had an identical setup to the cubase one in under 5 minutes! The sound is identical too. I think Ryan said that SX2+ compensated for the latencies of plugins but console sounds no different to me.
I've also noticed that CPU usage is slightly lower :)
All in all I'll be building a dedicated PC for this XO and use console instead of cubase and bin the DCX :D
What's the best plan of attack? Do I move the RME into the dedicated machine and buy a prodigy or Revolution for the main machine. I'd like to keep the signal digital if possible all the way to the amps as I have done now with the DCX.
Really just tell me how you'd do it. |
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| Vil |
plugin delay compesation is not important , because you need individual regulated delay for every driver anyway , so you can adjust right delay time for every chanel. also automatic delay compensation somethimes works not good and you can have very strange things going on (thats my personal expierence with Steinberg products ergh)
yes it would be nice to have a digital signal up to DAC .
if you wanna use S/pdif or ADAT digital interface for data to DAC transmission , you will not have best possible result . those two interfaces are very jitter sensitive .I started with s/pdif few years ago and finished with 4 balanced pair interface I2S , transmitted using LVDS transmitters .
Also dont forget lowest possible jitter master clock oscilator for clocking your DACs and sound card .No way to use asynchronous sample rate conversion , especially with Crystal CS8420 chips , that one kills music . those from Analog (AD1896A) are better but still some compromisses . |
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| ShinOBIWAN |
Another week another paint job :D
This time real pearlescent:
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| ScottG |
Uh-Oh.
Now your going to have to lacquer your seas woofers with multiple coats of c37 - which is expensive..
Check out the left side link C37 Speakers and you'll see what I mean.
http://www.ennemoser.com/
Be forwarned though, the speakers presented on this page have MULTIPLE coats of lacquer.
Here is a review on the stuff:
http://www.tnt-audio.com/accessories/c37_e.html |
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| Vigier |
| quote: | Originally posted by ShinOBIWAN
Another week another paint job :D
This time real pearlescent: |
Damn....this looks more like porn to me! ;)
A really beautifull finish!
Grtz, Joris |
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| GeWa |
| Damn, that thing looks shiny!!! |
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| ShinOBIWAN |
| quote: | Originally posted by ScottG
Uh-Oh.
Now your going to have to lacquer your seas woofers with multiple coats of c37 - which is expensive..
Check out the left side link C37 Speakers and you'll see what I mean.
http://www.ennemoser.com/
Be forwarned though, the speakers presented on this page have MULTIPLE coats of lacquer.
Here is a review on the stuff:
http://www.tnt-audio.com/accessories/c37_e.html |
I wouldn't dare stick that gunk on a half decent speaker.
Probably a waste on the Seas too, they're aluminium and the C37 seems to be for stiffening - alu is very rigid compared to paper cones which is what I suspect its aimed at looking at the speakers on the website - fostex, supravox, volt etc.
I think the ATC already has something similar on it - cheap looking gunk that is. |
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| Byrd |
Lol - A 'new' all magical audiophile sauce component :D. Has anybody tried it on their optical pickup lenses yet ;)
In the review they write about "speed mismatch". I would probably estimate that the entire frequency response was altered when the woofer was doped and thus the *carefully dezigned* crossover is now not optimal.
The aim of a cone is to be as light as possible to ensure that transient response is acceptable, while being as stiff as possible to avoid breakup. What this solution offers is a way of altering the ratio of stifness to weight chosen by the original dezigner of the driver. This is fair enough - especialy for the DIYers, however I think the view of what this treatment can do is overoptimistic. There will always be some compromise
ITO aluminium as cone material - it is one of the lightest most rigid materials used ( the only problem being the inherent resonance of the metal ). I doubt weather the laquer would have any positive effect of this driver.
The biggest problem with this of course, as stated in the review, is that it is not reversable. It is also possible that at some later stage in the life of the treatment it may crystalise, fracture and loose it's rigidity - leaving you in an even more precarious situation.
Shinobi - whew. Those doodabs are looking far too good. You have too much time on your hands. I could just image Anakin on the couch next Padme punching a button on his remote control and Taan Tan Tara - The sound belowing forth from beasts looking just like that. The pearlescent would go very nicely in the scenes done out at Lake Como |
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| edjosh23 |
ShinOBIWAN,
Any pictures of the rest of the enclosure?
Josh |
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| ShinOBIWAN |
| quote: | Originally posted by Vigier
Damn....this looks more like porn to me! ;)
A really beautifull finish!
Grtz, Joris |
ROFL
:D
| quote: | Originally posted by Byrd
Shinobi - whew. Those doodabs are looking far too good. You have too much time on your hands. I could just image Anakin on the couch next Padme punching a button on his remote control and Taan Tan Tara - The sound belowing forth from beasts looking just like that. The pearlescent would go very nicely in the scenes done out at Lake Como |
| quote: | Originally posted by pinkmouse
Very nice! :) |
Thanks fellas, appreciate the kind words.
Also Byrd, love the Starwars movies so your comments gave me a good laugh.
| quote: | Originally posted by edjosh23
ShinOBIWAN,
Any pictures of the rest of the enclosure?
