Thread: [Alsa-user] "best" card for "bitperfect" SPDIF I/O with external clock sync ?
Brought to you by:
perex
Menu
▾
▴
alsa-user
|
From: Paolo S. <pms...@ya...> - 2007-11-21 19:33:54
|
Hi everybody,
I have a question for you guys.
(I'm new to this list and hope this is the right place to ask
and not an FAQ... I've tried to search to see if this question
had been answered previously, but I have been not able to find
what I was looking for).
I'm thinking about using a dedicated (fanless, diskless, etc)
PC as a digital source for my hi-end Hi-Fi system.
Of course I plan to use the PC only to provide a "bitperfect"
(exact copy of the original media, normally CD) digital stream
to an external DAC.
As you probably know better than me, the one major known problem
when you strive for the highest possible quality in digital audio
reproduction is jitter... and the best (if not only) way to really
minimize it is to use a good, clean and stable clock close to the
DAC chip, slaving everything else to that one.
Thus, what I would need to do would be to "slave" the sound card
SPDIF output clock to the external DAC clock i.e. to make this one
become the "master clock" for the whole digital audio stream.
AFAIK, one possible way to do this is to set up a "fake" SPDIF
output from the external DAC and connect it to an input of the
sound card whose SPDIF output goes to the DAC for conversion.
Of course the sound card must be able to "slave" (synchronize)
its SPDIF output clock with the one coming from its SPDIF input.
(BTW: are there other -perhaps easier and/or better- ways to do
what I would like to do?)
Thus, I would need a sound card which must be:
* cabable of "bitperfect" (pass through) operation at CD standard
16bit/44.1KHz (as well as, possibly, also at higher resolutions and
sample rates such as 16/48, 24/48, 24/96 and 24/192).
* capable of "slaving" its SPDIF output clock to an external one,
such as the one reconstructed from its SPDIF input.
...last but not least, of course all of this must be done on Linux,
thus the sound card must be fully supported by ALSA! 8-)
Well, yet another requirement is... that it should possibly (and
hopefully) not cost me a fortune! $-)
Thanks in advance for your attention.
Ciao e grazie,
Paolo.
--
Skype: Paolo.Saggese
http://borex.lngs.infn.it/saggese
You can still escape from the GATES of hell: Use Linux!
|
|
From: Sergei S. <ste...@li...> - 2007-11-21 21:20:12
|
On Wed, 21 Nov 2007 20:33:39 +0100 Paolo Saggese <pms...@ya...> wrote: > Hi everybody, > > I have a question for you guys. > > (I'm new to this list and hope this is the right place to ask > and not an FAQ... I've tried to search to see if this question > had been answered previously, but I have been not able to find > what I was looking for). > > I'm thinking about using a dedicated (fanless, diskless, etc) > PC as a digital source for my hi-end Hi-Fi system. > > Of course I plan to use the PC only to provide a "bitperfect" > (exact copy of the original media, normally CD) digital stream > to an external DAC. > > As you probably know better than me, the one major known problem > when you strive for the highest possible quality in digital audio > reproduction is jitter... and the best (if not only) way to really > minimize it is to use a good, clean and stable clock close to the > DAC chip, slaving everything else to that one. > > Thus, what I would need to do would be to "slave" the sound card > SPDIF output clock to the external DAC clock i.e. to make this one > become the "master clock" for the whole digital audio stream. > > AFAIK, one possible way to do this is to set up a "fake" SPDIF > output from the external DAC and connect it to an input of the > sound card whose SPDIF output goes to the DAC for conversion. > > Of course the sound card must be able to "slave" (synchronize) > its SPDIF output clock with the one coming from its SPDIF input. > > (BTW: are there other -perhaps easier and/or better- ways to do > what I would like to do?) > > > Thus, I would need a sound card which must be: > > * cabable of "bitperfect" (pass through) operation at CD standard > 16bit/44.1KHz (as well as, possibly, also at higher resolutions and > sample rates such as 16/48, 24/48, 24/96 and 24/192). > > * capable of "slaving" its SPDIF output clock to an external one, > such as the one reconstructed from its SPDIF input. > > ....last but not least, of course all of this must be done on Linux, > thus the sound card must be fully supported by ALSA! 8-) > > Well, yet another requirement is... that it should possibly (and > hopefully) not cost me a fortune! $-) > > > Thanks in advance for your attention. > > > Ciao e grazie, > Paolo. > > -- > Skype: Paolo.Saggese > http://borex.lngs.infn.it/saggese > You can still escape from the GATES of hell: Use Linux! > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Alsa-user mailing list > Als...