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Ive recently been thinking about upgrading my modest desktop PC audio system to something more high end. Ive been doing alot of research and i plan on doin a DIY active speaker system witrh my PC as the source. The Xover was the biggest problem for me as I have no expereince with active Xovers and my budget is small. I stumbled upon the Xover plugin for Foobar and it seems to be perfect for what I want to do. Has anyone used it?? I have a bunch of questions about this Xover.
1. Do you have to use a 7.1 surround sound card?? Or can a card like the M-audio Delta66 series work, it has 6 analog outputs.
2. If it doesnt have to be a 7.1 card, can a card with multiple digital outputs work?? For example if the soundcard has 3 seperate digital outs, could you use a seperate dac for the lows, mids and highs?
3. Is the Xover ASIO compatible?
4. How is the sound quality of the Xover plugin? Does it degrade the sound or is it transparent?
I hope someone out there has expereince using this plugin, i havent had much luck finding much info on it. Id like to get the design for the system all sorted out before I start buying parts to build the amps and speakers. The Xobver has been holding me back the most and a software xover solution would be great. Please help.
thanks
Follow Ups:
Hey Frank:I spend quite a bit of time doing different active setups using three different DSP based crossovers for Foobar. One also works in Winamp.
1. Do you have to use a 7.1 surround sound card?? Or can a card like the M-audio Delta66 series work, it has 6 analog outputs.
With the foobar kernel streaming output plugin, you can use practically any 4, 6 or 8 channel soundcard - I've even made 8 channel onboard sound codecs work. (Of course, 4 gets you two way, 6 gets you three way and 8 gets you four way). You don't *need* an 8 channel souncard if you are only doing a 2 way for example, in which case a 4 channel card would work fine - and only four out of 8 crossover "poles" would be in use.
Just remember that only kernel streaming or ASIO will get you past the Windows Kmixer and provide for non-resampled bit-perfect playback. Foobar can play back 32/192 files if set up properly, whereas Winamp is only good up to 24/96. Be aware that for either program, your plugins (crossover included) will likely be the limiting factor in sample rate and bitdepth
2. If it doesnt have to be a 7.1 card, can a card with multiple digital outputs work?? For example if the soundcard has 3 seperate digital outs, could you use a seperate dac for the lows, mids and highs?
The best solution for digital is the AES16 (as Dawnrazor said) or perhaps a simlilar RME card that has multiple digital outputs. Beware of the prosumer cards with multiple optical outputs. Sometimes they are SPDIF in, SPDIF out, ADAT, and other functions and are NOT outputs for each set of stereo channel pairs. Three way digital out is a good way to go... but spending all that money and still using SPDIF/AES? I'd say you'd be better off getting the Lynx 2B and using it's analog outs. Going with a Lynx AES16 and three DACS is going to cost a lot - and to get better DACS than those in the Lynx2B you're looking at about $1000 per DAC at least. This would mean $3K to $6K in dacs and you have not even bought three stereo pre-amps yet! Digital might be marginally better, but its going to cost 3 times as much when bi/tri-amping using the Lynx 2B.
3. Is the Xover ASIO compatible?
All DSP plugins are inherently ASIO compatible because they have not much to do with the final output. Only output plugins need to be ASIO compliant if you want to use an ASIO compliant soundcard. There are good ASIO output plugins available for both Foobar AND Winamp from our friends at www.otachan.com.
4. How is the sound quality of the Xover plugin? Does it degrade the sound or is it transparent?
4a) Xover is very sonically transparent. I'd say it's almost as transparent as the Thuneau Allocator - and it only fails in comparison because as nice as it is it's still rather limited in it's functionality. You can only select "textbook" filters (Butterworth, Bessel and Chebyshev) which are "fixed Q" filters. No Linkwitz/Riley is a total bummer as well - since LR2 and LR4 are extremely popular. Delay control is awesome with this plugin, and so is channel mapping - but channel mapping is not always intuitive - ALWAYS test tweeter output using a midrange first. Be aware that XOVER is designed to work with 44.1 kilohertz sample rate. If you play back 48, 88.2, 96, or 192 material, your crossover frequencies are going to shift upward according to the sample rate ratio. Not good, but you won't blow a tweeter either. You *could* theoretically create a special settings file for each sample rate... but this is a rather convoluted approach if you ask me. Maybe creator Francois Bourdon will implement multiple sample rate support one day - maybe he won't. He's basically giving us access to HIS project for FREE so who knows where he will go with it.
