Frequency Allocator Light?
Here is a pretty advanced software crossover.
4-way (or more if multiple instances are used) up to 42dB/oct slopes, response graphs, measurements importing, EQ, delay, etc- it's all there.
All you need is a PC with multichannel ASIO sound card and $59.00.
No excuses if you can't make your speakers sound right with this cossover.
Check it out at
http://www.thuneau.com/alloclite.htm ] http://www.thuneau.com/alloclite.htm [/URL]
Yes, it's my website, but there isn't really anything out there that gives you this much power when it comes to software DSP crossovers for so little money.
So, while it's admittedly a shameless plug, it's an honest advice at the same time.
I developed this software because there was nothing like it available and I wanted these features in a software crossover.
Here is a pretty advanced software crossover.
4-way (or more if multiple instances are used) up to 42dB/oct slopes, response graphs, measurements importing, EQ, delay, etc- it's all there.
All you need is a PC with multichannel ASIO sound card and $59.00.
No excuses if you can't make your speakers sound right with this cossover.
Check it out at
http://www.thuneau.com/alloclite.htm ] http://www.thuneau.com/alloclite.htm [/URL]
Yes, it's my website, but there isn't really anything out there that gives you this much power when it comes to software DSP crossovers for so little money.
So, while it's admittedly a shameless plug, it's an honest advice at the same time.
I developed this software because there was nothing like it available and I wanted these features in a software crossover.
I've been reading most of this thread now, very interesting. Thanks for all the usefull info, especially ShinOBIWAN.
I'm looking for a new soundcard and I stumbled upon the Terratec Phase series. Would this PC XO work with e.g. the Phase 28?
http://audioen.terratec.net/modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=25
Kind regards,
Emiel
I'm looking for a new soundcard and I stumbled upon the Terratec Phase series. Would this PC XO work with e.g. the Phase 28?
http://audioen.terratec.net/modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=25
Kind regards,
Emiel
I agree that the Thunau FreqAllocator should, perhaps, be the first starting place for those wanting to do crossover functions on a Windows PC. I look forward to refinements of the program. The one I'm most interested in is a way to swith the Allocator into a super low latency mode (eg. like Allocator lite) for home theatre/DVD use.
I am going to begn experimenting wih partial digital/passive crossover.
The only issue I see with using the Thunau stuff is you still need to complement it with measurement tools. I havent found a good system for measuring MLS response, etc.
In particular, I need a good 48volt mic preamp and A/D convertor with AES/EBU outputs.... anyone know of something? And I'm also finding that an uncalibrated ECM8000 Behringer Mic isnt enough for serious measurements.
I am going to begn experimenting wih partial digital/passive crossover.
The only issue I see with using the Thunau stuff is you still need to complement it with measurement tools. I havent found a good system for measuring MLS response, etc.
In particular, I need a good 48volt mic preamp and A/D convertor with AES/EBU outputs.... anyone know of something? And I'm also finding that an uncalibrated ECM8000 Behringer Mic isnt enough for serious measurements.
Yes, it looks like a very good candidate for the Allocator Light. It has ASIO drivers, 8 outputs and digital I/O.
I said Allocator Light because I could not find the ASIO buffers descriptions for it. If it has the option to set them to 1024 or more, you should be able to run the Allocator with no problems.
The only thing of possible concern to users might be the cabling.
The break out connector uses TRS (tip ring sleeve) 1/4" connectors. The signal is balanced and from what I can find on their website- set to 0dBV nominal output level. That means it's on the hot side. You would need extra 10 dB of attenuation to make it work with consumer style hi-fi amps that have RCA inputs.
The Terratec Phase 88 has a nice break out box with RCA outputs that can be set to output -10dB signal and you don't have to get behind the computer to make connections. That one looks like a very nice interface for use with crossover software. I know it's much more costly, but you get the convenience and features that make it very good for the application.
I said Allocator Light because I could not find the ASIO buffers descriptions for it. If it has the option to set them to 1024 or more, you should be able to run the Allocator with no problems.
The only thing of possible concern to users might be the cabling.
