m0cea said:
Cheers m8 😉
I did go through all the effort of spraying them black but the outcome was dissappointing. All sorts of artifacts from using around 3 cans of spray that weren't colour matched leading to very slightly different shades when rubbing flat. Also a hell of a lot of work using cans to get a high gloss finish. You could very easily do a 2 sats using a compressor and gun in the tme it takes to do one with spray cans. Lots and lots of sanding with cans. Less so with the gun.
The worst bit is the 2 week wait for the paint to fully cure and harden before the rubbing flat. The impatient one inside me wants to see the finish really mirror like but waiting for the paint to harden pays dividends in the way of MDF joint creep showing through the paint. Do it too soon and the joint will be visable in the right angles/light. Let it harden and the paint doiesn't shift around because its soft. 10 coats of paint also help 😉 😀
Hello-
I just say you guys talking about rme cards and outputs- I have a recording studio and own an 9652 and have owned RME cards for years.
The ADAT output is an 8 chanel digital out on a toslink optical connector. It is limited to 4 channels at 96k-
Generally the external ADC's and DAC's are better than the RME in the box DAC's. Check out the ADI 8 ds- this is RME's top conveter- But is far from being THE top converter (check out meitner, dcs, mytek, apogee). SOTA for DAC's is around -120dB ~12 dB better then the RME stuff you have now.
Reguardless, for digital audio the 1st thing to know is that the converters specs are a distant 2nd to the accuracy of the clock source. If you have the $$ this unit-
http://www.antelopeaudio.com/products_iso_ocx.html
or this-
http://www.apogeedigital.com/products/bigben.php
will offer a phenominal improvment to your system. Especially with a digital XO.
I recently bought an apogee AD16x wich shares the same clock as the big ben. Switching to this unit as clock master from the RME card was a phenominal difference. Even with so-so motu DAC's.
Also I think that the best digital crossover is with a DAW- you probably have a version of Cubase LE that came with the RME card. This (Cubase)- and decent DAC's and decent clock, will easily outperform a DEQ let alone a DCX IMHO.
There are any number of VST plugins that you can use to make your XO- if you want FIR check out the waves linear phase EQ package www.waves.com - on a good computer these will run with less than 100ms total latency. (You would need video compensation but for audio only it works great).
Also for the ultimate room correction check out vexengo curve EQ- It is a FIR eq that will do freq matching. So analyze pink noise- then play the pink noise for a measurement mic, and match the measured response to the pink noise with 60 bands of FIR filters (and it does it all for you!!). This offers in an incredible improvement every time I've tried it- The problem with the DEQ and the DCX is the limited options (only 1 EQ algo) and the lack of proper clocking- After years of recording digital audio I absolutly refuse to use anything digital that does not have a Wordclock (BNC) input.
Anyway you already have alot of what you need to check it out so go for it!! I compare DSP standard algos to the EQ built into cubase LE- EG I wager that even at 44.1k with the marginal built in EQ you can outperform a DCX, purly do to the imporved clocking.
Let me know if you need some help with cubase- I could send you an example project-
RC
I just say you guys talking about rme cards and outputs- I have a recording studio and own an 9652 and have owned RME cards for years.
The ADAT output is an 8 chanel digital out on a toslink optical connector. It is limited to 4 channels at 96k-
Generally the external ADC's and DAC's are better than the RME in the box DAC's. Check out the ADI 8 ds- this is RME's top conveter- But is far from being THE top converter (check out meitner, dcs, mytek, apogee). SOTA for DAC's is around -120dB ~12 dB better then the RME stuff you have now.
Reguardless, for digital audio the 1st thing to know is that the converters specs are a distant 2nd to the accuracy of the clock source. If you have the $$ this unit-
http://www.antelopeaudio.com/products_iso_ocx.html
or this-
http://www.apogeedigital.com/products/bigben.php
will offer a phenominal improvment to your system. Especially with a digital XO.
I recently bought an apogee AD16x wich shares the same clock as the big ben. Switching to this unit as clock master from the RME card was a phenominal difference. Even with so-so motu DAC's.