Josh |
To be honest they're only at the primer stage, I wanted to be really happy with my choice of colour before doing the lot. Now I've found it they won't be far behind.
Should have pics of the finished cabinets within the next couple of weeks. Then I can sit back and enjoy! Looking forward to it. |
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| ShinOBIWAN |
The Behringer didn't last very long :devilr:
Its for sale in the trading post if anyone's interested in a virtually brand new example at a good price.
Damn these FIR filters are addictive, started looking at prices for the Antelope Isochrone OCX - methinks I'll have to save for that one. |
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| mbutzkies |
So what is your final solution for your crossover?
Did you get a satisfactory solution to work with TV and DVD-video? If so could you highlight your solution.
On another front, Your speakers look awesome and it looks like you are close to the finish line, It was great to see all the work involved to make a reference pair of speakers. This thread was a significant contribution to this site in my opinion. I think it would be informative fo everybody to document the time and cost it takes to build those speakers. I estimate just on direct raw materials you spend close to $5000 USD. I think newbies would like to know the man hours in research and design, and the actual build. These projects always seem simple at first!!!
I, however, am most interested in your paint, I have never been able to do a good paint job like that. How much time and materials was involved in the finish. I have always farmed out finishing, and I would like to get better at it. |
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| ShinOBIWAN |
| quote: | Originally posted by mbutzkies
So what is your final solution for your crossover?
Did you get a satisfactory solution to work with TV and DVD-video? If so could you highlight your solution. |
Yes TV, DVD and gaming all work very nicely. The only problems encountered were CPU usage. With all the option such as FIR filters time alignment and DRC it floats at around 20% which doesn't sound like much but I use heavy post processing of the video in Theatertek and with these options enabled I get some dropped frames on occasion.
Gaming again runs slower than without the XO, not hugely but I like it to be as fast as possible.
This will all be solved soon enough when I move out the XO tasks to a dedicated high powered machine for low latency <20ms.
The software setup I use is very similar to Vil's: Console as the plugin host with Waves Linear Phase EQ for FIR crossovers, Voxengo Curve EQ for the DRC and Voxengo Delay for the time/phase alignment.
On the hardware side I use an RME HDSP 9632 sound card and ASIO 2.0.
In the future when I have finished these I'm looking at upgrading my master clock to the Antelope Isochrone OCX. But at the moment I'm very happy.
| quote: | | On another front, Your speakers look awesome and it looks like you are close to the finish line, It was great to see all the work involved to make a reference pair of speakers. This thread was a significant contribution to this site in my opinion. I think it would be informative fo everybody to document the time and cost it takes to build those speakers. I estimate just on direct raw materials you spend close to $5000 USD. |
Thanks!
| quote: | | I think newbies would like to know the man hours in research and design, and the actual build. These projects always seem simple at first!!! |
I myself wasn't quite prepared for the task, it seems to have moved on significantly from my original plans.
With regards to time, well I spent a month modelling, researching and designing the enclosures/drivers. I'd say around 70-80hrs spent on this stage.
The construction is still ongoing but its been around 3 months now with a lot of time investment - actually aside from work, nearly every spare hour I have goes into these, so a big investment from that point of view. Upto now around 400hrs have been spent on the building. This does include constructing all the electronics aswell though.
| quote: | | I, however, am most interested in your paint, I have never been able to do a good paint job like that. How much time and materials was involved in the finish. I have always farmed out finishing, and I would like to get better at it. |
Finishing with spray materials is quite laborious BUT easy if you put the work in. Here's how I do it:
You have to extensively prep MDF prior to any finishing coats otherwise you get something of a dog dinner.
I build up a 5 coat thick layer of specialist MDF primer, which locks the surface and prevents excessive soak up of the finishing coats. I do this with a brush since its nice and quick if rather unsightly at first, I then wet sand the MDF primer flat using 300-400 grit paper.
Afterwards I use generic automotive grey primer and only apply 2 or 3 coats and just one wet sand with 800grit after the last coat of grey primer.
Now its time for the finishing coats, the pearl you see in the photo's requires 3 different types of paint to get that effect going. The first is a basecoat which set the overall tone of the colour, I chose 'Brilliant Black' for this. I applied 2-3 coats then sanded back with 800grit and then carefully added one even coat over this with no sanding afterward.
After the basecoat you then add your pearlescent effect coat, depending on how this was mixed and with what pigments determines the effect you see. My pearlescent mix had purple, green and blue pigments mixed together.
You only need 2 very even coats with this stuff and absolutely no sanding!!! This is a pearl finish and like metalics it doesn't like sanding or abrasives at all.
Onto the basecoat you spray this pearl mix and that is when you see the effect. With no light hitting it mine is a very dark purple, almost black, since I used a black basecoat. Use a lighter basecoat and the effect changes substantially. After experimenting on scrap MDF I found that black gave the most dramatic and pleasing colour shifts. Depending on the angle and light you see black, purple, pink, blue and green. With all shades in between.
The final coat is the clear coat laquer and this really adds to the effect pearl effect, deepens the finish, adds gloss and most importantly protects.
The clear coat is suitable for sanding so you can really go to town on flatting out the surface for a mirror reflection - something I still need to do to the cabinet in the picture above!
