Hi OzOnE,
I cannot help with DRC as I have not tried a lot.
Of course I can tell a lot with Acourate. But maybe the best is you send me pulse responses by private mail. Then I can prep correction filters for you and you can tell if you like it.
Uli
www.acourate.de
I cannot help with DRC as I have not tried a lot.
Of course I can tell a lot with Acourate. But maybe the best is you send me pulse responses by private mail. Then I can prep correction filters for you and you can tell if you like it.
Uli
www.acourate.de
Hi,
My setup consistently inverts the impulse response, I simply live with it, no problem at all.
I'm sorry I can't help with your other questions (no clue) but just as a suggestion, try the minimum-phase versions of the the DRC filters to discard any phase artifacts (DRC trying to correct some weird situation). In my setuo min-phase filters works fine, as the loudspeakers are "phase-perfect" from the beggining, and bass correction can be made min-phase to any practical purpose, IMHO.
And you can reach Denis Sbragion at the DRC mailing list. He is the real expert and is eager to help. You wont regret to ask him.
Cheers,
R
Hi Uli / RR,
Thanks for the offer Uli, but I think I'm getting closer to finding out what the problems are with my impulses. I also don't think any of my recorded sweeps are any good....
I tried measuring twice with exact settings and levels, but with one measurement going through ASIO / Console, and one going directly out of WDM. Both resulting DRC filters (generated from these different sweeps) sounded exactly the same, so it seems perfectly fine to loop the WDM sound through Console when recording the sweeps - even through the LINEQ2 filters.
(In fact, the sub bass sounds slightly better when measuring / running via Console and with the LINEQ2 filters acting as crossovers.)
The only filter I have which produces a very similar sound to my best one is named "left44k boosted preamp through LINEQ2", so it appears the problem of the hollow / harsh sounding filters is probably down to low SNR of the log sweep recordings....
I think the "boosted" filter was generated from a sweep using the modified mic, and a preamp gain of around 42dB!! I think the next thing I should try is boosting the preamp gain again?
I'm using the unmodified WM-61A capsule now with a non-boosted preamp, and I'm starting to get better results.
I think that when I first did the log sweep recordings, I used a fairly high volume output from the speakers to make sure I got a good recording. I'm worried about doing this again, because I don't want to kill my speakers. I also don't have an SPL meter yet, so I can't be sure of the proper levels?
When recording the sweeps, I generally use a level of -30dB on a Denon AVC-A11SR with M&K LCR-55 speakers - Maybe I should just buy an SPL meter instead of trying to work this stuff out!!
The other reason for trying the unmodded capsule now is because I'm still unsure about which polarity the capsule is after modification? I know that the diagram on the Linkwitz site uses a 9V battery in reverse to give a "negative" power feed to the modded capsule, but I don't think I can easily acheive that with my preamp without adding a second battery?
Did anyone find out what the definitive answer was on how to connect a modified capsule to a standard preamp? ie. is it simply a matter of connecting the metal case (pin 2/3) to "ground" on the preamp, then pin 1 is the signal, or should it be reversed? It would be preferrable to record a signal with the correct phase.
I know a little bit about electronics, but this DRC thing is really taxing my feeble brain.
OzOnE.
P.S. Actually, tell a lie, I think the preamp gain is about 48dB at the moment!!! (even with an unmodded capsule connected). I'm sure most of you will tell me that 48dB is too much gain.
I'm using the following preamp circuit with half of an OPA2604 opamp btw.....
http://www.zen22142.zen.co.uk/Circuits/Audio/lf071_mic.htm
...with R2 as 47k, and R1 as 1k = 48dB gain??
Plus, a 10k resistor is added between the mic capsule and +9V to supply power to the mic. I don't think the preamp is clipping, but surely this is way too much gain for the purposes of DRC?
The preamp is connected near the mic, and the output of the preamp goes via a 5 meter long cable directly into the line-in socket on a Prodigy 7.1 sound card?
Thanks for the offer Uli, but I think I'm getting closer to finding out what the problems are with my impulses. I also don't think any of my recorded sweeps are any good....