@li... > https://lists.sourceforge.net/lists/listinfo/alsa-user > Extracting data digitally from CD is unrelated to soundcard. Perform web search for 'cdparanoia'. In fact, if you want reliable sound, first transfer data from _all_ your audio CDs to HD and then play it from there. Regarding soundcard and syncrhonization - M-Audio Revolution 7.1, and quite possibly M-Audio Revolution allow you to use external clock source. I have never used SPDIFF myself though. Regards, Sergei. |
|
From: Bill U. <un...@ph...> - 2007-11-21 22:48:59
|
On Wed, 21 Nov 2007, Sergei Steshenko wrote: > On Wed, 21 Nov 2007 20:33:39 +0100 > Paolo Saggese <pms...@ya...> wrote: > >> Hi everybody, >> >> I have a question for you guys. >> >> (I'm new to this list and hope this is the right place to ask >> and not an FAQ... I've tried to search to see if this question >> had been answered previously, but I have been not able to find >> what I was looking for). >> >> I'm thinking about using a dedicated (fanless, diskless, etc) >> PC as a digital source for my hi-end Hi-Fi system. >> >> Of course I plan to use the PC only to provide a "bitperfect" >> (exact copy of the original media, normally CD) digital stream >> to an external DAC. >> >> As you probably know better than me, the one major known problem >> when you strive for the highest possible quality in digital audio >> reproduction is jitter... and the best (if not only) way to really >> minimize it is to use a good, clean and stable clock close to the >> DAC chip, slaving everything else to that one. The clock jitter tends to be in the ppm range. This means that the frequency jitter is very low (if I believe the ppm then at the level of -120dB) which is completely inaudible. My cheap Transit card reliably gives me noise levels of the order of -90db below the level of the signal, again inaudible. >> >> Thus, what I would need to do would be to "slave" the sound card >> SPDIF output clock to the external DAC clock i.e. to make this one >> become the "master clock" for the whole digital audio stream. SPDIF is a digital output stream. Its "clock" is irelevant. The clock on the machine that converts that stream to analog is the important one. >> >> AFAIK, one possible way to do this is to set up a "fake" SPDIF >> output from the external DAC and connect it to an input of the >> sound card whose SPDIF output goes to the DAC for conversion. ???? >> >> Of course the sound card must be able to "slave" (synchronize) >> its SPDIF output clock with the one coming from its SPDIF input. >> >> (BTW: are there other -perhaps easier and/or better- ways to do >> what I would like to do?) >> >> >> Thus, I would need a sound card which must be: >> >> * cabable of "bitperfect" (pass through) operation at CD standard >> 16bit/44.1KHz (as well as, possibly, also at higher resolutions and >> sample rates such as 16/48, 24/48, 24/96 and 24/192). bit perfect digitial operation is trivial Almost anything can do that. Computers do it at GHz frequencies so 96KHZ is completely and totally trivial. The only crucial thing is the Digital to analog converter. >> >> * capable of "slaving" its SPDIF output clock to an external one, >> such as the one reconstructed from its SPDIF input. >> >> ....last but not least, of course all of this must be done on Linux, >> thus the sound card must be fully supported by ALSA! 8-) >> >> Well, yet another requirement is... that it should possibly (and >> hopefully) not cost me a fortune! $-) >> >> > Extracting data digitally from CD is unrelated to soundcard. > > Perform web search for 'cdparanoia'. > > In fact, if you want reliable sound, first transfer data from _all_ your > audio CDs to HD and then play it from there. > > Regarding soundcard and syncrhonization - M-Audio Revolution 7.1, and > quite possibly M-Audio Revolution allow you to use external clock > source. > > I have never used SPDIFF myself though. |
|
From: Paolo S. <pms...@ya...> - 2007-11-22 21:00:46
|
On Wednesday 21 November 2007 23:49, Bill Unruh wrote:
> The clock jitter tends to be in the ppm range. This means that the
> frequency jitter is very low (if I believe the ppm then at the level of -120dB)
> which is completely inaudible. My cheap Transit card reliably gives me noise
unfortunately, it's not as simple as it seems... in the real world,
with real (not ideal) DACs, jitter DOES matter, much more than you
can think. Even at ppm or ppb levels!