The break out connector uses TRS (tip ring sleeve) 1/4" connectors. The signal is balanced and from what I can find on their website- set to 0dBV nominal output level. That means it's on the hot side. You would need extra 10 dB of attenuation to make it work with consumer style hi-fi amps that have RCA inputs.
The Terratec Phase 88 has a nice break out box with RCA outputs that can be set to output -10dB signal and you don't have to get behind the computer to make connections. That one looks like a very nice interface for use with crossover software. I know it's much more costly, but you get the convenience and features that make it very good for the application.
Daveis said:
For HT use there is no way to switch the Allocator into a super low latency mode. But there is a way of delaying the video signal to align it with the Allocator's audio stream.
You just have to use a good DVD playback software. On my board someone brought up the FDshow dll that is a free download and apparently adds this functionality to any software player.
One thing to keep in mind though is the CPU load. You will need a very good machine to keep up with DVD decoding/playback and multiple instances of Allocator.
Perharps a fast dual processor machine like the new dual cores would be enough.
Daveis said:
That's actually a pretty good idea. I'm not sure how easy it would be to implement, but it would make it easier to get 'one stop shopping'.
I use SoundEasy to measure 'through' Allocator, and it mostly works ok. The big issue I'm having I believe is due to the Emu drivers and not anything to do with Allocator, which is that latency is wildly unpredictable from measurement to measurement. Makes it really tough to look at time alignment. I'm planning to swithc measurement over to an M-Audio card, but then Allocator won't be in the loop. However, it should at least allow me to get solid reference FRD's that are normalized.
What problems are you finding with the ECM8000? And, I guess how can you tell if you don't have a 'good' pre with phantom power? My take is that it's pretty good uncalibrated up to about 10k or so, and above that I consider things to be 'adjust to taste' anyway.
A good standalone A/D with AES/EBU out will probably be somewhat pricey.. The only cheap one I know of in production (I think) is the M-Audio 'flying cow', which appears to have AES/EBU out, but only goes to 48k. I have a Symetrix 620 that does both AES/EBU and spdif out, but once again limited to 48k and it's been discontiued for quite a while.
PreSonus has a combination mic pre +A/D converter (the DigiTube) but it's spdif only. You can probably use a transformer to feed it into an AES input, though.
dwk123 said:
See above. No way to put the full Allocator into a low latency mode. That's the reason Allocator Light exist.
dwk123 said:
If you wanted to just confirm the combined frequency response you could open the preset saved in Allocator in Allocator Light (the preset files are compatible). This way your measurements would not have the extra delay in them.
If you wanted to confirm the phase as well, you have to use the Full allocator and deal with the latency issues.
I guess I really should look into a measurement tool specifically written for use with the Allocator.
Maybe after the next product that's coming up is finished.
dwk123 said:
I think that all you need is a mic preamp in the audio interface. The M-audio 410 has it, the Presonus Firebox has it, the Emu 1820 has it.
Are you running into problems acessing yours from two applications simultaniously?
Thunau said:
For the folks that purchased Allocator, it would be a nice extra to bundle in Lite for those times you needed to watch a movie. Does a full Allocator purchase entitle the user to use Lite? I'm fuzzy on that.
I think I'll look into a MAudio Flying Cow. I thought those were discontinued, but I know you can still find them.
My main complaint with the uncalibrated ECM8000 was indeed it's questonable response above 10Khz.
dwk123 said:
Please have a look at the nice little utility that I wrote (Simple Automated IR Measuring Tool) to capture impulse responses available from the room correction wiki:
http://www.duffroomcorrection.com/wiki/Simple_Automated_IR_Measuring_Tool
It should run on Windows (ASIO/Direct Sound), Linux (Alsa, OSS or Jack) and OS X
Sure it's command line, but it's supposed to be part of a larger tool which can do measurement and analysis.
Many soundcards have trouble doing sample accurate sync of playback and recording. It's a driver level issue, but compounded by many audio API's not offering this as a solid option, eg it's only possible under ASIO on windows, not direct sound.
The app above will allow sample accurate recordings using ASIO, Alsa and possibly Jack and others. OSS would also be possible with a simple patch but I haven't bothered to add it.