I tried measuring twice with exact settings and levels, but with one measurement going through ASIO / Console, and one going directly out of WDM. Both resulting DRC filters (generated from these different sweeps) sounded exactly the same, so it seems perfectly fine to loop the WDM sound through Console when recording the sweeps - even through the LINEQ2 filters.
(In fact, the sub bass sounds slightly better when measuring / running via Console and with the LINEQ2 filters acting as crossovers.)
The only filter I have which produces a very similar sound to my best one is named "left44k boosted preamp through LINEQ2", so it appears the problem of the hollow / harsh sounding filters is probably down to low SNR of the log sweep recordings....
I think the "boosted" filter was generated from a sweep using the modified mic, and a preamp gain of around 42dB!! I think the next thing I should try is boosting the preamp gain again?
I'm using the unmodified WM-61A capsule now with a non-boosted preamp, and I'm starting to get better results.
I think that when I first did the log sweep recordings, I used a fairly high volume output from the speakers to make sure I got a good recording. I'm worried about doing this again, because I don't want to kill my speakers. I also don't have an SPL meter yet, so I can't be sure of the proper levels?
When recording the sweeps, I generally use a level of -30dB on a Denon AVC-A11SR with M&K LCR-55 speakers - Maybe I should just buy an SPL meter instead of trying to work this stuff out!!
The other reason for trying the unmodded capsule now is because I'm still unsure about which polarity the capsule is after modification? I know that the diagram on the Linkwitz site uses a 9V battery in reverse to give a "negative" power feed to the modded capsule, but I don't think I can easily acheive that with my preamp without adding a second battery?
Did anyone find out what the definitive answer was on how to connect a modified capsule to a standard preamp? ie. is it simply a matter of connecting the metal case (pin 2/3) to "ground" on the preamp, then pin 1 is the signal, or should it be reversed? It would be preferrable to record a signal with the correct phase.
I know a little bit about electronics, but this DRC thing is really taxing my feeble brain.
OzOnE.
P.S. Actually, tell a lie, I think the preamp gain is about 48dB at the moment!!! (even with an unmodded capsule connected). I'm sure most of you will tell me that 48dB is too much gain.
I'm using the following preamp circuit with half of an OPA2604 opamp btw.....
http://www.zen22142.zen.co.uk/Circuits/Audio/lf071_mic.htm
...with R2 as 47k, and R1 as 1k = 48dB gain??
Plus, a 10k resistor is added between the mic capsule and +9V to supply power to the mic. I don't think the preamp is clipping, but surely this is way too much gain for the purposes of DRC?
The preamp is connected near the mic, and the output of the preamp goes via a 5 meter long cable directly into the line-in socket on a Prodigy 7.1 sound card?
A thought about EQ / Filter plugins.
Having tried both Waves and Algorithmix Plugins i favour the Algorithmix. It has higher cpu consumption and latency but i think it sounds more neutral in the bass frequencies (especially when you boost them to counter dipole rolloff).
Perhaps its just an illusion. But as long as i cant proof it scientifically even a illusion seems real.
Having tried both Waves and Algorithmix Plugins i favour the Algorithmix. It has higher cpu consumption and latency but i think it sounds more neutral in the bass frequencies (especially when you boost them to counter dipole rolloff).
Perhaps its just an illusion. But as long as i cant proof it scientifically even a illusion seems real.
3way on Audiotrak Prodigy LT
Hi everyone, sorry to jump in like this, but I have some question;
I have try to read whole thread couple time but never really have time to read it from beginning to the end so...
so please, have that in mind if this question was answered before
a) is there some free ware to make stereo 3 way crossover ? ( I have Audiotrak prodigy LT )
b) what is the bast way to protect tweeter while playing with active crossover ? ( if I remember correctly ShinOBIWAN manage to burn one while playing with this )
thanks
Josip
Hi everyone, sorry to jump in like this, but I have some question;
I have try to read whole thread couple time but never really have time to read it from beginning to the end so...