Also I think that the best digital crossover is with a DAW- you probably have a version of Cubase LE that came with the RME card. This (Cubase)- and decent DAC's and decent clock, will easily outperform a DEQ let alone a DCX IMHO.
There are any number of VST plugins that you can use to make your XO- if you want FIR check out the waves linear phase EQ package www.waves.com - on a good computer these will run with less than 100ms total latency. (You would need video compensation but for audio only it works great).
Also for the ultimate room correction check out vexengo curve EQ- It is a FIR eq that will do freq matching. So analyze pink noise- then play the pink noise for a measurement mic, and match the measured response to the pink noise with 60 bands of FIR filters (and it does it all for you!!). This offers in an incredible improvement every time I've tried it- The problem with the DEQ and the DCX is the limited options (only 1 EQ algo) and the lack of proper clocking- After years of recording digital audio I absolutly refuse to use anything digital that does not have a Wordclock (BNC) input.
Anyway you already have alot of what you need to check it out so go for it!! I compare DSP standard algos to the EQ built into cubase LE- EG I wager that even at 44.1k with the marginal built in EQ you can outperform a DCX, purly do to the imporved clocking.
Let me know if you need some help with cubase- I could send you an example project-
RC
Thanks Ryan,
I don't have the 9652 but the more analogue orientated 9632, it doesn't have the TOSlink opticals as standard. It does have the ADAT and you can add in a wordclock module for external clocks.
Could you help me out with this some more? I'd like to give it ago.
I have Cubase but not the LE version?
Also I've looked at the waves website but can't find the linear phase filter plug-in.
The vexengo curve EQ for DRC is also illusive. Could you provide a line pls?
Will I be able to use the PC based XO that you describe for theatertek? I believe it uses wave out on RME rather than ASIO, not sure though.
Cheers,
Ant
I don't have the 9652 but the more analogue orientated 9632, it doesn't have the TOSlink opticals as standard. It does have the ADAT and you can add in a wordclock module for external clocks.
Could you help me out with this some more? I'd like to give it ago.
I have Cubase but not the LE version?
Also I've looked at the waves website but can't find the linear phase filter plug-in.
The vexengo curve EQ for DRC is also illusive. Could you provide a line pls?
Will I be able to use the PC based XO that you describe for theatertek? I believe it uses wave out on RME rather than ASIO, not sure though.
Cheers,
Ant
Ouch!
Just found the Linear Phase EQ from Waves - $900!!!
No way I'm spending that on a flight of fancy.
I was hoping to try this out too but not with that price tag,
EDIT: Found it much cheaper from edonkey 😉
Just found the Linear Phase EQ from Waves - $900!!!
No way I'm spending that on a flight of fancy.
I was hoping to try this out too but not with that price tag,
EDIT: Found it much cheaper from edonkey 😉
Cool-
What version of cubase?- SX 2 should be fine- You will want at least that though for the automatic delay compensation.
Adat is toslink, toslink can be a # of things, it can be 2 channel optical spdif, 8 ch adat or 64 ch madi. Your card can do opti spdif or adat but not madi. This is where the "extra" channels are.
but that doesn't matter- a pretty basic PC can handle ALOT of audio I/O's if you already have 6 outs you are ready to go for a 3 way. Your card can handle plenty and if you want to do a multichanel active there are a # of solutions at different prices.
Anyway where is the music that you will be listening to coming from? EG do you have a CD player or are you ripping wav files etc? Also full 3 way active??- Do you plan to do 44.1k or higher? If you use a ripper like Nero (with jitter correction) at low speeds and then play back from the computer you will easily outperform any stand alone CD player, especially if you upsample it to 88.2 . Why? the clock- and improved jitter correction (redbook is aweful, cd jitter correction is FAR below that of NTFS or MACFS). And cd's are far worse with any file system then hard drives.
How fast is your computer? Just a bit of advice- a fast machine with no junk on it will run a quality XO W/O glitching but if you are sufing the web, playing games etc etc then it can glitch- and if you have no other protection it will damage your tweeter and possibly mid.