Here its best to add a good few coats, I normally do around 5 or 6 and get a really deep look to the finish after sanding and buffing.
I use 1200 grit to flat the surface and then go to a fine rubbing compound such as Safecut from Turtlewax, then I use a high quality polish/wax.
After all this you should have something that looks very nice. Like I said its quite laborious but well worth it once you stand back and take stock of the finished article.
Also take special care to equip yourself with a good vapour and particle mask. When your spraying with this quantity of paint it really pays to protect your health. I have a friend that used to graffiti regularly in his younger days and what I'm about to tell you is absolutely the gods honest truth:
He's told me stories of when he used to work for hours at a time on a large piece of work and during/afterward he had some quite scary health issues related to the spray paint. Things like burning sensations in his throat whilst spray, asthma-like attacks of breathing, tightness of the chest, dizzyness and headaches and blood from the nose. He bought himself a decent coal filter mask back then and all this stopped!
I cannot stess this enough after some of the stories I've been told. |
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| mbutzkies |
I am glad you got your crossover to work. I guess TheaterTek adds some significant latency.
I used Winamp, Adaptx, and Waves linEq, and got lip sync problems but no dropped frames. I also used Aedio divider with Foobar and like that solution too, but it was audio only. Even with those problems I had, FIR and DRC definitely rocks, and someday I will get serious about writing a low latency FIR filter.
Since you firmly crossed over to PC crossovers, might I suggest these cases.
http://www.zalman.co.kr/eng/product...dx=152&code=020
Before you get too excited, I only found one vendor for it at $1100.
A more reasonable case manufacturer
http://www.silverstonetek.com/product-case.htm
These can be had for $100-150, many vendors
Also, do you have test software, it would be neet to see some graphs? |
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| ShinOBIWAN |
Theatertek 2 has lip sync delay so you can compensate easily for that.
I have flawless playback if I disable all the tricks such as Lanczos 8 Tap filtering and 3D Denoise.
The strange thing is even with these options enable my CPU is at 80%, I guess TT needs some headroom. As long as I'm at around 50-60% load I get perfect panning.
I do have measurement software and I will post some graphs etc. after I get all the drivers back in the finished cabinets. |
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| ShinOBIWAN |
Vil & Ryan,
Voxengo curveEQ is giving me hassle in console.
CPU usage is around 20% and I'm getting 2-3second dropout in audio every 30 second or so.
I've tried removing Waves LineEQ's to lower the CPU usage to around 5% but it still does this!
This is with just plain old audio routed from the RME using internal loopback, everything works fine if I remove the CurveEQ plugin.
Any thoughts? |
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| Vil |
Voxengo EQ somethimes does strange things , thats right .
one more plugin to check - Firium .something similar to Voxengo EQ . |
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| Vil |
for Console users I made small usefull script . its starts Console , loads last project file , turns procesing on and minimize . very usefull . free script software is here :
http://www.autoitscript.com/autoit3/
project file attached |
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| Vil |
| there is fast digital frequency measurement of Footbar2000 with divider , and you can see crossover frequencies . |
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| Vil |
| and another one - something similar using Waves Linear Phase EQ |
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| ShinOBIWAN |
Aw man, I can't believe this s**t!
Why am I so stupid? I've completely wrecked the sat shown in the photo's through my incessant colour changing.
Thing is I've had 5 colours on there now, so I'd say around 20+ layers. This of course meant that the finish will never dry or at least not within the next 6 months or so.
Now the funny part comes when I decided to strip back the paint to the MDF and redo. I started with some 50 grit wet sanding but because the paint was permanantly tacky it just clogged the paper up real bad, think I used around 5 sheets in an orbital sander just doing around 10 square inches.
So gave that up as bad job and moved onto paint stripper, that just turned the paint to gunk and was very messy and particularly effective.
Then I decided to go all out and remembered how effective heat was when removing old paint from doors etc. So I pulled the blow torch out :D and then tried it on the paint, within seconds it was a flaming ball of MDF! Clearly the paint was considerable more flammable that the stuff you use on your interior doors.
I had to get the hose pipe out to put it out but alas it had been burning for around a minute or two before I could get the hose all locked in and unwound to reach.
It didn't actually look to bad until I started sanding the fried paint, then when I reached the MDF it had turned chunks of it into carbon and it just fell out leaving large craters on the surface :D
My own stupid fault but I can't help but think of all the hours spent on that one sat and then the thought of having to redo another.
I'll post some comedy photo's of the toasted sat when I get the batteries charged for the camera :) |
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| m0tion |
| My condolences for your loss. Trust me man, I know I've been there and I think most of us can relate. You take something you've spent countless hours on (and probably a fair amount of money) and totally destroy it by trying to "get it just right". It's part of the game, but it's the part that sucks. |
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| ShinOBIWAN |
To cheer myself up after the loss of the first satalite, I finished the second one off. What you see here is the final product, no more finishing required.
After seeing the finished results of this one its given me impetus to forge on with the others after a bit a low with the destroyed sat.
 |
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| m0cea |
That looks wicked! :hot: :hot:
Not WAF friendly though.:D |
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| Mutley666 |
ShinOBIWAN, that finish looks awesome! A fine job indeed!