so please, have that in mind if this question was answered before
a) is there some free ware to make stereo 3 way crossover ? ( I have Audiotrak prodigy LT )
b) what is the bast way to protect tweeter while playing with active crossover ? ( if I remember correctly ShinOBIWAN manage to burn one while playing with this )
thanks
Josip
Re: 3way on Audiotrak Prodigy LT
a) I think you can go with foobar and this http://xover.sourceforge.net/
b) put a cap before the tweeter to protect it, I always test with fullrange drivers though.
yoke said:
a) I think you can go with foobar and this http://xover.sourceforge.net/
b) put a cap before the tweeter to protect it, I always test with fullrange drivers though.
thuneau, (not sure if you're still reading this thread, but I'll address the question here since it's relevant)
It seems like your program is the only that supports 2 in 8 out from a 8channel output soundcard. Do you know if there's a way for me to do the same by using Console? (I already have Console since I first read this thread)
It seems like your program is the only that supports 2 in 8 out from a 8channel output soundcard. Do you know if there's a way for me to do the same by using Console? (I already have Console since I first read this thread)
ok ok ok.
i read MOST of this thead. let's get back to basics. the thread is called PC XO. i want to know how to do just the XO part. no EQ'ing, no fancy stuff.
i know about the foobar plugin, but i also need it to work with theatertek, and audio in (the computer serves as a processor, so it had audio coming into it).
i want a SIMPLE 2-way crossover. meaning, i have two channels of stereo. i want to split them into 4 channels, 2 having frequencies 60hz and above, the other 2 having frequencies 60hz and below. there would be a nice 12 or 24db/octave slope. of course, it would be nice if these things could be variable (6/12/24/48db/octave slopes, and adjustable frequency). also, i want to combine the two low-pass channels into a mono channel.
this sounds SO simple to me, after seeing what you guys are doing. but why is it so hard to figure out? i know about waves lineEQ, but it was mentioned it doenst necessarily do what i want it to do, and quite frankly, its VERY pricey and too complex for that. i already have a silent (no fans, no moving parts, AT ALL) HTPC, and dont want or need another one. it should be plenty to run a crossover and the programs too. what do i have to do to accomplish this?
oh, i use an m-audio delta 410 right now, but ordered a lynx 2B that will be here in a few days.
i read MOST of this thead. let's get back to basics. the thread is called PC XO. i want to know how to do just the XO part. no EQ'ing, no fancy stuff.
i know about the foobar plugin, but i also need it to work with theatertek, and audio in (the computer serves as a processor, so it had audio coming into it).
i want a SIMPLE 2-way crossover. meaning, i have two channels of stereo. i want to split them into 4 channels, 2 having frequencies 60hz and above, the other 2 having frequencies 60hz and below. there would be a nice 12 or 24db/octave slope. of course, it would be nice if these things could be variable (6/12/24/48db/octave slopes, and adjustable frequency). also, i want to combine the two low-pass channels into a mono channel.
this sounds SO simple to me, after seeing what you guys are doing. but why is it so hard to figure out? i know about waves lineEQ, but it was mentioned it doenst necessarily do what i want it to do, and quite frankly, its VERY pricey and too complex for that. i already have a silent (no fans, no moving parts, AT ALL) HTPC, and dont want or need another one. it should be plenty to run a crossover and the programs too. what do i have to do to accomplish this?
oh, i use an m-audio delta 410 right now, but ordered a lynx 2B that will be here in a few days.
im having a few problems getting console up and running. im trying to just play around with it a bit, but its not working.
i add in my m-audio as an ASIO device. i throw down the input and output. however, foobar and console cant both use the ASIO at the same time. what am i supposed to do? using directsound in console only give me 2 channels. how are you guys using an audio player (like foobar) to play something, and using console to do its thing?
also, when i make a connection with the linEQ filter, it always says "cannot execute in real time", whatever that means. nothing is really working at all...
i add in my m-audio as an ASIO device. i throw down the input and output. however, foobar and console cant both use the ASIO at the same time. what am i supposed to do? using directsound in console only give me 2 channels. how are you guys using an audio player (like foobar) to play something, and using console to do its thing?
also, when i make a connection with the linEQ filter, it always says "cannot execute in real time", whatever that means. nothing is really working at all...