However, to work around this you can use a second "loop back" channel in the software - by recording the sweep on a second channel (which goes straight out and back in again) we can then normalise the signal for freq response and also for delays in starting the recording and playback.
If you are pretty technical then you will have no problem adopting the basic code to measure all kinds of cool things such as static tones, etc.
With regards to Mic's. The basic ECM8000 is extremely good, it's just not calibrated. Add some random generic calibration such as you will find on the internet to make things a bit better, or buy one ready calibrated for a touch more money (lots of firms make mics based on the Panasonic capsule, but ready calibrated).
Good luck all
Ed W
Thunau said:
The problem really isn't the amount of latency (although that does require a bit of scrolling around the window playing 'find the impulse'), it's the variability. One test may put the main inpulse at 120ms, the next at 60 or 150. I'm pretty sure this is due to the Emu WDM driver, but since I can't run SE via asio, I don't actually know that for sure.
Either way, though, the 'combined system' stuff is OK, the only problem is in trying to get aligned impulses for each driver so that offset can be determined.
Given the request for AES/EBU, I was guessing that the original poster is using a Lynx AES-16 card, since most other AES/EBU cardshave some level of analog section to it.
ewildgoose said:
I've seen that, and have been meaning to check it out for DRC impulse generation. My speakers and xovers are still evolving, though, so I haven't gotten to that stage yet.
Plus, my current PC can't handle xover +DRC, but I'm almost done building out a much quieter and faster box which will handle both.
I've stopped using a reference channel for now, since I was getting completely bogus results when doing so - the reference channel was far from flat, which screwed up the results. I didn't spend much time investigating, though. Since the results I'm getting without the reference channel tend to aggree with other folks measurements of the same drivers, running without one seems to be working OK.
You're right that this makes determining the offset far easier, though.
dwk123 said:
I wonder if wiring the loopback internally for example in the Console would solve the problem? You would not compensate for the card's frequency response irregularities, but the impulse and delay should be right on.
All you would have to do is perform a measurement of latency of the DA/AD section first, then insert a delay equal to that latency in the path of the loopback.
Thunau said:
Writing a GUI should be a process of about 2 mins... There is a fixed command line and you usually don't even change any params except filename..
To be honest I don't bother because I just cut and paste the sample command out of the usage help and that's about all you need to run it! It even guesses the channel and card you are likely to want to use, so you can leave most params unentered. Basically I really don't feel the need for a gui myself...
Also as a cmd line program I tend to use it as part of a longer script that I just click on and it measures everything, calculates the filters and then drops them into my convolver. Probably it's *harder* to use a GUI than this!
MLS is not likely to be something that I implement. Log Sweeps are so much better for many reasons and have only one disadvantage which is that it's reasonably easy to melt your tweeter if you run them very loud. The SNR is vastly better and they are unaffected by speaker distortion or overload, all in all I see little reason for an MLS measurement (convince me otherwise?)
One of the biggest reasons for log sweep is the immunity to many forms of distortion... Overload the speaker into distortion and it's no problem because the distortion component drops out of the IR and can be seen (and easily measured) as seperate spikes just in front of the main IR... Nice feature!
Ed W
I've been using Thunau Allocator Lite this evening. Latencies of 6 msec seem stable on my PC using an RME card and Athlon64 3500. 3msec works but with occasional glitches.
Can anyone tell me how I can figure out the relative delay between my woofer/mid/tweeter?
I've encountered the variable latency problem even when using a reference channel.
As far as MLS vs. log sweep... isn't MLS purely an attempt to remove the room's response from the measurement. When you do long sine sweeps you are recording alot of room echo even if the mic is up close. At least with an MLS response I've been able to "see" the mismatch in delay between mid/tweeter/woofer.
I never was able to get Ed's rec_impASIO to work. And currently, I'm less interested in summed frequency response as I am impulse response. From eyeballing an MLS graph I can see at least 3 msec difference between woofer and mid.
Anyone have experience with Praxis, Sample Champion, and Sound Easy? I've heard them all mentioned for speaker measurement. I used a trial of Sample Champion for a while.