4b) The Aedio Japan FIR based crossovers are totally different animals. They come in up to four-way, but use FIR crossover filters with adjustable filter and delay "taps". These are linear phase crossovers and thus transient perfect, but there is no "phase correction" for the loudspeakers themselves, so although the filters themselves are transient perfect, the system will most likely not be. On the standard version, you must select one crossover slope setting (40 to 400db/ocatve) [!] which will automatically use the right tap for each filter based on crossover points. There is another version I have which is slightly different - you can enter custome filter taps for each filter, but one disadvantage: for the midband you can only select ONE tap for both highpass and lo-pass. This is a dead end if you are trying to do a "real serious" FIR crossover setup that would require separate lowpass and highpass tap values. The Aedio crossovers are not sample rate dependent however, which is a nice touch. Steep slopes like this are the cats meow for getting ribbon tweeters down as low as they will go as well. Shallow slopes + ribbow tweeters = bad news.
Also, although linear phase filters are by themselves transient perfect, some claim the "pre-ringing" associated with FIR filters is audible. Others claim it's not audible. And I definately don't want to debate that here. That's a topic for Propeller Head Plaza!! :P
4c) The Thuneau Allocator is by far the best DSP based crossover plugin available. It's not free... but it's a far more comprehensive crossover AND it can be ordered with a phase correction section that compensates not only for filter and driver phase shifts, but phase shifts associated with low frequency enclosure alignments. It's a VST plugin and comes with it's own VST host, but will work will other VST hosts as well. The Arbitrator works with free 2-channel VST hosts for Winamp, but 2in/8-out VST hosts for Winamp are only on the breadboard at this time - so you need to use ASIO output plugins to use the Allocator. Using multile bi-quad filters with FULLY adjustable Q, you can tune filters to have any Q value (and also create the standard Linkwitz/Riley, Butterworth, Chebychev and Bessel filters if you desire). Channel delays are a snap. The trick with the Allocator is that you NEED an ASIO based card. Also, if you are using a media player and not an external sound source, you *may* need to use two output channels to stream the player output (using an ASIO output plugin) onto the ASIO bus, and use the remaining crossover channels as filtered outputs. This way, an 8 channel card (4 stereo pairs) is only good for THREE way operation (6 stereo pairs). The Thuneau Allocator and Arbitrator work with both Foobar AND Winamp, provided you use the right output plugin(s) and have the right sound card(s). This is the case with the Creative X-Fi that I just got working with the Allocator this week. Sound? For $99 it can't be beat. But the analog outputs of the X-fi pale in comparison to the $999 Lynx 2B - as you would expect for 1/10th the price. But for 1/10 the price they don't sound 10 times as bad by any means! (Only maybe half as bad... wink wink). The Xfi is really only good for it's bitperfect and ASIO compliant spdif output, which is truthfully why I bought it. The analog outputs are "not bad" though, and definately beat my M-Audio Revolution 7.1 for high end detail, sizzle and shimmer. I think the M-Audio has a somewhat fuller bass though. PCI card analogue outputs on inexpensive cards just don't cut it truthfully.
The Allocator is not sample rate dependent - it will output the same sample rate as what is put in. It's output, however, is always 24 bit regardless of input bitdepth. The Allocator *for sure* uses 64-bit processing, where I am not sure what the XOVER and Aedio crossovers are doing (likely 32 bit?). This could also be why the Allocator sounds better to my ears. IIR filters are known to have no "pre-ringing" like FIR filters, are moreso approximate real-world passive filters, but also suffer from the same phase response of these filters... until now. The Arbitrator gives you the transient perfect nature of FIR filters, but uses IIR filters with no "pre-ringing". Brilliant actually.
My advice?
Go with the Lynx 2B with the Thuneau Allocator and Phase Arbitrator.