I would definitly recomend at least getting a 6 channel volume control as this way it will be off when you are not using it. You may also consider adding a couple caps to the domes for now- just put them on the edge of your XO HP freq- Once you get bugs worked out you can take them off, but you won't be needing to repair anything. I would recomend this for the DCX too.
This is the bleeding edge here- but if you are like me you will laugh everytime you hear sombody say "FIR filters are not audibly better". Trust me it's worth it and I have been gathering evedence to do a post on this subject. You will never hear transients like an FIR XO from any IIR setup.
Anyway a computer can be as stable as a dedicated device if you take a computer, and dedicate it! That's my feeling-
also i mistyped voxengo- It is with an o -
http://www.voxengo.com/
This one is cheaper- but do what you will. you will also want to get their free sample delay plugin- You can use this to time allign the drivers on a sample per sample basis. The CurveEQ is the great room correction one- it does not do HP LP filters so nogo on using it for the XO- but the room tuning will take your system up another notch.
If you give me your email address I will send you my XO project and it will make setup/walkthrough a bit easier. If you are playing the music from a different program let me know too- You will need to route it back in usuing totalmix on the 2.9 drivers. Also are you mac or PC?
You will also need to be careful with totalmix- You can route anything to anywhere- so you can accedentally route a full range signal to your tweets!
After trying this with IIR filters I immediatly sold my DCX, with FIR and the FIR room correction it will pass a near perfect 50hz square wave- especially with higher sample rates- but wait 'till you hear it!
RC
OH BTW the $900 version of the linear EQ is for TDM- protools, you would need another 10k worth of hardware to run it. The masters bundle is about $650 for native (ie cubase) or you might be able to pick it up, or just the EQ on ebay for less.
What version of cubase?- SX 2 should be fine- You will want at least that though for the automatic delay compensation.
Adat is toslink, toslink can be a # of things, it can be 2 channel optical spdif, 8 ch adat or 64 ch madi. Your card can do opti spdif or adat but not madi. This is where the "extra" channels are.
but that doesn't matter- a pretty basic PC can handle ALOT of audio I/O's if you already have 6 outs you are ready to go for a 3 way. Your card can handle plenty and if you want to do a multichanel active there are a # of solutions at different prices.
Anyway where is the music that you will be listening to coming from? EG do you have a CD player or are you ripping wav files etc? Also full 3 way active??- Do you plan to do 44.1k or higher? If you use a ripper like Nero (with jitter correction) at low speeds and then play back from the computer you will easily outperform any stand alone CD player, especially if you upsample it to 88.2 . Why? the clock- and improved jitter correction (redbook is aweful, cd jitter correction is FAR below that of NTFS or MACFS). And cd's are far worse with any file system then hard drives.
How fast is your computer? Just a bit of advice- a fast machine with no junk on it will run a quality XO W/O glitching but if you are sufing the web, playing games etc etc then it can glitch- and if you have no other protection it will damage your tweeter and possibly mid.
I would definitly recomend at least getting a 6 channel volume control as this way it will be off when you are not using it. You may also consider adding a couple caps to the domes for now- just put them on the edge of your XO HP freq- Once you get bugs worked out you can take them off, but you won't be needing to repair anything. I would recomend this for the DCX too.
This is the bleeding edge here- but if you are like me you will laugh everytime you hear sombody say "FIR filters are not audibly better". Trust me it's worth it and I have been gathering evedence to do a post on this subject. You will never hear transients like an FIR XO from any IIR setup.
Anyway a computer can be as stable as a dedicated device if you take a computer, and dedicate it! That's my feeling-
also i mistyped voxengo- It is with an o -
http://www.voxengo.com/
This one is cheaper- but do what you will. you will also want to get their free sample delay plugin- You can use this to time allign the drivers on a sample per sample basis. The CurveEQ is the great room correction one- it does not do HP LP filters so nogo on using it for the XO- but the room tuning will take your system up another notch.