Can anyone tell me how I can figure out the relative delay between my woofer/mid/tweeter?
I've encountered the variable latency problem even when using a reference channel.
As far as MLS vs. log sweep... isn't MLS purely an attempt to remove the room's response from the measurement. When you do long sine sweeps you are recording alot of room echo even if the mic is up close. At least with an MLS response I've been able to "see" the mismatch in delay between mid/tweeter/woofer.
I never was able to get Ed's rec_impASIO to work. And currently, I'm less interested in summed frequency response as I am impulse response. From eyeballing an MLS graph I can see at least 3 msec difference between woofer and mid.
Anyone have experience with Praxis, Sample Champion, and Sound Easy? I've heard them all mentioned for speaker measurement. I used a trial of Sample Champion for a while.
Ok, after doing some research on MLS vs. log sweep, it appears that log sweeps can indeed be used for both distortion measurements and impulse response measurements. And that there are numerous benefits to log sweeps, including better S/N, etc.
I even read something about being able to eliminate room response, but it has something to do with using very long log sweeps... longer than the room decay.
A link
http://support.supermegaultragroovy.com/wiki/index.php/Log_Sweeps_vs_MLS
I even read something about being able to eliminate room response, but it has something to do with using very long log sweeps... longer than the room decay.
A link
http://support.supermegaultragroovy.com/wiki/index.php/Log_Sweeps_vs_MLS
Please read the first paper referenced in the page at the link below. It's extremely thorough and there is also an even longer version of the same paper floating around with much more detail still
However, the basic point is that MLS and low swept sine produce the *same* output result. Both of them are intended to capture the same information, and yes you can see both room reflections and direct sound on both - no it does not really matter about the length of the sweep as long as we are talking about real rooms and sensible length sweeps (sure measuring some big Cathedral may take some slightly different params!). More than 2 seconds or so should be plenty and in general I would suggest a 45 second sweep
Basically both MLS and sweeps are mathematical tricks to calculate the room IR without using significant CPU power. There is a trick with the sweep where you basically do a trick by inverting the original signal and convolve it with the recorded signal and the IR drops out. However, in theory at least you could play the radio out through your speakers, record the sound and from this recording you could figure out the IR - this requires doing some FFTs though and hence is less common (although not that complicated to code up if anyone is actually interested)
Sweeps also have the nice property that any *harmonic* distortion gets automatically removed from the impulse response, ie if you crank up your speakers to any appreciable volume then they will distort quite a bit - all this distortion is *included* in the MLS results, but *excluded* from the sweep results - cool huh!
However, there are still some types of non-linear distortion which will mess up even a log sweep measurement - typically the effects of a cheap soundcard which is resampling everything to 48Khz fall into this bucket.
If anyone is having trouble with rec_imp then please drop me an email offlist and I will endevour to fix the problem. Windows has really bad audio apis and so we need to just get problem reports for each audio card and work around the windows bugs...
If you tried a much older version of rec_imp then it was known to have plenty of windows issues, but the latest version should work reasonably well for many people. Also try installing Asio4all which can often give you more stable drivers and often allows sample accurate record and playback (ie the delay is constant and you can more easily measure driver distance)
Using a loopback channel *will* allow you to correct for soundcards which don't have sample accurate record and playback. The only situation which would break this system is where you had a soundcard which had different clocks for left and right audio channels, but such a card would never play back decent stereo signals either so I can't imagine that such a card exists. The basic idea is that you do a sweep through your audio system and also a sweep straight out and back into the soundcard. Then the gap you are looking for is the gap between the loopback impulse and the speaker measured impulse - this gap will be constant even though the absolute positions of each impulse varies
Please all consider buying something like ETF or Sample Champion if you enjoy fiddling with this stuff. I used to muck around with SPL meters for hours and yet the information you can get from a single sweep and 10 mins on ETF is *VASTLY* better than spending hours of your life with an SPL meter trying to guess what is going on! Honestly, there is a bit of learning to figure out how to interpret the graphs, but after that you will have a tool which just gives you all the crucial info about your system in one easy measurement.
