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In Reply to: Re: Foo_Xover plugin answers. All of them. At once. (Long) posted by Dawnrazor on January 12, 2007 at 19:46:58:
"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).
Follow Ups:
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".
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