If you give me your email address I will send you my XO project and it will make setup/walkthrough a bit easier. If you are playing the music from a different program let me know too- You will need to route it back in usuing totalmix on the 2.9 drivers. Also are you mac or PC?
You will also need to be careful with totalmix- You can route anything to anywhere- so you can accedentally route a full range signal to your tweets!
After trying this with IIR filters I immediatly sold my DCX, with FIR and the FIR room correction it will pass a near perfect 50hz square wave- especially with higher sample rates- but wait 'till you hear it!
RC
OH BTW the $900 version of the linear EQ is for TDM- protools, you would need another 10k worth of hardware to run it. The masters bundle is about $650 for native (ie cubase) or you might be able to pick it up, or just the EQ on ebay for less.
Oh and to answer your questions-
Use with theatertek- yes. However the XO will incur latency (part do to IO buffering, most do to the nature of FIR filters) you can determine the latency and if the DVD player allows you to delay the video you can line it back up. I think the waves lineq is about 84ms so add that to your buffer x2 and it will sync up. You will need a monster machine for all that though. Try it out (with caps on those domes please!!) and if you like it, and get it to work, you might want to sell the DCX and build a 2nd machine.
With IIR filters you can get a buffer low enough and CPU usage low enough to be sub 20ms so the latency SHOULD not be noticable. I have one that gets me down to 5ms I to O- this is suitable for realtime stuff (eg playing an instrument through the system). Then I switch back to Lineq for mixing and mastering.
This is where you will need to route the O's to the I's- totalmix can do this for you with the 2.9 driver and related firmware update. RME cards are multiclient- so cubase will use asio, and theatertek will use MME simultainiously. Just make sure that the DVD player is routed to outputs that do not go to your speakers. Then you can route it back in to cubase for the XO.
RC
Use with theatertek- yes. However the XO will incur latency (part do to IO buffering, most do to the nature of FIR filters) you can determine the latency and if the DVD player allows you to delay the video you can line it back up. I think the waves lineq is about 84ms so add that to your buffer x2 and it will sync up. You will need a monster machine for all that though. Try it out (with caps on those domes please!!) and if you like it, and get it to work, you might want to sell the DCX and build a 2nd machine.
With IIR filters you can get a buffer low enough and CPU usage low enough to be sub 20ms so the latency SHOULD not be noticable. I have one that gets me down to 5ms I to O- this is suitable for realtime stuff (eg playing an instrument through the system). Then I switch back to Lineq for mixing and mastering.
This is where you will need to route the O's to the I's- totalmix can do this for you with the 2.9 driver and related firmware update. RME cards are multiclient- so cubase will use asio, and theatertek will use MME simultainiously. Just make sure that the DVD player is routed to outputs that do not go to your speakers. Then you can route it back in to cubase for the XO.
RC
Wow, thanks Ryan.
Lots of info there.
My email is zeroex_15@hotmail.com if you could send me the files you were talking about that would be great!
The Behringer DCX arrived today but I'm going to put off setting it up in favour of this method. Then I can do a comparison once I'm happy with the PC based XO.
My PC specs are AMD Athlon 64 4000+, 1Gb RAM, 2 x 74Gb 10k RPM Raptor Hard drives & 1 x 200Gb backup drive. I use a fair amount of video post processing in Theatertek so I may have to back that down when using FIR filters
EDIT: MY Cubase is the SX3.0 version, any good?
Also how did you find out about this stuff???
Lots of info there.
My email is zeroex_15@hotmail.com if you could send me the files you were talking about that would be great!
The Behringer DCX arrived today but I'm going to put off setting it up in favour of this method. Then I can do a comparison once I'm happy with the PC based XO.
My PC specs are AMD Athlon 64 4000+, 1Gb RAM, 2 x 74Gb 10k RPM Raptor Hard drives & 1 x 200Gb backup drive. I use a fair amount of video post processing in Theatertek so I may have to back that down when using FIR filters
EDIT: MY Cubase is the SX3.0 version, any good?
Also how did you find out about this stuff???