Good luck
However, the basic point is that MLS and low swept sine produce the *same* output result. Both of them are intended to capture the same information, and yes you can see both room reflections and direct sound on both - no it does not really matter about the length of the sweep as long as we are talking about real rooms and sensible length sweeps (sure measuring some big Cathedral may take some slightly different params!). More than 2 seconds or so should be plenty and in general I would suggest a 45 second sweep
Basically both MLS and sweeps are mathematical tricks to calculate the room IR without using significant CPU power. There is a trick with the sweep where you basically do a trick by inverting the original signal and convolve it with the recorded signal and the IR drops out. However, in theory at least you could play the radio out through your speakers, record the sound and from this recording you could figure out the IR - this requires doing some FFTs though and hence is less common (although not that complicated to code up if anyone is actually interested)
Sweeps also have the nice property that any *harmonic* distortion gets automatically removed from the impulse response, ie if you crank up your speakers to any appreciable volume then they will distort quite a bit - all this distortion is *included* in the MLS results, but *excluded* from the sweep results - cool huh!
However, there are still some types of non-linear distortion which will mess up even a log sweep measurement - typically the effects of a cheap soundcard which is resampling everything to 48Khz fall into this bucket.
If anyone is having trouble with rec_imp then please drop me an email offlist and I will endevour to fix the problem. Windows has really bad audio apis and so we need to just get problem reports for each audio card and work around the windows bugs...
If you tried a much older version of rec_imp then it was known to have plenty of windows issues, but the latest version should work reasonably well for many people. Also try installing Asio4all which can often give you more stable drivers and often allows sample accurate record and playback (ie the delay is constant and you can more easily measure driver distance)
Using a loopback channel *will* allow you to correct for soundcards which don't have sample accurate record and playback. The only situation which would break this system is where you had a soundcard which had different clocks for left and right audio channels, but such a card would never play back decent stereo signals either so I can't imagine that such a card exists. The basic idea is that you do a sweep through your audio system and also a sweep straight out and back into the soundcard. Then the gap you are looking for is the gap between the loopback impulse and the speaker measured impulse - this gap will be constant even though the absolute positions of each impulse varies
Please all consider buying something like ETF or Sample Champion if you enjoy fiddling with this stuff. I used to muck around with SPL meters for hours and yet the information you can get from a single sweep and 10 mins on ETF is *VASTLY* better than spending hours of your life with an SPL meter trying to guess what is going on! Honestly, there is a bit of learning to figure out how to interpret the graphs, but after that you will have a tool which just gives you all the crucial info about your system in one easy measurement.
Good luck
Hi Jan
I've finally gotten around to checking out the Allocator/Arbitrator.
Things are looking pretty good so far.
I'll post a thorough writeup in after I've had a lot more exposure.
A couple of things missing that immediately struck me as being obviously needed:
- Ability to setup seperate XO schemes for the left and right channels. This is important however trivial it may sound. Imaging in particular really takes on a pin sharp quality once you have very closely matched pairs of speakers. I should imagine this feature will be fairly easy to implement.
- Dynamic skins; The program chops information off the display and those controls are inaccessible with any less than 1024x768. This isn't much good for me as my main display is a widescreen projector with a resolution of 1280x720 and my secondary display is a small 7.5" TFT with a resolution of 800x600. This may not be a problem for the masses but its something thats annoying for us that have less tradition displays. I always really like the waves stuff because they were compact but had all the options laid out nicely.
The following isn't essential but would greatly increase the appeal of the program:
- FIR filters implemented. If this is done I'd like to see the ability to customise the coefficients and window functions, as well as the more generic types. The design package LEAP is excellent in this regard but it doesn't have a crossover emulator to allow you to 'listen' to these. Again this is a big ask and I'd totally understand if you turn around and say 'ain't gonna happen buddy'. Didn't you mention a possible FIR version of the software in our emails?
- DRC integrated into the whole package so as to prevent cluttered plugins syndrome (perhaps you could get in touch with Dennis and see about using his freely available source and packaging it with your own program, all you'd then need to do is implement a convolver). This can be achieved through using Console with the addition of Voxengo Pristine Space or a similar convolver such as Waves IR1.