It's PROVEN to work with the Thuneau products, and is probably the best sounding PCI soundcard out there. It's expensive, but not as expensive as buying MID-FI CARD after MID-FI CARD (like I have) to find out that they all basically sound mediocre to good at best. With the Lynx 2B, if you DON'T like the sound, at least you can sell it (as it should hold a fair bit of value) and NEVER look back at PCI solutions - since there is likely nothing better out there in the PCI arena.
USB solutions are the talk of the town in the PC Audio Forum these days, and some folks have built some real dandies - but you would need three of the darned things to tri-amp, and there are no reports out as of yet that confirm whether or not you can use multiple instances of the same USB sound device and manage to map the crossover channels (low/mid/hi etc.) sucessfully.
There you go. Everything (and more) that you ever wanted to know about DSP crossover plugins. Hope this helped. Lucnh time is over and now I gotta do some *real* work really fast...
Thanks again to Francois Bourdon (Xover), the folks at Aedio Japan (Foobar FIR crossovers), the Otachan folks (Foobar and Winamp ASIO output Plugins), Jan at Thuneau (TOP product support - thanks Jan), Chun Yu for the original Winamp kernel streaming output plugin and Steve Monks for improving it. But most of all - the creators of Winamp and Foobar - for making this whole PC playback (with or without DSP crossovers) a possibility for all of us in the FIRST PLACE! :o)
Cheers,
PrestoNow here's where to get all the goodies!!
Foobar/Winamp/External Source IIR crossover with Phase Correction:
www.thuneau.comFoobar FIR crossovers:
http://www.aedio.co.jp/download/Foobar X-over (you have this already)
http://xover.sourceforge.net/Foobar ASIO Output Plugins:
foo_output_asio(dll).dll (dll version) Ver. 0.51 (2006/2/25)
http://otachan.com/foo_output_asio(dll)_051.7zfoo_output_asio(exe).dll (exe version) Ver. 0.54 (2006/2/25)
http://otachan.com/foo_output_asio(exe)_054.7z*************************************************************
Winamp Kernel Streaming Output Plugins:
http://www.stevemonks.com/ksplugin/Winamp Otachan ASIO output plugin (dll)
http://otachan.com/out_asio(dll)_067.7zWinamp Otachan ASIO output plugin (exe)
http://otachan.com/out_asio(exe)_070.7z
Hey Presto,THanks for all the good info.
On the Thuneau Allocator, can you clarify what you mean about the 8 channels shrinking to 6, and 6 to 4 when using a media player versus an external source? Can you explain why this would be the case, and not the reverse, and the "may" part.
See, if I get this, it will be for 2 primary reasons: 1. better sound and flexibility, and 2. it handles sources outside a media player.
If I loose 2 of my Lynx 2bs outputs with Foobar, then this is no longer a viable options.
The lynx 2b actually lists for $899. Ironically, it is the cheapest version of the Lynx 2 platform, with the B being the most expensive do to the 3 pairs of inputs. Thanks to Lynx for pricing it based on pro audio recording needs!
Also if I could be so bold as to include Patrick Cazales in your list of helpful people. OK, his acxo is essentially useless these days when compared to other options, but he was ahead of his time, and still is the only program that combines a music player, crossovers and room correction.
Finally, have you tried the Foobar xover with onboard 5.1 audio options? I was wondering if this could make a viable budget system with active crossovers. Think of it, for say $300 or so one could have a PC with active crossovers matched with some of those $30 T-amps with the right speakers, that could be a killer budget system.
"If I loose 2 of my Lynx 2bs outputs with Foobar, then this is no longer a viable options."I don't think so...this was just the case with the X-fi I was experimenting with. I think there are few Allocator/Lynx 2B users on the Thuneau forum - maybe post there as well. I think the X-Fi ASIO mixing capabilities (as impressive as they are) do not have a "back door" way of piping the audio stream onto the ASIO bus. You then need to use Front L/R channels to import the audio stream. Then, you have 6 out of 8 channels left for use with the Allocator outputs. There may be a way to do channel routing/assignments (or perhaps channel shifting) so that all 8 are usable.
Thinking Lynx 2B allows use of all 8 though...would need to confirm with Jan.
"Also if I could be so bold as to include Patrick Cazales in your list of helpful people."
Sure, but he was not on the list because I only was acknowleging the specific people who's work I was referencing in my post. There's Denis Sbragion with DRC as well... there are *so* many helpful people out there... like I said, that was not a "all inclusive list", but just thank-you's for those who's material I linked to.