Nowadays there is plenty of documentation (and discussions) on the
subject. Try googling for e.g. "digital audio jitter" if you want to
know more (though the most interesting things are closed in the JAES
articles, sadly out of the reach of most people).
If the digital audio folks have had a lesser simplistic approach in
the first place, we would not have had to wait some 20 years before
gettin' a barely acceptabe sound out of a CD...
(what's worse is that, nevertheless, even now a good analogue system
can still sound MUCH better than any CD: it actually takes a good SACD
or DVD-A system to come close to the good old LP when it comes to the
real perceived audio quality!)
> SPDIF is a digital output stream. Its "clock" is irelevant. The clock on
> the machine that converts that stream to analog is the important one.
unfortunately, SPDIF is a synchronous interface... go figure. ;-)
Ciao,
Paolo.
--
Skype: Paolo.Saggese
http://borex.lngs.infn.it/saggese
You can still escape from the GATES of hell: Use Linux!
|
|
From: Rene H. <ren...@ke...> - 2007-11-22 21:18:39
|
On 22-11-07 22:00, Paolo Saggese wrote: > (what's worse is that, nevertheless, even now a good analogue system can > still sound MUCH better than any CD: it actually takes a good SACD or > DVD-A system to come close to the good old LP when it comes to the real > perceived audio quality!) If we're talking about a particularly thick LP with one song per side so as to allow for better dynamic range than the utterly pathetic 70 dB or so of a normal LP that is. And, ofcourse, a turntable with motor-control with absolute world-class stability. Particularly difficult one that... And, ofcourse, a world-class element with a completely new needle to turn the groove into electrical signals without distorting it. And, ofcourse, a world-class phono-amplifier to compensate for the cowardly dynamics of LP versus CD, certainly for anything but classic or jazz. And, ofcourse, a completely new record. And, ofcourse... well, you get the point. Other than that, I agree that CD is significantly worse than SACD ;-) Rene. |
|
From: Paolo S. <pms...@ya...> - 2007-11-22 18:44:41
|
Hi everybody,
first of all, I'd like to thank you all so much for the prompt
and many replies I've got... Wow, they came in faster then I was
able to read! :-)
(there's still someone claiming that Linux has no support?! :-)
On Wednesday 21 November 2007 22:19, Sergei Steshenko wrote:
> Extracting data digitally from CD is unrelated to soundcard.
>
> Perform web search for 'cdparanoia'.
mmmh... maybe I've not been clear enough with my question.
Of course I know how to extract CD data. :-)
BTW, in case someone could be interested (and sorry for
the off topic):
[OT on]
In fact, unfortunately cdparanoia is not (or perhaps no
longer) good enough to do "perfect" DAE.
Sadly I had to resort to a non-free, closed source windoze
software, namely "EAC" (at least it works well under Linux
using wine and is freeware/"cardware").
I guess the problems with cdparanoia may be due to the fact
that most (if not all) recent CD and DVD drives use to cache
audio data and that perhaps fools cdparanoia checks.
AFAIK in cdparanoia CVS there is/was a patch to disable
audio caching, but again AFAIK it does not work and was
never included in any release.
If you don't believe me, try to extract several times the
same CD on different drives and compare the resulting wav
files... (e.g. using "shntool cmp -s").
Even after disregarding the possible "byte-shift" which
is almost always present when extracting using different
drives, the extracted data may (and often will) differ.
IME it depends a lot on the drives used as well as on the
CD being extracted.
Sometimes you may find differences even when repeating the
extraction on the same drive, but IME it looks like -for a
given CD- the same drive tends to repeat the same errors in
the same place(s). Thus this problem is definitely easier
to detect comparing extractions from different drives.
I have also got differences in extracted data (using the
same drive) depending on the cdparanoia extraction "mode",
i.e. whether extracting the whole CD at once or track by
track in "batch" mode!! ( =:-O :|
I guess this may be very drive dependent, though I usually
use good (or at least so believed) drives for DAE, such as
Plextor ones (for the sake of curiosity I have also tried
with some cheap LGs, too).
In fact I used to trust cdparanoia, and had a bad surprise
when I have done this tests. :-(
Luckily, I did 'em before beginning the mass-extraction of
my rather large CD collection...
[OT off]
> In fact, if you want reliable sound, first transfer data from _all_ your
> audio CDs to HD and then play it from there.
that's exactly what I was planning to do... 8-)
(actually, the final plan is to use a remote RAID storage
while the local, dedicated machine should be both fanless
and diskless to be completely quiet).