- The ability to measure drivers and generate .frd profiles within the program without relying on 3rd party apps to do the job, Without the .frd files, I've found that Allocator/Arbitrator is very limited in scope otherwise. Its full potential is only realised when your using actual driver data measured from the final enclosures. I know this is NOT a trivial thing to implement and perhaps relying on 3rd party apps is essential to maintain a reasonable cost for the app but I'd strongly suggest writing a more thorough manual just for this subject as you'd certainly be aiding ease of use for those newer to this. For those who've been messing with this stuff for a while now and have programs like ETF and SC, this isn't a problem.
Finally I don't mean to sound negative. All of these were just things that I noticed within the first 20 minutes of playing around with the program. Overall I think it looks promising. I'll be comparing it to the DEQX in terms of sound quality alone. I've been extremely pleased with the DEQX over the last 6 months so it will be interesting to see if I can approach and hopefully exceed what I've done with that.
I'd also love to compare it the Waves/Voxengo/DRC setup I had going very early on this year - I still swear it was better sounding than the DEQX but alas. I some 'issues' seemed to creep into the implementation. These still seem to be at large so I doubt any meaningful comparison can be drawn.
I'll get back to you with more thoughts as I get into testing more deeply.
Cheers
Ant
I've finally gotten around to checking out the Allocator/Arbitrator.
Things are looking pretty good so far.
I'll post a thorough writeup in after I've had a lot more exposure.
A couple of things missing that immediately struck me as being obviously needed:
- Ability to setup seperate XO schemes for the left and right channels. This is important however trivial it may sound. Imaging in particular really takes on a pin sharp quality once you have very closely matched pairs of speakers. I should imagine this feature will be fairly easy to implement.
- Dynamic skins; The program chops information off the display and those controls are inaccessible with any less than 1024x768. This isn't much good for me as my main display is a widescreen projector with a resolution of 1280x720 and my secondary display is a small 7.5" TFT with a resolution of 800x600. This may not be a problem for the masses but its something thats annoying for us that have less tradition displays. I always really like the waves stuff because they were compact but had all the options laid out nicely.
The following isn't essential but would greatly increase the appeal of the program:
- FIR filters implemented. If this is done I'd like to see the ability to customise the coefficients and window functions, as well as the more generic types. The design package LEAP is excellent in this regard but it doesn't have a crossover emulator to allow you to 'listen' to these. Again this is a big ask and I'd totally understand if you turn around and say 'ain't gonna happen buddy'. Didn't you mention a possible FIR version of the software in our emails?
- DRC integrated into the whole package so as to prevent cluttered plugins syndrome (perhaps you could get in touch with Dennis and see about using his freely available source and packaging it with your own program, all you'd then need to do is implement a convolver). This can be achieved through using Console with the addition of Voxengo Pristine Space or a similar convolver such as Waves IR1.
- The ability to measure drivers and generate .frd profiles within the program without relying on 3rd party apps to do the job, Without the .frd files, I've found that Allocator/Arbitrator is very limited in scope otherwise. Its full potential is only realised when your using actual driver data measured from the final enclosures. I know this is NOT a trivial thing to implement and perhaps relying on 3rd party apps is essential to maintain a reasonable cost for the app but I'd strongly suggest writing a more thorough manual just for this subject as you'd certainly be aiding ease of use for those newer to this. For those who've been messing with this stuff for a while now and have programs like ETF and SC, this isn't a problem.
Finally I don't mean to sound negative. All of these were just things that I noticed within the first 20 minutes of playing around with the program. Overall I think it looks promising. I'll be comparing it to the DEQX in terms of sound quality alone. I've been extremely pleased with the DEQX over the last 6 months so it will be interesting to see if I can approach and hopefully exceed what I've done with that.
I'd also love to compare it the Waves/Voxengo/DRC setup I had going very early on this year - I still swear it was better sounding than the DEQX but alas. I some 'issues' seemed to creep into the implementation. These still seem to be at large so I doubt any meaningful comparison can be drawn.
I'll get back to you with more thoughts as I get into testing more deeply.
Cheers
Ant
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.