"Finally, have you tried the Foobar xover with onboard 5.1 audio options?"
No - but xover would not work for 5.1 audio. It's a 2-in 8-out crossover plugin. You would need 6-in 11-out for 5.1 with two-way satellites and 6-in 16-out for 5.1 with three-way satellites... etc. etc. If you attempt to use xover with multichannel plugins, it's only going to "see" the front L/R channels and divide THEM, and do nothing with the center, rear and sub channels. (At least that is how I imagine it would go.)
That's not to say it can't be done - but not with xover (or Allocator or Aedio crossovers) as they are currently written. It would take DSP-multiple-instance tricks and other channel mapping tricks to work - not to mention all those channels you would need to export (so multiple soundcards and the issues with *that* would also come into play).
Hey presto,I wasn't being all that clear. In my world it is a given that no one in their right mind would want to do home theater. So what I meant to say was has any one used the 2 channel Foobar crossover plugin with on board audio....using 6 of the 6 channel outputs for a 3 way output.
THe plugin uses "sub" and "right rear" for example in its mapping, so I fu\igured that using onboard audio with 5.1 outputs (3x2 actual jacks) one could get a 2 way, if not a 3 way crossover.
I might be wrong that there are some pcs that have 3 miniplug jacks that support 5.1, but I can't think of how else 5.1 pc speakers would connect.
Dawnrazor:I agree. I would not bother going through expense and complexity of doing active multi-way surround speakers to do compressed "sound effects" for movies. Someone who collects DTS or 24/96 multichannel music may disagree and may even have a point, but I have so little software of this nature, I can't really justify even the time to come up with the design.
Anyways,
I think Frank has all the information he needs thanks to your post there about the Lynx 2B.
I am going to spend *as much* as a Lynx on cheap soundcards because I just enjoy buying cheap soundcards and seeing how much performance I can 'squeeze' out of them.
When I get tired of this game, indeed, I will just by a Lynx - and sync it to a helluva good DAC. None of this asynchronous stuff! :o)
What exactly do you mean when you said sync the Lynx to a good dac?? Do you mean like a word clock that they use in pro recording? How would you do this with a dac made for hifi audio??
Frank:You'd be surprised how many "hi-end" DACs have a wordclock output that will spend it's lifetime just sitting there collecting dust. 99% of "hi-end" guys simply use SPDIF over TOSLINK or COAX in "Asynchronous mode".
The Pros (and some hi-fi guys) know that asynchronous works - but it can really be prone to jitter. Typical "receivers" are CS8416, CS8420, and CS8421 which use what is called a "phase locked loop" (PLL) to deal with the fact that the clocked data coming is was clocked with a DIFFERENT clock that the clock than the one used for the receiver (and dac at the receiver end). The clocks don't have to be out by very much to be "skewed" - and without the PLL the clocks would eventually drift out of sync. The apparent trouble with PLL's is that any jitter before the SPDIF receiver is a whole lot worse going OUT of the receiver into the DAC.
So what some folks do is follow what is done in professional environments that handle digital transfers all of the time - they do synchronous transfers in what is called "master" mode. In master mode one device is deemed the master and it's clock is used. ALL other devices in the signal chain are then slaved to the master and use that same clock. One clock. No skew. No PLL. No sync issues. Brings a tear to your eye doesn't it?
Now to do this with the Lynx, I imagine that you would need to make it the slave and set it up to use an external clock. Why? I think quite a few DACs have "wordclock" output, but not too many at all can be set up as a slave and have a "wordclock" input. So, if I was the owner of a Lynx 2B, I would find a DAC that has a wordclock output and slave it. Another reason (and perhaps a better one) to make the Lynx a slave and not a master is that it seems using a PC to generate audio clocks is a "foul word". No matter HOW good the card, it seems folks around here are nervous about using PC's for digital OR analog circuits because of the noise that plagues computer power busses. Other folks counter-argue that good filtering of the supply is possible. Myself, I think "PC noise ruins everything" is a very very crude model: why can PC's have PRISTINE noiseless video output via a PCI card that relies on scanning frequencies in the kilhoertz but be totally useless for *audio*? I think the reason why so many PCI cards suck for digital is because they use asychronous SPDIF (and worse yet optical SPDIF which requires an electrical-to-optical and then optical-to-electrical conversion). I think PC SPDIF is no better that consumer SPDIF and depends on the hardware and the ability of the circuit designer to deal with noise and get clean power to the transceiver chip.