> I have never used SPDIFF myself though.
well, I did... but my current SC is a cheap piece of ****, only
allows 48KHz output (aargh, resampling required... that's bad!)
and "of course" does not allow synchronization to any external
clock in any way.
Nevertheless, even in such a desperate and "lo-fi" setup, using
ALSA it didn't sound too bad when connected to my external DAC
(of course playing a CD from the dedicated transport the overall
quality is in another league... but I am confident that with a
proper setup I can even better the CD transport results).
Ciao,
Paolo.
--
Skype: Paolo.Saggese
http://borex.lngs.infn.it/saggese
You can still escape from the GATES of hell: Use Linux!
|
|
From: Rene H. <ren...@ke...> - 2007-11-22 20:53:46
|
On 22-11-07 19:44, Paolo Saggese wrote: > [OT on] > In fact, unfortunately cdparanoia is not (or perhaps no > longer) good enough to do "perfect" DAE. Over the course of ripping 735 CDs now to my harddrive with cdparanoia, I have found that all's well as long as cdparanoia leaves nothing but spaces in its output (the command line interface). Every time I went to the trouble of comparing I've gotten byte-identical rips in those cases. But anything else, and the next rip will most likely differ even though cdparanoia may have promised to correct something. Yes, it's probably dependent on drive. Generally Plextor drives as well. Problems are most frequent at the end of tracks for some reason. The EOT exclamation mark is quite often present... Rene. |
|
From: Sergei S. <ste...@li...> - 2007-11-21 21:24:36
|
On Wed, 21 Nov 2007 23:19:58 +0200 Sergei Steshenko <ste...@li...> wrote: > Regarding soundcard and syncrhonization - M-Audio Revolution 7.1, and > quite possibly M-Audio Revolution allow you to use external clock > source. > I meant "M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1 allow you to use external clock source. --Sergei. |
|
From: Clemens L. <cla...@go...> - 2007-11-22 10:08:45
|
Paolo Saggese wrote: > Of course I plan to use the PC only to provide a "bitperfect" > (exact copy of the original media, normally CD) digital stream > to an external DAC. > > As you probably know better than me, the one major known problem > when you strive for the highest possible quality in digital audio > reproduction is jitter... and the best (if not only) way to really > minimize it is to use a good, clean and stable clock close to the > DAC chip, slaving everything else to that one. > > Thus, what I would need to do would be to "slave" the sound card > SPDIF output clock to the external DAC clock i.e. to make this one > become the "master clock" for the whole digital audio stream. An SPDIF input _always_ derives its clock from its signal. Besides, the clock for the actual DAC has to be a multiple of the bit clock anyway, so there must be a PLL to derive the DAC's clock from the input signual, i.e., the amount of DAC clock jitter depends more on the PLL implementation than on the input signal quality. > AFAIK, one possible way to do this is to set up a "fake" SPDIF > output from the external DAC and connect it to an input of the > sound card whose SPDIF output goes to the DAC for conversion. > > Of course the sound card must be able to "slave" (synchronize) > its SPDIF output clock with the one coming from its SPDIF input. There are sound cards that have a word clock input (M-Audio Delta and others based on the ICE1712 chip), but a word clock is only used to force multiple devices to run at the exactly same frequency, i.e., to prevent their clocks from drifting apart. This clock can _not_ be used as a bit clock for digital signals. Even if it were possible, the extra SPDIF connection would introduce additional jitter, thus the SPDIF signal received by the DAC would have more jitter, compared to the case where the sound card uses its own clock for the SPDIF signal. In other words, the sound card's internal clock is of higher quality than any clock that has to be received from an external device. > Thus, I would need a sound card which must be: > > * cabable of "bitperfect" (pass through) operation at CD standard > 16bit/44.1KHz (as well as, possibly, also at higher resolutions and > sample rates such as 16/48, 24/48, 24/96 and 24/192). Any cheap CMI8738-based card can do 16 or 24 bits at 44.1 or 48 kHz. Cards based on the CMI8768 chip (and some later 8738 models) or based on the ICE1712 chip support 24/96. Regards, Clemens |