Hi, cowanrg...
I can only speak for the Audiotrak Prodigy card, but with the Audiotrak drivers, the DirectWIRE feature allows me to loop the Directsound 'outputs' back into ASIO, which Console can then process. This way, everything which plays through Windows / DS goes through Console before it reaches the speakers.
I've attached a screenshot of the basic setup on my Audiotrak driver. I'm guessing an M-Audio or Lynx card must have a similar system for routing signals?
(The 'WDM' outputs are actually the Directsound outputs in the example below.)
In the screenshot, I also have the ASIO (speaker) outputs looped back to the channel 2 (right) WDM input because I was trying to get rec_imp.exe to work at the time (btw, it still doesn't work, but intend to mess with it again today). You can ignore the loop back signals for this example, but you can see what I mean by how the driver works.
This also works for multi-channel stuff like DVDs when using AC3Filter, or a software player like PowerDVD etc. The six (or seven / eight) channels are output from the WDM (DS) outputs, which get looped through Console so I can apply DRC or EQ to every channel simultaneously.
I've seen quite a few examples of people having a similar problem to yours (like this one)....
http://www.presonus.com/forums/showthread.php?t=925
....but unfortunately, I can't see many easy ways around it if your sound card drivers don't support internal routing between WDM and ASIO.
One way might be to use Console as a VST plugin directly inside Foobar2000. I think you can get a plugin for Foobar which should let you play music directly into Console's ASIO 'inputs'. I can't seem to find the plugin atm, but I know it's out there somewhere.
OzOnE.
P.S. @houstonian - I always check my recorded log sweeps with Cool Edit to make sure they're not clipping. I'm also fairly certain the mic preamp isn't clipping as I think the mic capsule will only clip the preamp with very high SPL levels. I think my problem is still down to having cheap 'n' nasty cables for the mic, and an old ball-point pen casing as a mic wand!
I can only speak for the Audiotrak Prodigy card, but with the Audiotrak drivers, the DirectWIRE feature allows me to loop the Directsound 'outputs' back into ASIO, which Console can then process. This way, everything which plays through Windows / DS goes through Console before it reaches the speakers.
I've attached a screenshot of the basic setup on my Audiotrak driver. I'm guessing an M-Audio or Lynx card must have a similar system for routing signals?
(The 'WDM' outputs are actually the Directsound outputs in the example below.)
In the screenshot, I also have the ASIO (speaker) outputs looped back to the channel 2 (right) WDM input because I was trying to get rec_imp.exe to work at the time (btw, it still doesn't work, but intend to mess with it again today). You can ignore the loop back signals for this example, but you can see what I mean by how the driver works.
This also works for multi-channel stuff like DVDs when using AC3Filter, or a software player like PowerDVD etc. The six (or seven / eight) channels are output from the WDM (DS) outputs, which get looped through Console so I can apply DRC or EQ to every channel simultaneously.
I've seen quite a few examples of people having a similar problem to yours (like this one)....
http://www.presonus.com/forums/showthread.php?t=925
....but unfortunately, I can't see many easy ways around it if your sound card drivers don't support internal routing between WDM and ASIO.
One way might be to use Console as a VST plugin directly inside Foobar2000. I think you can get a plugin for Foobar which should let you play music directly into Console's ASIO 'inputs'. I can't seem to find the plugin atm, but I know it's out there somewhere.
OzOnE.
P.S. @houstonian - I always check my recorded log sweeps with Cool Edit to make sure they're not clipping. I'm also fairly certain the mic preamp isn't clipping as I think the mic capsule will only clip the preamp with very high SPL levels. I think my problem is still down to having cheap 'n' nasty cables for the mic, and an old ball-point pen casing as a mic wand!
Attachments
Hi again,
I've just got Console working from within Foobar2000, but it still doesn't allow me to use all ASIO outputs, so I only get 2-channel output.