RyanC said:
Yup.. I think its funny that RME specs their best jitter results at <1 nanosecond. Hmm, lets think about that: how many picoseconds are in a nanosecond? 1000! The best overall levels recorded in Stereophile is less than 200 pico seconds (not just the clock itself). The better readily available clocks (though freq. discreet) are about 2 pico seconds. DIY member Guido Tent sells them here (under the components page):
http://www.tentlabs.com/
Also Ryan, do you have an MSN Messenger account that I could chat to you about this?
The more I look into the software you've mentioned the more exciting this becomes. I can see many possibilities and endless improve I could make.
If I'm happy with the results I'd like to build a dedicated machine just to run cubase and the plugins. Then buy a second RME to use in my main machine and connect them via the wordclock to sync the digital. Is this doable?
What minimum specs would you recommend for the second machine? Obviously it wouldn't need the fancy graphics card but what about memory and processing. I plan to run the stereo 3-way FIR filters along with the FIR room correction and that's all on the machine, no other junk just a streamlined XO box.
The more I look into the software you've mentioned the more exciting this becomes. I can see many possibilities and endless improve I could make.
If I'm happy with the results I'd like to build a dedicated machine just to run cubase and the plugins. Then buy a second RME to use in my main machine and connect them via the wordclock to sync the digital. Is this doable?
What minimum specs would you recommend for the second machine? Obviously it wouldn't need the fancy graphics card but what about memory and processing. I plan to run the stereo 3-way FIR filters along with the FIR room correction and that's all on the machine, no other junk just a streamlined XO box.
Hehe-
I have messanger turned off because this machine runs my xo and it gives me probs-
Anyway yes- every new plugin along these lines opens up the floodgates even more.
"If I'm happy with the results I'd like to build a dedicated machine just to run cubase and the plugins. Then buy a second RME to use in my main machine and connect them via the wordclock to sync the digital. Is this doable?"
thats what I'm doin- this guy is actually only a 1.2ghz athlon with 512mB of ram- The faster the machine- the lower you can run your buffer. But with FIR filters there will always be a delay (AFAIK) so i would suggest a 1.5GHZ or better. With things as cheap as they are now you might as well get a faster one ~3ghz. I would recomend P4 because the PCI bus tends to be less problematic if you are getting up to higher bandwidths (EG surround sound system) Asus is well trusted in the studio world. Also consider that each doubling of Sf doubles the CPU usage- so as of now doing this at 192K would require a screamin machine like my main daw (dual 3.6GHZ xeons 2GB ram). Or those new waves hardware boxes!!!
Again I cannot stress the importance of a good clock enough- if you are looking to do single sample rates on a budget look at the aardsync II on ebay- for $300 this will typically offer a improvement over the RME's- I think poor clocking has led many to throw the baby out with the bathwater when it comes to FIR XO's.
Also you will not need another RME card for the main PC- just a card with digital outs. Also it is well known/documented that EQ's compute more accuratly at higher sample rates- and therfor sound better= better passage of square wave= more accurate transient response also antialias filtering inhibits passage of a perfect square so the higher it is the better. How much I don't know- my system will not run at 88.2 or 96 with the Fir filters (yet!!!). And I don't have the $$ for a good SRC, and I don't like resolving double Sf to a single WC or the other way around.
I am in the process of preparing a post where I intend to argue that FIR XO's are better and that you can hear and measure the difference easily. I'm almost done and I will have a more extensive "how to" assciated with it. In the meantime give me your email and i will send you my XO project- I think that you will be able to reverse engineer my XO to suit your purposes pretty easily. It is easy to get your head around the routing when you look at how it is setup (cubase has to be a bit jeri rigged to do XO's). One thing to be aware of- the Lineq has some limitations to how it can be setup- you cannot do a high q, hi pass below 254hz. You can do a lower q one all the way down- I imagine this would not be a problem for you though as the ATC should be 350 or above right?
Anyway my email is ryan2244 @ comcast . net (without the spaces). My project is a NPR (nuendo) but cubase should open it just fine.
RC
I have messanger turned off because this machine runs my xo and it gives me probs-
Anyway yes- every new plugin along these lines opens up the floodgates even more.