|
From: Rene H. <ren...@ke...> - 2007-11-22 10:44:50
|
On 21-11-07 20:33, Paolo Saggese wrote: > Thus, I would need a sound card which must be: > > * cabable of "bitperfect" (pass through) operation at CD standard > 16bit/44.1KHz (as well as, possibly, also at higher resolutions and > sample rates such as 16/48, 24/48, 24/96 and 24/192). 16/44.1 is basic enough that any card will be able to operate its S/PDIF output at it. Traditionally, together with 16/32 and 16/48, and probably 20-bits instead of 16 possible. "Basic S/PDIF" is limited to those 20-bits with 24-bit as only an option in the standard but these days any card should do 24-bit fine. > * capable of "slaving" its SPDIF output clock to an external one, > such as the one reconstructed from its SPDIF input. I suppose your external DAC has no actual WordClock (BNC connection) output? If it does, you have a range of professional products available (and semipro in for example the TerraTec EWS 88 D/MT coupled with a TerraTec ClockWork module). I know the M-Audio Audiophile 24/96 and Audiophile 192 can sync the master clock with their S/PDIF in and both are great cards, also analogue, and not expensive. The 24/96 you'll even be able to locate 2nd hand by now for +/- EUR 50 or so. Check eBay -- I see them starting at EUR 0,99 ;-) As far as I'm aware, both card function well on Linux. I have neither card myself, so ask around a bit for any possible specific trouble though. Rene. |
|
From: Rene H. <ren...@ke...> - 2007-11-22 10:51:40
|
On 22-11-07 11:39, Rene Herman wrote: > On 21-11-07 20:33, Paolo Saggese wrote: > >> Thus, I would need a sound card which must be: >> >> * cabable of "bitperfect" (pass through) operation at CD standard >> 16bit/44.1KHz (as well as, possibly, also at higher resolutions and >> sample rates such as 16/48, 24/48, 24/96 and 24/192). > > 16/44.1 is basic enough that any card will be able to operate its S/PDIF > output at it. Traditionally, together with 16/32 and 16/48, and probably > 20-bits instead of 16 possible. > > "Basic S/PDIF" is limited to those 20-bits with 24-bit as only an option in > the standard but these days any card should do 24-bit fine. > >> * capable of "slaving" its SPDIF output clock to an external one, >> such as the one reconstructed from its SPDIF input. > > I suppose your external DAC has no actual WordClock (BNC connection) output? > If it does, you have a range of professional products available (and semipro > in for example the TerraTec EWS 88 D/MT coupled with a TerraTec ClockWork > module). > > I know the M-Audio Audiophile 24/96 and Audiophile 192 can sync the master > clock with their S/PDIF in and both are great cards, also analogue, and not > expensive. The 24/96 you'll even be able to locate 2nd hand by now for +/- > EUR 50 or so. Check eBay -- I see them starting at EUR 0,99 ;-) > > As far as I'm aware, both card function well on Linux. I have neither card > myself, so ask around a bit for any possible specific trouble though. Oh, and ofcourse, the 2496 does 96 kHz and the 192... 192 kHZ (that is, the 2496 is limited to somewhere around 100 kHz for its masterclock and the 192 to somewhere around 200 kHz). Rene. |
|
From: Clemens L. <cla...@go...> - 2007-11-22 14:24:41
|
Rene Herman wrote: > On 21-11-07 20:33, Paolo Saggese wrote: >> Thus, I would need a sound card which must be: >> >> * cabable of "bitperfect" (pass through) operation at CD standard >> 16bit/44.1KHz (as well as, possibly, also at higher resolutions and >> sample rates such as 16/48, 24/48, 24/96 and 24/192). > > 16/44.1 is basic enough that any card will be able to operate its S/PDIF > output at it. Most cards support only 48 kHz output. This is either a consequence of being a wavetable card that has to mix all data together at a common frequency, or the low-budget design of the AC'97 and HDA standards. Regards, Clemens |
|
From: Rene H. <ren...@ke...> - 2007-11-22 14:58:22
|
On 22-11-07 15:25, Clemens Ladisch wrote: > Rene Herman wrote: >> On 21-11-07 20:33, Paolo Saggese wrote: >>> Thus, I would need a sound card which must be: >>> >>> * cabable of "bitperfect" (pass through) operation at CD standard >>> 16bit/44.1KHz (as well as, possibly, also at higher resolutions and >>> sample rates such as 16/48, 24/48, 24/96 and 24/192). >> 16/44.1 is basic enough that any card will be able to operate its S/PDIF >> output at it. > > Most cards support only 48 kHz output. No. Not in the semi-pro range we were talking about. You'd have a hard time finding _any_ that only do 48 kHz there. > This is either a consequence of being a wavetable card that has to mix > all data together at a common frequency, or the low-budget design of the > AC'97 and HDA standards. Ofcourse, in the horse-shit range, you will. Rene. |