How long has it taken to achieve that level of finish? :confused: |
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| ShinOBIWAN |
| quote: | Originally posted by Mutley666
ShinOBIWAN, that finish looks awesome! A fine job indeed!
How long has it taken to achieve that level of finish? :confused: |
Not actually that long.
I started on this three days before I set the other one on fire.
So around a week from priming to finished product. Quite a few hours work though, all depends on how picky you are really, I probably need some psychiatric help in that department. |
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| David Gatti |
Beautiful work Shinobiwan. Truly one of the best I've ever seen!
And never mind, we all have our stuff-up stories. |
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| ShinOBIWAN |
| quote: | Originally posted by David Gatti
Beautiful work Shinobiwan. Truly one of the best I've ever seen!
And never mind, we all have our stuff-up stories. |
David your too kind, I wouldn't go so far as to say they are the best ever seen though :) Although I am pleased with the results. |
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| ShinOBIWAN |
Ok getting close to finishing one of them!
The bass cabinet is all sprayed up but needs sanding and then buffing up like the sat. I'll also wax them both when finished to give even more reflectivity.
Apart from that, the sat needs its isolation spikes adding and the plinth for the bass cabinet is missing but I'm struggling to come up with a colour that will contrast well against the pearl finish. Any idea's???
Here's the work in progress shots of 'nearly' finished speaker:
 |
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| m0tion |
| Can't see those last pictures you posted |
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| ShinOBIWAN |
| quote: | Originally posted by m0tion
Can't see those last pictures you posted |
If I still have the photo's I'll try uploading them again to the webspace.
They were working, at least for me, when I originally posted that :confused: |
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| agent.5 |
| WOW! your project is impressive. Thanks for linking me here. |
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| jleaman |
| quote: | Originally posted by ShinOBIWAN
To cheer myself up after the loss of the first satalite, I finished the second one off. What you see here is the final product, no more finishing required.
After seeing the finished results of this one its given me impetus to forge on with the others after a bit a low with the destroyed sat.
|
This flip flop paint that is pruple blue and red is the same on my friends truck.. just to add NOT CHEAP PAINT EITHER!!! |
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| Vil |
| beautiful . how is ir possible to screew ATC dome not going thru holes from speaker front ?my ATC drivers are waiting for cabinet too. |
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| curva |
| quote: | Originally posted by Vil
I am using Audio Console software with ASIO2 and waves , voxengo etc. stuff for procesing. |
With the amount of money you paid for waves, I would have sure otten much better hardware(talking about xo, amplifiers here) ;)
Otherwise plain amazing look you've got... |
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| ShinOBIWAN |
| quote: | Originally posted by curva
With the amount of money you paid for waves, I would have sure otten much better hardware(talking about xo, amplifiers here) ;)
Otherwise plain amazing look you've got... |
And what makes you think I paid a thing for Waves??? ;)
I've taken a look at everything out there that I could find regarding digital XO's. Trust me when I say the PC is by far the most superior, EVERY element is upgradable leading to better performance than any of commercial stuff. DEQX/TACT look good but your still limited to the onboard DAC's, software algo's and processing power.
Amplifiers are being moved over to KrellClone KSA50's so I should have some class-a amplification. |
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| ogp |
This thread has been very interesting. I have a feasability question. Would I be able to use something like an M-Audio Delta 410 and run my fronts and center as active 3 ways with a parametric eq on my sub(s) while running my surrounds passively? So I could use a computer to do the same thing as a DCX2496 and DSP1124P? Would I have any sync issues if I ran just the fronts/center active while having my surrounds passive? Just doing some brainstorming. Thanks for any input.
Dave |
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| ShinOBIWAN |
| quote: | Originally posted by ogp
This thread has been very interesting. I have a feasability question. Would I be able to use something like an M-Audio Delta 410 and run my fronts and center as active 3 ways with a parametric eq on my sub(s) while running my surrounds passively? So I could use a computer to do the same thing as a DCX2496 and DSP1124P? Would I have any sync issues if I ran just the fronts/center active while having my surrounds passive? Just doing some brainstorming. Thanks for any input.
Dave |
You'd need 12 outputs to do what your wanting. Not if this is possible on the M-Audio 410.
The Parametric EQ can be done via CurveEQ or LinearEQ and both offer FIR phase perfect filters - superior to the DSP1124.
Sync issues aren't a problem for me but I would have to agree that it is easy to be a victim of it. A combination of Voxengo Delay and Theatertek lip sync delay sorted out my very minor sync issue. Moving over to a dedicated PC XO box will also provide much better performance.
You wouldn't have any delay on the passive rears if you passed the signal through just a digital delay in the PC XO. Voxengo Delay would be perfect for this. |
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| ogp |
| Sorry, I was a bit vague. I wasn't planning on having 3 subs. That does bring up another question though. If I have stereo subs, do I need to connect them individually or can they be connected to one output with a y-cable when doing parametric eq? |
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| curva |
| quote: | Originally posted by ShinOBIWAN
Trust me when I say the PC is by far the most superior, EVERY element is upgradable leading to better performance than any of commercial stuff. DEQX/TACT look good but your still limited to the onboard DAC's, software algo's and processing power. |
flexibility, that is the word...
nothing compares to being able to rip all you cd's, compress them lossless to flac or ape or anything else and listen to anything you want without touching no cd at all.
then you got all thos wonderful professional plugins(they are really pro) and pair it with a decent ASIO card and bingo...