To get this to work, I had to install VST Host DSP for Winamp, then install the Winamp DSP Bridge plugin for foobar. This allows foobar to use Winamp's plugins, then the Winamp VST Host plugin allows Console to be loaded. phew!
Again, this obviously won't work for what you need atm, but I'm assuming there's a much simpler method for foobar, like this....
http://sourceforge.net/projects/xover/
You just need to get a routing plugin which allows you to route the xover outputs to each ASIO (speaker) output.
OzOnE.
I've just got Console working from within Foobar2000, but it still doesn't allow me to use all ASIO outputs, so I only get 2-channel output.
To get this to work, I had to install VST Host DSP for Winamp, then install the Winamp DSP Bridge plugin for foobar. This allows foobar to use Winamp's plugins, then the Winamp VST Host plugin allows Console to be loaded. phew!
Again, this obviously won't work for what you need atm, but I'm assuming there's a much simpler method for foobar, like this....
http://sourceforge.net/projects/xover/
You just need to get a routing plugin which allows you to route the xover outputs to each ASIO (speaker) output.
OzOnE.
OzOnE_2k3 said:
hum, thanks for the help, but i dont think that will work. im not trying to JUST get winamp to do this, im trying to get it to work for every audio stream that the soundcard sees, that's why im trying to use console. i know plugins exist for foobar that let me do this, but i need it to work with theatertek, and others.
it seems as though when foobar is running, it 'captures' the ASIO device and console cant recognize it's there. if i close foobar, console has access to the ASIO device, but foobar wont be able to load. both of them cant share.
cowanrg said:
The problem is two clients can't share the same ASIO channel. Install the waveout output plugin for winamp and use that. Then pick that up in Console. Theatertek uses waveout so that not a problem.
I'm not familiar with the routing on the Audiotrak cards but when I used RME and Lynx cards they had a feature called internal loopback which took the output from an ASIO or waveout playback channel and looped it back into whichever inputs you chose. I then grabbed the audio from these inputs in Console and this avoided the conflicts from having two programs sharing a playback channel.
If Console isn't working out for you then give Frequency Allocator from Thanau a try.
ok, that makes sense.
i did some more playing around, and im getting a bit frustrated. i was just using theatertek as a source, (since foobar is using ASIO right now). i setup the wave in and a wave out. i connected a linEQ in between then and just added a high-pass filter. i played a movie that i knew had low bass, hoping that the bass would be gone (just as a test). when the audio started, all i got was loud digital sounding reverb. it kept going even after i closed the movie player. the only way to stop it was to stop console.
maybe my soundcard just doesnt like console, but this isnt working at all for me.
i did some more playing around, and im getting a bit frustrated. i was just using theatertek as a source, (since foobar is using ASIO right now). i setup the wave in and a wave out. i connected a linEQ in between then and just added a high-pass filter. i played a movie that i knew had low bass, hoping that the bass would be gone (just as a test). when the audio started, all i got was loud digital sounding reverb. it kept going even after i closed the movie player. the only way to stop it was to stop console.
maybe my soundcard just doesnt like console, but this isnt working at all for me.
Two clients can't share the same ASIO channels. You must setup console to use different output channels that you use for foobar.
In console, you use the menu option edit/preferences. Click on "Audio Device" and then on "Channel". Then select the input and output channels you want to use with console.
In console, you use the menu option edit/preferences. Click on "Audio Device" and then on "Channel". Then select the input and output channels you want to use with console.
cowanrg said:
You've got playback channels conflicting, the reverb isn't reverb at all but rather the buffer filling up with garbage and getting stuck in a loop. This is a classic sign of conflicts, other are pops and stuttering.
It was a while ago now but I know a couple of people in this thread had console working with Audiotrak cards such as the Prodigy.
You need to take a close look at the channel routings in the Audiotrak drivers and make sure each channel has a clean path. This is the path that it took when I did this:
Go from playback -> loopback into an input -> grab the input in Console -> process in console and output -> pick up the playback channel from console output and route to a physical output
Its a pretty complicated routing scheme and one mistake will cause you issues.
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