|
From: Paolo S. <pms...@ya...> - 2007-11-22 19:38:49
|
On Thursday 22 November 2007 11:39, Rene Herman wrote: > I suppose your external DAC has no actual WordClock (BNC connection) outp= ut?=20 indeed, it does not. As it does not have an SPDIF output, either... but I'm a DIY guy (with an EE degree...) and can add whatever output I need or even build a new dedicated DAC for the task myself, should I prefer not to modify my valuable and expensive commercial DAC. 8-) > If it does, you have a range of professional products available (and semi= pro=20 > in for example the TerraTec EWS 88 D/MT coupled with a TerraTec ClockWork= =20 > module). interesting...=20 BTW, what about a TerraTec PHASE 22 ?=20 mmmh, it's not even in the Matrix... :-( > I know the M-Audio Audiophile 24/96 and Audiophile 192 can sync the maste= r=20 > clock with their S/PDIF in and both are great cards, also analogue, and n= ot=20 > expensive. The 24/96 you'll even be able to locate 2nd hand by now for +/= =2D=20 > EUR 50 or so. Check eBay -- I see them starting at EUR 0,99 ;-) yep, I've heard of those ones, and like 'em. Though I may probably prefer TOSlink to avoid ground loops and noise=20 transmission...=20 But can they do what I need using ALSA?=20 Perhaps I'm wrong but -if I remember correctly-, I've read somewhere that=20 their support is still only partial, and one can't do what I need on Linux. Am I wrong? (doh, I hope so! :-) Another card I have seen in the "semi-pro" arena which seems to be almost perfect (both RCA and TOSlink I/O, support for up to 24/192, etc.) is the=20 "ESI Juli@", http://www.esi-pro.com/viewProduct.php?pid=3D43 =46rom the ALSA matrix, it looks like it's based on the same chipset of the M-Audio 192. Would this be also suitable and worth the extra money? How well is it supported by ALSA? Other options? Ciao e grazie, Paolo. =2D- Skype: Paolo.Saggese http://borex.lngs.infn.it/saggese You can still escape from the GATES of hell: Use Linux! |
|
From: Rene H. <ren...@ke...> - 2007-11-22 20:37:33
|
On 22-11-07 20:38, Paolo Saggese wrote: > On Thursday 22 November 2007 11:39, Rene Herman wrote: > >> I suppose your external DAC has no actual WordClock (BNC connection) output? > > indeed, it does not. As it does not have an SPDIF output, either... > > but I'm a DIY guy (with an EE degree...) and can add whatever output > I need or even build a new dedicated DAC for the task myself, should > I prefer not to modify my valuable and expensive commercial DAC. 8-) It probably doesn't matter all that much. And anyways, you might need to take Clemens' reply into account which seemed to say you're basically out of luck anyway as any S/PDIF in sync is not going to help any. I dunno... >> If it does, you have a range of professional products available (and semipro >> in for example the TerraTec EWS 88 D/MT coupled with a TerraTec ClockWork >> module). > > interesting... > > BTW, what about a TerraTec PHASE 22 ? > > mmmh, it's not even in the Matrix... :-( It is supported: sound/pci/ice1712/phase.c. However, near the top: "Digital receiver: CS8414-CS (not supported in this release)" so it might not be the best choice. I do believe it otherwise supports the same kind of external master clock sync that the Audiophiles do. >> I know the M-Audio Audiophile 24/96 and Audiophile 192 can sync the master >> clock with their S/PDIF in and both are great cards, also analogue, and not >> expensive. The 24/96 you'll even be able to locate 2nd hand by now for +/- >> EUR 50 or so. Check eBay -- I see them starting at EUR 0,99 ;-) > > yep, I've heard of those ones, and like 'em. > > Though I may probably prefer TOSlink to avoid ground loops and noise > transmission... Not too fond of TOSlink myself but on short distances and when no particular subtlety of cable is desired I guess it matters not one bit. One bit. Ha. > But can they do what I need using ALSA? > > Perhaps I'm wrong but -if I remember correctly-, I've read somewhere that > their support is still only partial, and one can't do what I need on