So this is why I've been waiting and have not sprung for ANY high-buck PC solution yet. I had/have so much to learn. I want to do this right. The best way. The low-jitter way. And if that means getting devices that use synchronous / master-slave connections with separate word clock connections then no problem. It's going to cost more sure, but not if I had done it the "trial and error" audiophile way.
Digital transfers are engineered solutions. The nice thing about engineered solutions is that they can work for you the first time. This is because the first 9 times were done "virtually" (on paper) and thrown in the garbage can. In my case, the "engineering time" (90% of which is spent learning) is free because this is my hobby. So it makes sense to do "free engineering" first and buy equipment later.
I've only bought cheap soundcards in the meantime just to keep myself amused, and learn about asynchronous SPDIF connections, and how a PCI card that has a world-class DAC that specs like a NASA rocket can end up sounding like thin and bloated crap. This, in my opinion, is not because of "parasitic PC power supply noise", but because of a multi-channel analog section that was scaled down to fit into the size of a postage stamp to fit on a PCI card.
With S/PDIF, or ADAT, or even I2S, the data transfer is always "synchronous" -- **not** asynchronous. The sender's clock is always the master clock; the receiver has no independent clock. The clock may be encoded in the data (as usual with S/PDIF) or sent via a separate wire (as with I2S, or using the "word clock" output of a high-end sound card or an Apogee Big Ben), but it's always provided by the sender.The only time things get "asynchronous" (i.e., when there's an **independent** clock in the receiver) is 1) when an asynchronous sample-rate converter (ASRC) is being used to change the sample-rate of the incoming data to match the receiver's clock or 2) when there's a whole lot of buffering going on (as in the old Genesis Digital Lens).
Ordinary S/PDIF is **not** "asynchronous". It may be more jitter-prone than having an independent clock line, but it's still synchronous.
Oh, BTW, there's another style of synchronous that reverses the direction of the clock. Instead of the clock being sent by the (data )sender to the (data) receiver, the clock is in the receiver (the DAC) and is sent **back** to the sender (transport, or whatever) to determine the rate at which the sender sends the data.
"Ordinary S/PDIF is **not** "asynchronous". It may be more jitter-prone than having an independent clock line, but it's still synchronous."Okay. That makes sense. Just because it's clock is embedded does not mean it's asynchronous... I'll buy that.
"Oh, BTW, there's another style of synchronous that reverses the direction of the clock. Instead of the clock being sent by the (data )sender to the (data) receiver, the clock is in the receiver (the DAC) and is sent **back** to the sender (transport, or whatever) to determine the rate at which the sender sends the data."
There we go. Different master/slave configurations. Still synchronous, but just different sources for the clock signals.
Thanks!
So what's your take on slaving a soundcard to a DAC instead of just using regular old SPDIF? Worth doing?
> So what's your take on slaving a soundcard to a DAC instead
> of just using regular old SPDIF? Worth doing?Yeah, I guess. The only consumer DACs I've heard of with
word clock outputs are those that have been designed to send a
clock signal back to that same company's own matching CD transport.
There haven't been all that many of them. Whether such a DAC
would work with a high-end sound card with a clock input (a Lynx?),
I do not know. If you had your heart set on using a DAC that
way, it's something you'd want to try before you actually paid
for it.
> Typical "receivers" are CS8416, CS8420, and CS8421 which use what
> is called a "phase locked loop" (PLL) to deal with the fact that
> the clocked data coming is was clocked with a DIFFERENT clock that
> the clock than the one used for the receiver (and dac at the
> receiver end).But the receiver (and DAC at the receiver end) **doesn't** have an independent clock in this case! The clock is recovered **by** the receiver, and its (average) frequency is **identical** to whatever's coming in over S/PDIF. With a sound card getting S/PDIF, you'd set CLOCK to "EXTERNAL" (**not** "INTERNAL" -- in the latter case there **would** be an independent clock at the receiving end. Have you ever done this by accident? It doesn't sound pretty!)