BTW, what do you use for ASIO routing? Is that an application that came with the prodigy card or it's a 3'th party app? I used to use winamp paired with a directx plugin adapter for being able to use some plugins, but that's far from what I want. That ASIO router of yours looks really cool :) |
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| m0tion |
OT:
Does anyone know if the Behringer DCX2496 unit has a turn on thumb? My real question is I suppose, if I power cycle the Behringer unit while my power amps are on, will this probably destroy the tweeter in my speakers? I'm not using any passive filtering on any of the drivers. I know this is slightly off-topic, but there are DCX2496 users participating in this thread and I was hoping someone could confirm or deny that this is a problem to look out for. |
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| ShinOBIWAN |
| quote: | Originally posted by Vil
beautiful . how is ir possible to screew ATC dome not going thru holes from speaker front ?my ATC drivers are waiting for cabinet too. |
They're just screwed in from the rear instead of the front. I used some 1" 4.8mm self tappers. They don't fit the threads of the ATC mounting but I use large washers to provide a stable footing.
Just a quick note to make sure you don't choose screws which are too long and end up comming out through the front - that wouldn't be good at all.
Its fiddly but worth it IMO.
| quote: | Originally posted by curva
flexibility, that is the word...
nothing compares to being able to rip all you cd's, compress them lossless to flac or ape or anything else and listen to anything you want without touching no cd at all.
then you got all thos wonderful professional plugins(they are really pro) and pair it with a decent ASIO card and bingo...
BTW, what do you use for ASIO routing? Is that an application that came with the prodigy card or it's a 3'th party app? I used to use winamp paired with a directx plugin adapter for being able to use some plugins, but that's far from what I want. That ASIO router of yours looks really cool :) |
I think you may have me and Vil mixed up. Perhaps I've been replying to posts intended for Vil.
But to answer your questions, the ASIO/VST/DX plugin host me and Vil use is called Console:
http://www.console.jp/eng/en_about.html
I don't actually use a Audiotrak Prodigy soundcard but Vil does, instead I have an RME HDSP 9632 with the add-on boards for more channels.
| quote: | Originally posted by m0tion
OT:
Does anyone know if the Behringer DCX2496 unit has a turn on thumb? My real question is I suppose, if I power cycle the Behringer unit while my power amps are on, will this probably destroy the tweeter in my speakers? I'm not using any passive filtering on any of the drivers. I know this is slightly off-topic, but there are DCX2496 users participating in this thread and I was hoping someone could confirm or deny that this is a problem to look out for. |
The Behringer is a dangerous machine when using tweeters unprotected.
I've read a post on here regard someone destroying some Aurum Cantus G2's after powering on the behringer whilst the amps were on.
The thump is quite severe and I'd always recommend turning it on first and off last. Or better still just leave it on 24/7. |
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| ShinOBIWAN |
| quote: | Originally posted by jleaman
This flip flop paint that is pruple blue and red is the same on my friends truck.. just to add NOT CHEAP PAINT EITHER!!! |
It isn't cheap, £260 for 1ltr and that makes 1.5ltr after thinning. Thats just for the pearl coat, you still need to add the basecoat and clearcoat.
Altogether you can expect to pay £350 for all the paint required to paint a pair of cabinets this size.
The paint I bought comes from these folks:
http://www.rage-extreme.com
And here's a few of their other stuff:
| quote: | Originally posted by ogp
Sorry, I was a bit vague. I wasn't planning on having 3 subs. That does bring up another question though. If I have stereo subs, do I need to connect them individually or can they be connected to one output with a y-cable when doing parametric eq? |
You'll need a very minimum of 12 channels for 5.1, it looks like this:
3 channels for left main (3-way)
3 channels for right main (3-way)
3 channels for center (3-way)
1 channel for left rear (passive)
1 channel for right rear (passive)
1 channel for LFE
So that's 12 channels needed in total.
Start wanting stereo subs and it goes up further.
I really wouldn't advise just processing the fronts (L,C,R) and leaving the subs and surrounds to an external processor. Best to keep it all in the PC to avoid sync issues. |
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| ShinOBIWAN |
Progress so far:
 |
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| edjosh23 |
Wow, those look amazing!
How many hours of sanding did it take for the boxes to look so good, especially with the finish, it should be easy to find any imperfections?
Josh |
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| tiroth |
| quote: | Originally posted by ogp
Would I be able to use something like an M-Audio Delta 410 and run my fronts and center as active 3 ways with a parametric eq on my sub(s) while running my surrounds passively? |
| quote: | Originally posted by ShinOBIWAN
You'd need 12 outputs to do what your wanting. Not if this is possible on the M-Audio 410. |
Delta drivers support up to four cards, or 32 analog + 8 digital channels. You can even mix and match, e.g. Delta 410 for the surround and the much superior Delta 1010 for the mains.
ShinOBIWAN, I assume that your techniques require a card with multiclient drivers; you must be routing the directsound driver inputs to the the ASIO inputs, and disabling the directsound outputs?