"If I'm happy with the results I'd like to build a dedicated machine just to run cubase and the plugins. Then buy a second RME to use in my main machine and connect them via the wordclock to sync the digital. Is this doable?"
thats what I'm doin- this guy is actually only a 1.2ghz athlon with 512mB of ram- The faster the machine- the lower you can run your buffer. But with FIR filters there will always be a delay (AFAIK) so i would suggest a 1.5GHZ or better. With things as cheap as they are now you might as well get a faster one ~3ghz. I would recomend P4 because the PCI bus tends to be less problematic if you are getting up to higher bandwidths (EG surround sound system) Asus is well trusted in the studio world. Also consider that each doubling of Sf doubles the CPU usage- so as of now doing this at 192K would require a screamin machine like my main daw (dual 3.6GHZ xeons 2GB ram). Or those new waves hardware boxes!!!
Again I cannot stress the importance of a good clock enough- if you are looking to do single sample rates on a budget look at the aardsync II on ebay- for $300 this will typically offer a improvement over the RME's- I think poor clocking has led many to throw the baby out with the bathwater when it comes to FIR XO's.
Also you will not need another RME card for the main PC- just a card with digital outs. Also it is well known/documented that EQ's compute more accuratly at higher sample rates- and therfor sound better= better passage of square wave= more accurate transient response also antialias filtering inhibits passage of a perfect square so the higher it is the better. How much I don't know- my system will not run at 88.2 or 96 with the Fir filters (yet!!!). And I don't have the $$ for a good SRC, and I don't like resolving double Sf to a single WC or the other way around.
I am in the process of preparing a post where I intend to argue that FIR XO's are better and that you can hear and measure the difference easily. I'm almost done and I will have a more extensive "how to" assciated with it. In the meantime give me your email and i will send you my XO project- I think that you will be able to reverse engineer my XO to suit your purposes pretty easily. It is easy to get your head around the routing when you look at how it is setup (cubase has to be a bit jeri rigged to do XO's). One thing to be aware of- the Lineq has some limitations to how it can be setup- you cannot do a high q, hi pass below 254hz. You can do a lower q one all the way down- I imagine this would not be a problem for you though as the ATC should be 350 or above right?
Anyway my email is ryan2244 @ comcast . net (without the spaces). My project is a NPR (nuendo) but cubase should open it just fine.
RC
voxengo sample delay works good it's per sample so however you are measuring it you have to work 1/44100th of a second. Also sx2 and above auto corrects for the latency of the plugin itself (or it is supposed to at least)- so you will only need this for drivers.
very interesting ! tipically I am too lazy for any comments but last posts are very interesting so I will add my 2c .
I am using computer for 3way xover . after a lot experiments with different pro audio cards (creamware scope, rme , echo etc) i found jitter level way too big if using adat or s/pdif (master and slave configurations) so i did it in another way . I found right cheap sound card , Audiotrack Prodigy 7.1 and took 24bit I2S digital data directly from envy chip . also i did some test and found those 24bits are true , without any recalculations . latter i used LVDS transmiters to send digital and clock data to converter units , they are located insight left and right speakers with 3 way amplifiers . additional low jitter clock generator is located near sound card and clocks dac units+ sound card and reclocks I2S data (just in case) .I am using Audio Console software with ASIO2 and waves , voxengo etc. stuff for procesing.
I am using computer for 3way xover . after a lot experiments with different pro audio cards (creamware scope, rme , echo etc) i found jitter level way too big if using adat or s/pdif (master and slave configurations) so i did it in another way . I found right cheap sound card , Audiotrack Prodigy 7.1 and took 24bit I2S digital data directly from envy chip . also i did some test and found those 24bits are true , without any recalculations . latter i used LVDS transmiters to send digital and clock data to converter units , they are located insight left and right speakers with 3 way amplifiers . additional low jitter clock generator is located near sound card and clocks dac units+ sound card and reclocks I2S data (just in case) .I am using Audio Console software with ASIO2 and waves , voxengo etc. stuff for procesing.