> The clocks don't have to be out by very much to be "skewed" - and
> without the PLL the clocks would eventually drift out of sync.Well, no -- without the PLL (either in the receiver or an external one), there would still be a (recovered) clock whose frequency (**on average**) would be identical to the sender's clock. There would be no "drifting out of sync". Without a PLL (or multiple PLLs), there'd just be more jitter (variations around that average recovered frequency), that's all.
> The apparent trouble with PLL's is that any jitter before the
> SPDIF receiver is a whole lot worse going OUT of the receiver
> into the DAC.On the contrary, PLLs **reduce** the jitter in the recovered clock.
Now, in something like the old Genesis Digital Lens, there **is** an independent clock that's (more or less) out of sync with the recovered clock from the S/PDIF input. And there has to be a large buffer (that's either slowly filled or slowly drained) to compensate for the differences in the clock frequencies. That scheme would have done more good if there'd been a DAC chip **inside** the Lens!
There's the culprit:The Behringer DOES use an ASRC with a PLL.
So this is asynchronous then?
This is indeed Asynchronous Sample Rate Conversion...
So why do it? Does it keep cost and complexity down to have a single bitdepth and sample rate coming into your process? I imagine it would.
But what are the disadvantages as far as audio quality is concerned?
> The Behringer DOES use an ASRC with a PLL.
> So this is asynchronous then?
> This is indeed Asynchronous Sample Rate Conversion...Yes. It's the ASRC (and the independent clock) that make
it asynchronous. The latest digital input receivers
(CS8414, CS8416, etc.) have PLLs built into them. In any
case, there are theoretical reasons why it's a good idea to
eliminate as much jitter as possible (ahead of the ASRC)
even if you're using an ASRC.> So why do it? Does it keep cost and complexity down to have
> a single bitdepth and sample rate coming into your process?
> I imagine it would.Well no, adding an extra chip (an ASRC) doesn't save any money.
If you're doing studio work and you have sources at different
sample rates that you need to edit together, then an SRC
(or an ASRC) becomes desirable (the Kmixer problem, again ;-> ).In consumer equipment, ASRCs are showing up because they're:
1) available, 2) relatively cheap and 3) quite good, in the
latest generation. They let a piece of equipment perform
"upsampling" which, whether or not the designer thinks it actually
improves the sound, is no doubt a marketing advantage. Also,
having an independent clock right next to the DAC chip has
theoretical advantages as far as jitter is concerned.> But what are the disadvantages as far as audio quality is
> concerned?ASRCs have their own effect on the sound (measurable, whether it's
audible or not is another question -- some folks have claimed
it is).
Hey Presto,If you owned a Lynx 2b, I am willing to bet that you wouldn't use the digital out too much, since the onboard dacs are stellar for the price...well unless you have deep pockets and just want to play around.
Not an expert by any means on the sync you are talking about, but the Lynx does seem to have that capability, and I remember that Fmak and Tuckers were discussing that very thing a few months ago.
ANyhow, it is really hard to beat a Lynx 2b if you are doing active crossovers like Frank is discussing.
Dawnrazor:I agree with you. Lynx2B analog outputs with three way stereo capability...for $1000? You can't beat the performance to price ratio.
Going with a Lynx AES16 ($650) out to a Lynx Aurora8 external D/A converter ($2200) is about $3000 with tax. Although it does give you the ability to slave the AES16 to the Aurora (DAC), whether it's worth about three times the price to do the same "function" remains to be seen.
There would have to be a definate improvement in jitter performance going the synchronous route - and it's hard to say. The Lynx 2 family supposedly has jitter under control, and since there is no tranmission of the digital signal (it goes straight to the DACs located on the same board so it's likely I2S and not converted to SPDIF or AES) this could well be the case.
I'm not worried about the cost of the Lynx. It's the cost of the three identical amps I want, complete with a 6-channel passive line stage that is going to cost me what a small car would cost.
And small car is what the girlfriend wants...
Groan.
P,Good luck on the girfriend front. The potetnial is that the wants and needs will get even worse when they become wifes. Unless you pick a really good one, like I did.