Does anyone know for a fact that the M-audio Delta drivers support this? |
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| ShinOBIWAN |
| quote: | Originally posted by edjosh23
Wow, those look amazing!
How many hours of sanding did it take for the boxes to look so good, especially with the finish, it should be easy to find any imperfections?
Josh |
Quiet a few hours, next time I build I'm using void free plywood as MDF requires a lot of preparation to achieve a good sprayed finish.
Its actually taken longer on the finishing than the construction.
| quote: | Originally posted by tiroth
ShinOBIWAN, I assume that your techniques require a card with multiclient drivers; you must be routing the directsound driver inputs to the the ASIO inputs, and disabling the directsound outputs? |
Yes the RME drivers are multiclient. I also use internal routing to move the signals around, the Prodigy which is a cheaper card also has multiclient and internal routing if anyone is interested in trying this out on a budget. |
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| Vil |
| the most important thing is to have posibility to route digitaly sound from wave driver to ASIO and then back to card's out . just few cards can do this . RME latest ones , creamware , egosys (audiotrack), maybe a couple more . m-audio don't have that option , at least for a moment . |
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| dwk123 |
| quote: | Originally posted by Vil
the most important thing is to have posibility to route digitaly sound from wave driver to ASIO and then back to card's out . |
I'm pretty sure the Emu's can do it, but their wave out driver is WDM only - not sure if other cards will do this for say KS (or directsound). |
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| m0tion |
ShinOBIWAN:
Could you please post a screenshot of your CONSOLE setup? |
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| ShinOBIWAN |
| quote: | Hi, I saw your post on how you use Console and various VST's in your speaker
project, and since I have a computer as my source for my current speakers I
decided to download Console and some various VSTs (Right now I am trying
GlissEQ).
How do you get this to work exactly? Can console work if the same computer is
the source and the output, or do you have to have a seperate source to feed into
the line in, which gets processed and pushed out the line out?
Does that make any sense? |
I received this email from bjackson.
Rather than answering this via email I thought it better to put it up here so as others may read and gain information.
Right, well you really want a card that affords internal routing of signals whilst still in the digital domain. Why? Well you can route these physically through a line out than back in through the line-in if you use the PC as source and XO but this costs you in both outputs used/wasted and more importantly one digtal-analogue conversion and another analogue-digital conversion when the signal comes back into the card.
So really you need a card that offers internal routing such as an Audiotrak Prodigy, RME Hammerfall series or Creamware. |
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| cph2000 |
This is turning out to be a a very interesting thread.
Could you be more specific about what exactly to look for (e.g. wording, feature) in the company hype, when looking for a sound card that supports this digital routing. |
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| Vil |
there is my fast modification of Audiotrack Prodigy card .
I found 4 I2S data signals onboard and transmitted them thru CAT6 cables to external DAC's .
Also there is located low jitter oscillator for clocking of everything . |
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| m0tion |
| Would the Delta 1010LT be a good candidate for use with CONSOLE and VST plugins? Also, what kind of processor would be needed to 3-way crossover, equalise, etc a stereo audio stream with a seperate subwoofer audio stream? Would an Athlon XP 2800+ w/ 1GB ram do it? |
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| ShinOBIWAN |
| quote: | Originally posted by m0tion
Would the Delta 1010LT be a good candidate for use with CONSOLE and VST plugins? Also, what kind of processor would be needed to 3-way crossover, equalise, etc a stereo audio stream with a seperate subwoofer audio stream? Would an Athlon XP 2800+ w/ 1GB ram do it? |
Difficult to say, looking at the website they hint that the card may have internal routing: "software driven patchbay/router"
You really need to ask a dealer some questions to be fully sure otherwise it might not do what you need.
I can say that Audiotrak cards and RME HDSP cards all have internal routing for sure. Others have suggested Creamware does to. You'll also need a card that is multiclient - again the RME and Audiotrak cards do this.
Regarding CPU power, I'd certainly say the more the better. Even better still would be a dedicated box if you plan to use the XO with games, DVD's and external inputs. |
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| agent.5 |
Just wondering if anyone here tries a software based routing instead of using the soundcard to reroute signals. I found this VST host that will do routing to multiple channels.
http://www.xlutop.com/html/chainer.html |
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| agent.5 |
| quote: | | Also, what kind of processor would be needed to 3-way crossover, equalise, etc a stereo audio stream with a seperate subwoofer audio stream? Would an Athlon XP 2800+ w/ 1GB ram do it? |
probably, if you do eq before crossover. Of couse, you can do eq after crossover, and every channel will have its own eq, but you will use up a lot more processing cycles.