Attachments
Cool vil-
Yea- in my experience you should never derive clock data from a digital audio stream It should always have a dedicated line- especially optical. That is what asio 2 affords us aswell (inside the computer).
That is what wordclock is for in pro setups. But it sounds like you have a well designed system there. One question- you said there is clocking on both ends? My understanding is that excesive PLL and resoloution to different clocks is also a compromise. Any thoughts? That's cool that you use JP console. I looked at that app too.
Also Vil what do you think about FIR vs IIR XO's I still feel that the transients are different and better with FIR and I thought I had it measured but I need to experament with it more-
Also shinobiwan- art teknika console (the program that vil is using here) is very easy to configure. It would be much simpliler than cubase without any negative side effects- you can see his XO is bassically identical to mine- 3 way lineq XO and voxengo sample delay for tuning. Then vil is using L2 ( a peak limiter) to protect from output clipping- One benifit of FIR filters I find is less peaky output so i avoid the limiters myself.
VIL- try the voxengo curveeq for room tuning- especially if you have a measurement mic- It is very good for this!! I do stereo and wave the mic around the listening position. Thanks -
Ryan
Yea- in my experience you should never derive clock data from a digital audio stream It should always have a dedicated line- especially optical. That is what asio 2 affords us aswell (inside the computer).
That is what wordclock is for in pro setups. But it sounds like you have a well designed system there. One question- you said there is clocking on both ends? My understanding is that excesive PLL and resoloution to different clocks is also a compromise. Any thoughts? That's cool that you use JP console. I looked at that app too.
Also Vil what do you think about FIR vs IIR XO's I still feel that the transients are different and better with FIR and I thought I had it measured but I need to experament with it more-
Also shinobiwan- art teknika console (the program that vil is using here) is very easy to configure. It would be much simpliler than cubase without any negative side effects- you can see his XO is bassically identical to mine- 3 way lineq XO and voxengo sample delay for tuning. Then vil is using L2 ( a peak limiter) to protect from output clipping- One benifit of FIR filters I find is less peaky output so i avoid the limiters myself.
VIL- try the voxengo curveeq for room tuning- especially if you have a measurement mic- It is very good for this!! I do stereo and wave the mic around the listening position. Thanks -
Ryan
OH-
You are using curve eq- funny how two people end up at the exact same end point!! Cool- we MUST be onto somthing
RC
You are using curve eq- funny how two people end up at the exact same end point!! Cool- we MUST be onto somthing
RC
wordclock is OK but not the best option !!! system needs 256F or even more and wordclock is just 1F so PLL is used to get 256F . just superclock is the right ONE . I did in another way (look to the attachment)
by the way LVDS is very cool , i think thats is best low cost solution for clock and data transmission . I am using separate cat6 cable(one pair of 4) for clock transmision and another for I2S data .
by the way LVDS is very cool , i think thats is best low cost solution for clock and data transmission . I am using separate cat6 cable(one pair of 4) for clock transmision and another for I2S data .
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Ryan - my work is still in progress , no final opinion abuot FIR and IIR , i need better speakers , so I am working on that . Just got Hyquphoon OWII tweeter and ATC 150S mid unit , waiting for Lambda15S woofer . I will go for sealed box and will use Voxengo EQ as Linkwitz transform .
with my current speakers(Tannoy) FIR sounds better .
Also I must say I am working on totally quiet PC using pentium M cpu with fanless cooler and fanless power suplly .... HDD wil be damped too .
I am using Monkey's audio ape format audio data storage and I am very happy with it .Absolutely same quality compared to wav.
Also DAC's (just PCB's withuot cases) are very very good ones from LessLoss > www.lessloss.com
with my current speakers(Tannoy) FIR sounds better .
Also I must say I am working on totally quiet PC using pentium M cpu with fanless cooler and fanless power suplly .... HDD wil be damped too .
I am using Monkey's audio ape format audio data storage and I am very happy with it .Absolutely same quality compared to wav.
Also DAC's (just PCB's withuot cases) are very very good ones from LessLoss > www.lessloss.com
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