Anyhow, I don't think there is a need for that passive line stage.
I know the text books say that one shouldn't use digital volume controls, but if you tried using the Lynx 2bs direct in to an amp, I doubt you would even consider adding a passive.
Maybe it is because of outputting 32bits in Foobar, but all that loosing bits and resolution just doesn't shake out in practice.
Said another way, spending the money on better amps and speakers will yeild a better sound than running lessor quality speakers and amps through a passive.
BUt, you could buy the 2b run it direct, and then buy a passive...just make sure you can return it, which I suspect you will.
Thanks guys for all the info, ive learned alot the last few days. Hey Presto I know you like to mess around with inexpensive soundcards. Have you hecked out the new one from Onkyo the SE-200 pci?? What makes it interesting to me is that its a 7.1 card, but has a seperate high quality 2-channel section with its own dac, analogue stage and outputs. Now I know you cant tell much just from looking at it, but the parts quality look to be far better than the Xfi. A freind of mine has the older Onkyo SE-150 and he likes the sound better than the Maudio 24/192 which he also owns.
Frank:The Onkyo might be a superior solution for those that want a simple 2-channel card and use analog outputs. They seem to have made the analog stage more of a priority... with large caps and some kind of screening around the analog section. Definately an "atypical" looking card.
I think I am going to stick with tri-amping digitally though. The process is now so transparent (and can be had with phase correction for IIR filters) that I think it's worth doing.
This does call for either a Lynx 2B using it's analog outs, or a Lynx AES16 with multiple external DACs.
There is no end to how much someone can spend, as always.
Everyone has to draw their own "law of diminishing returns" line in the sand.
But thanks for bringing to light that rather unique looking soundcard. Definately seems to be a purpose built solution by audio people and not just another "computer card that does audio".
Frank:I mentioned the two channel VST plugin for Winamp to run ONLY the Thuneau "Arbitrator" (phase correction) with OR without ASIO. Kernel streaming works well with this plugin.
http://www.savioursofsoul.de/Christian/VST/dsp_vst.zip
The author, Christian Budde, is a really smart and stand-up guy and has provided me with awesome help on other related projects.
Thanks for all your efforts Christian!
Thanks, im starting to think this project is alot more complicated then i first though. Maybe a pair of active monitors plus a sub would be easier and have a better chace of sounding good. Actve Quad12l or Paradigm 20's come up for sale once in a while. Ive always wanted a full active system ever since I heard a freinds active Linn setup years ago.
Frank:PC Audio does not have to be complicated.
For 2-channel, you either go USB or SPDIF out to a good DAC or you go with a good analog out card like the Lynx2b. Then you decide whether or not you want to use the PC for DSP upsampling, digital room correction, and/or crossover functions. If you go DSP for crossover functions, you either get a Lynx2b (and use all six channels) or Lynx AES16. Digital or analog out. Pick your poison.
Oh. By the way. I think the Lynx 2B is 2-in 6-out. I think the way they make them is 4in/4out (A model) 2in/6out (B model) and 6in/2out (C model) There *IS NO* 8 out model. So I don't understand why you are "losing 2 channels with the lynx and getting 6" thing. You never had 8 to begin with. But I do understand the losing 2 out of 6 concern. NO! I don't think you will lose two out of six with the Lynx - there are many guys tri-amping using ASIO with the Lynx but please to confirm with Jan at Thuneau before you buy. Using the Allocator requires there to be a ASIO software mixer input to which one can stream audio from an output plugin of a software player . I know guys are using the Lynx analog cards with the Allocator but I'm not sure how many channels they are using. Best is to be sure.
Now, as I said, with the X-fi, you "lose" 2 out of 8 (getting the audio stream onto the ASIO bus) but you still get six! ;) And then you STILL get full-range out of the Front left/right ANALOG output, but to use them with another crossover (or just for headphones or something) you would need to do latency compensation. The digital output is useless though - it's the input signal plus all of the individual crossover channels summed up and happening after the process delay. (The Allocator and Arbitrator processes take TIME which means using the full range input signal for additional external functions is possible but would take some delay aka latency compenstation. No big deal if you really needed it though - just dial in the some delay on the external crossover and use an impulse test to line the two up.)