For those who want to move the PC with noisy fans to another room, or those who need to add more processing cycles to their audio system, one solution is
http://www.fx-max.com/fxt/
Run the big computer with noisy fans in a separate room via a LAN, and control the VSTs using a computer with no fan, say something with mini-itx form factor. |
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| Vil |
>>>>>Just wondering if anyone here tries a software based routing instead of using the soundcard to reroute signals. I found this VST host that will do routing to multiple channels.
thats something like Console , but with this you still need any sound card with possibility to route from wave to> ASIO and> analog out . its just one way to do this without sound card support - Virtual Audio Cable . but it works not good , and no 24bit support (at least for a moment).
software solutions for ASIO FX are :
Cubase SX 1,2,3
Nuendo 1 , 2 , 3
Logic 5
SpinAudio Virtual Mixing Console
ARTeknika Console
SpinAudio AsioFX
Xlutop Chainer VST
after testing ALL of those , I found the best one for Xovers:
ARTeknika Console
bad things about console :
there is no autoload of last project and procesing on option at Windows start , so for this I wrote script (please read my post few pages ago) . |
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| Vil |
>>>For those who want to move the PC with noisy fans to another room, one solution is http://www.fx-max.com/fxt/
no thats not a solution .
I have totally no noise PC with enought cpu power up to 96k/24bit FIR . my point of view :
take Pentium M cpu ~1.6-2,13 ghz .
use Asus ASUS CT-479 CPU Upgrade Adapter - http://forums.legitreviews.com/about1570-0-asc-0.html
reduce CPU power voltage to 1V - 1.1V (works without any problem at ~2ghz)
reduce Cpu fan speed to ~700-900rpm (thats NO NOISE AT ALL)
use fanless power supply
use notebook HDD for less noise .
use acoustic damping stuff insigt computer case
use 40x4 LCD display for foobar2000 monitoring
use something like Girder +com port receiver for remote control
use low jitter master clock generator <- very important !
use good sound card (or made right modifications on cheap one)
use ape or wav or other lossless sound format recordings
use good quality DAC's
use good quality amps
use good quality speakers
use good quality cables
think about mains voltage filtering
AND ENJOY THE SOUND
thats my system . sounds just great (but works still in progress) |
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| Vil |
>Run the big computer with noisy fans in a separate room via a LAN, and control the VSTs using a computer with no fan, say something with mini-itx form factor.
you will have jitter isues here (almost for sure ). I had .
I am using LAN cable but not LAN protocol to transmit data !
I am just using balanced pairs of LAN cat6 cable for I2S data transmitting with LVDS , because thats good solution for transfer up to ~500mhz with small jitter .
for DAC /ADC s/pdif aes/ebu or adat works really bad here (my personal experence of 10 years )
>probably, if you do eq before crossover. Of couse, you can do eq after crossover, and every channel will have its own eq, but you will use up a lot more processing cycles.
yes , for sure , its just ONE way to go .probably all of us are doing same way . |
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| ogp |
| So the internal routing makes the Audiotrak Prodigy card a good candidate because you can change the outputs to inputs and such? Because according to the website, it only has 2 analog inputs..... |
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| m0tion |
Could someone please explain what a "multiclient" sound card is and why it's important for this application. Also, what does having "internal routing" mean exactly and why is it important for this application?
Vil:
Whats your take on the M-Audio Delta 1010LT? Also, could you give an example of a "low jitter master clock generator", how it is integrated into your system, and why it is important?
It seems like all the sound cards you guys are mentioning are well outside of my price range, but the 1010LT just barely makes the cut. If it's usable for this application I'd really like to be able to look into it. |
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| curva |
| quote: | Originally posted by Vil
there is my fast modification of Audiotrack Prodigy card .
I found 4 I2S data signals onboard and transmitted them thru CAT6 cables to external DAC's .
Also there is located low jitter oscillator for clocking of everything . |
That sure doesn't look like a fast thing to do... But really looks like a good one. I think I will get myself such a card as well :)
Would you mind sharing with us where you take your I2S from and how you transmit it? Would it be possible to transmit it over a 4-5 meters cable or only over shorter? CAT6 sounds like a nice solution, is it better than good quality coax? |
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| Vil |
>>>That sure doesn't look like a fast thing to do... But really looks like a good one. I think I will get myself such a card as well
Would you mind sharing with us where you take your I2S from and how you transmit it? Would it be possible to transmit it over a 4-5 meters cable or only over shorter? CAT6 sounds like a nice solution, is it better than good quality coax?
well I say fast because I made that pcb in few evenings , not thinking a lot about aesthetical isues . I even don't have any schematics or so , just used data pdf files from National .
look to LVDS family at National .
I tested it with ~15 meters cat 5 cable without any problems so with cat 6 it should work to 20 or more meters .
good quality coax can be solution too , but you will need 5 or 6 cables and much more complicated schematic to send and receive signal with good quality (low jitter) . |
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| Vil |
>>>Vil: Whats your take on the M-Audio Delta 1010LT?
there is no possibility to route wav to ASIO internally .or maybe new drivers will do this , I don't know .
>>Also, could you give an example of a "low jitter master clock generator", how it is integrated into your system, and why it is important?
example o good clock - Elso kwak clock , LC audio stuff etc.
I am using my own design , which I made for LessLoss .
If you have good quality DAC's , clock jitter specifies what you can hear , detailed , nice analog like sound or just messy jittered s******* . |
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| m0tion |
ShinOBIWAN/Vil:
What if I wasn't concerned with being able to output audio directly from the 1010LT and was only concerned with processing external input. Would the 1010LT be applicable in this situation? |
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| m0tion |
Sorry, sorry, I'm really going to have to apologize here. I feel like I am totally hi-jacking this thread, but I have a similar project (conceptually similar, not in terms of quality of components however) going on and I'm at the point where I need to decide what kind of active crossover I'm going to use.