For multi-channel, you once again have two choices - digial or analog out. But when it comes to the crossover game, for multi-channel I'd say it would be better to stay external - even if you are hell bent on going active UNLESS you are a PC / software / programmer guru.
Sorry for my babbling. I'm worse than ever the last few days. I've been tired, more verbose than usual (if at all possible) and retyping what I am writing and backspacing so much I think I'm getting tired of being tired.
Is that ringing inside my head? Oh. It's just the fridge. Phew.
Thanks for all the help. I think that the idea of a DIY active speaker may be a little out of my league right now anyways. But It is still a very interesting idea for the future when I have a little more expereince. I think im going to stick with a more "normal" system. Something like this1. Foobar2000 using Asio or KS for playback
2. upgrade my Xfi to a better sound card(lynx, rme, emu...)
3. Upgrade my small M-audio DX-4 with something bigger/better, I really like the sound of the Mackie 624/824
4. Queit down my pc, its a high end dual core machine and its a bit noisy, or possibly just build a new "silent" PC.
So those are my plans for the near future. And I will have the hardware in case sometime down the road I want to try out the DSP stuff like DRC, Xover, Upsampling...etc.
Thanks alot for all your help and answering my newbie questions:)
Frank:Hey no sweat Frank.
Just one thing to mention. Keep the X-Fi around. I am still not sure what the jitter situation is coming out of that card - and it *does* do SPDIF in "Audio Creation / ASIO" mode, meaning you can stream to the card using an ASIO output plugin. Otachan's ASIO output plugin has a built in Upsampler too! :o)
To be honest, I have my druthers about SPDIF out of a PC. USB is looking more and more attractive, simply because it so neatly sidesteps the jitter issue by sending only the audio data over, where it is clocked at the receiver end. SPDIF in asynchronous mode can be very jitter prone. SPDIF in synchronous mode requires hardware that can be set up in master/slave mode. Lynx cards have a sync cable, meaning that they can be operating in synchronous mode with a DAC, and the DAC can be slaved to the Lynx (or the other way around) so that you only have ONE clock source (the master). This is *allegedly* much better but is not necessarily a panacea depending on who you talk to.
But keep the X-fi. My X-fi Xtreme Music was had for $99. If it can do a decent SPDIF with ASIO it's worth keeping around. I'm gonna find out why it sounds different than my other SPDIF outs that I've experimented with. I honestly can't say whether it's worse, better or just plain different. But for some reason (probably jitter numbers) it *IS*.
Cheers,
Presto
1. No. It works fine with my Lynx 2b's 6 outputs. Did I mention that you should get this card???2. Never tried it as the Lynx I have only has one digital out, but yes I think that it would work fine. THe Lynx AES 16 mixer should work no differently than the 2b's mixer.
3. Yes. Actually, that is the only way it works TMK.
4. I have only been listening to it through headphones, and a recent move will allow me to set up the main system. THe head phones have not showed anything that would indicated a major reduction in clarity.
Ultimately I think I will spring for the thuneau allocator, but will use the Foobar one for a while once I get the system up and running.
Dawnrazor."DUH!" :p
You *HAVE* the Lynx 2B. So I guess you know how it works too! lol
Thanks for helping us out.
Yes. That *is* good news. So you DON'T need to use 2 out of the 6 to get the stream onto the ASIO bus - the Lynx has sufficient software mixing capabilities to do it.
That's very important to know.
They have one in stock at the local audio shoppe... :D
Hmmmm.....
Thanks DR.
That is with Foobar that it works.But I got the sense that he Allocator might function differently since it was "global" and independant of any particular program.
Are you saying that it indeed wouldn't use 2 outs on the Lynx to do what it does?
Dawnrazor:No no... :o)
I was confirming what I thought *you* said.
Well, the easiest way to know for sure is to try.
Since you have the 2B, you could try the Allocator demo and attempt to stream audio from a software player and report back.
I thought you had already tried this. Okay. I have lost the ability to read in the last couple of days. :o) I'm overtired. This thread proves it.
Sorry for the confusion.
Yeah, I know what you mean. I think I was born overtired!I HAVE tried the FOOBAR crossover. Not the Allocator.
Foobar doesn't use up an extra channel, but since the Allocator works all the time, all bets are off.
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