Usually I'm pretty quick with these things, but there are a lot of concepts being thrown around that are new to me. I looked up the price on the Prodigy 7.1 (and the Prodigy 192, which looks a little better) and it is totally reasonable. I want to use it with CONSOLE and the VST plugins you guys have mentioned here (which I already have) to take a stereo input, process it, and output the 6 channels to my 3-way speakers. I'd also like to be able to eliminate my other sound card by being able to output stereo audio as well as AC3 audio from the the same sound card that does the processing. I suppose this would mean using the internal router to route WAV or Directsound or whatever to ASIO and having it processed with the stereo input. I believe that in order to do this, since there may potentially be input audio at the same time that I'm using my WAV/DS output the card needs to be multiclient (is this correct?). I am assuming outputing AC3 audio from this card will be as easy as telling Theatertek to use the digital output from whichever card I choose. I'm leaning heavily toward the Prodigy 192, but also wanted to mention that for around the same price as the M-Audio Delta 1010LT (which doesn't look like it supports internal routing, although I'm almost positive now that it is multiclient) you can get the Maya 1010.
I know I may have turned into a real PITA here, but I'm really making progress with my design concept now. Can someone confirm that what I want to accomplish is possible with either the Prodigy 192 (if not, then the 7.1?) or the Maya 1010? |
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| ShinOBIWAN |
| quote: | Originally posted by m0tion
Sorry, sorry, I'm really going to have to apologize here. I feel like I am totally hi-jacking this thread, but I have a similar project (conceptually similar, not in terms of quality of components however) going on and I'm at the point where I need to decide what kind of active crossover I'm going to use.
Usually I'm pretty quick with these things, but there are a lot of concepts being thrown around that are new to me. I looked up the price on the Prodigy 7.1 (and the Prodigy 192, which looks a little better) and it is totally reasonable. I want to use it with CONSOLE and the VST plugins you guys have mentioned here (which I already have) to take a stereo input, process it, and output the 6 channels to my 3-way speakers. I'd also like to be able to eliminate my other sound card by being able to output stereo audio as well as AC3 audio from the the same sound card that does the processing. I suppose this would mean using the internal router to route WAV or Directsound or whatever to ASIO and having it processed with the stereo input. I believe that in order to do this, since there may potentially be input audio at the same time that I'm using my WAV/DS output the card needs to be multiclient (is this correct?). I am assuming outputing AC3 audio from this card will be as easy as telling Theatertek to use the digital output from whichever card I choose. I'm leaning heavily toward the Prodigy 192, but also wanted to mention that for around the same price as the M-Audio Delta 1010LT (which doesn't look like it supports internal routing, although I'm almost positive now that it is multiclient) you can get the Maya 1010.
I know I may have turned into a real PITA here, but I'm really making progress with my design concept now. Can someone confirm that what I want to accomplish is possible with either the Prodigy 192 (if not, then the 7.1?) or the Maya 1010? |
Not thread hijacking at all m8. Way I see it is we're all here to share, so it doesn't bother me where that happens.
First off, good choice of soundcard, I really was trying to steer you in that direction but was unsure if you were a Delta owner since you mention it quite often ;)
Could I suggest that you go with the Prodigy 7.1 rather than the Prodigy 192. The reason is that the 7.1 uses good quality Wolfson DAC's and the 192 uses inferior parts, Strange but true, see the reviews here:
Prodigy 7.1:
http://www.digit-life.com/articles2...trak-prodigy71/
Prodigy 192:
http://www.digit-life.com/articles2...prodigy192.html
You can also bin your other sound card and exclusively use the Prodigy 7.1 for AC3 with TT and stereo with the external input. All that's required is you to route the signal's to the correct destinations for Console and its plugins to work its magic.
You'll also be able to create a 2.1 mix from the stereo input using the Waves Linear EQ and then route this to a sub.
It all looks a little intimidating at first (I thought so too) but actually its incredibly simple, easy to use and most of powerful stuff. I doubt you'll quiet believe the quality when you correctly setup the DRC stuff. It makes a considerable difference to the quality and the best words I could use to describe it would be that it makes the sound much more intelligible, realistic and cohesive.
Passive filters are quite laughable compared to the power and quality of the PC XO.
Also bear in mind that should you like what you hear and can live with the PC as an XO. You should really consider a dedicated box with a good pro soundcard and external DAC. Personally I'll be getting an Antelope Isochrone OCX clock when I've saved up the cash since building these has cleared my hifi budget out.
Certainly if your more technically adept then you could modify your Prodigy 7.1 as per Vil's instructions and have something that would sound close to the Isochrone for a fraction of the cost.
So to sum up, all that you have set out in your requirements can be accomplished with the setup outlined above and also go with the Prodigy 7.1 if possible - its a better sounding card thanks to the Wolfson DAC's. |
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| Vil |
| onboard ADC/DAC on prodigy 7.1 or 192 sounds baaad . Don't use them in that way . if you wanna take signal from external analog source use sound card with better quality ADC converter , something better - RME , Echo Gina 3G , even delta 1010 will work better here . |
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