Linux. > > Am I wrong? (doh, I hope so! :-) As said, I have neither card myself, so I'll not say anything definite. If I look at the driver, it seems that digital in is supported at least. Just saw you get a quality reply from Vladimir. If you're expanding into external cards, many more options. > Another card I have seen in the "semi-pro" arena which seems to be almost > perfect (both RCA and TOSlink I/O, support for up to 24/192, etc.) is the > "ESI Juli@", http://www.esi-pro.com/viewProduct.php?pid=43 > >>From the ALSA matrix, it looks like it's based on the same chipset of the > M-Audio 192. > > Would this be also suitable and worth the extra money? > > How well is it supported by ALSA? Sorry, no idea. Rene. |
|
From: Rene H. <ren...@ke...> - 2007-11-22 11:12:37
|
On 22-11-07 11:10, Clemens Ladisch wrote: > Paolo Saggese wrote: >> Of course I plan to use the PC only to provide a "bitperfect" >> (exact copy of the original media, normally CD) digital stream >> to an external DAC. >> >> As you probably know better than me, the one major known problem >> when you strive for the highest possible quality in digital audio >> reproduction is jitter... and the best (if not only) way to really >> minimize it is to use a good, clean and stable clock close to the >> DAC chip, slaving everything else to that one. >> >> Thus, what I would need to do would be to "slave" the sound card >> SPDIF output clock to the external DAC clock i.e. to make this one >> become the "master clock" for the whole digital audio stream. > > An SPDIF input _always_ derives its clock from its signal. > > Besides, the clock for the actual DAC has to be a multiple of the bit > clock anyway, so there must be a PLL to derive the DAC's clock from the > input signual, i.e., the amount of DAC clock jitter depends more on the > PLL implementation than on the input signal quality. I don't quite understand your reply. He isn't asking to sync the S/PDIF in to the signal, he's asking to sync the S/PDIFF _out_ to the S/PDIF in. Ie, to have a dummy S/PDIF connection DAC --> Card S/PDIF In that exists only to supply a clock to the card, to which Card S/PDIF Out --> DAC will then be synced. This is a valid wish, isn't it? And at least the M-Audio Audiphile cards can sync themselves to the S/PDIF In clock. Rene. |
|
From: Clemens L. <cla...@go...> - 2007-11-22 14:23:47
|
Rene Herman wrote: > On 22-11-07 11:10, Clemens Ladisch wrote: >> An SPDIF input _always_ derives its clock from its signal. >> >> Besides, the clock for the actual DAC has to be a multiple of the bit >> clock anyway, so there must be a PLL to derive the DAC's clock from the >> input signual, i.e., the amount of DAC clock jitter depends more on the >> PLL implementation than on the input signal quality. > > I don't quite understand your reply. He isn't asking to sync the S/PDIF in > to the signal, he's asking to sync the S/PDIFF _out_ to the S/PDIF in. > > Ie, to have a dummy S/PDIF connection > > DAC --> Card S/PDIF In > > that exists only to supply a clock to the card, to which > > Card S/PDIF Out --> DAC > > will then be synced. This is a valid wish, isn't it? And at least the > M-Audio Audiphile cards can sync themselves to the S/PDIF In clock. They can, but the word clock is used only as a sample clock. It will not reduce the amount of jitter of the card's S/PDIF output. Regards, Clemens |
|
From: Sergei S. <ste...@li...> - 2007-11-22 19:49:58
|
Just a random thought on jitter - if you like music recorded originally before the digital era - don't bother. I.e. analog tape recorder wow and flutter ( http://en.wikipedia.org/wiki/Wow_and_flutter ) is much higher than digital jitter. And that wow and flutter is already in the now digital signal. --Sergei. |
|
From: Rene H. <ren...@ke...> - 2007-11-22 20:02:21
|
On 22-11-07 20:49, Sergei Steshenko wrote: > Just a random thought on jitter - if you like music recorded originally > before the digital era - don't bother. > > I.e. analog tape recorder wow and flutter > > ( http://en.wikipedia.org/wiki/Wow_and_flutter ) > > is much higher than digital jitter. > > And that wow and flutter is already in the now digital signal. Exactly. And us purists like our wow and flutter as original as possible! Rene. |
|
From: Vladimir M. <mosgalin@VM10124.spb.edu> - 2007-11-22 20:15:11
|
Hi Sergei Steshenko! On 2007.11.21 at 23:24:35 +0200, Sergei Steshenko wrote next: > > > Regarding soundcard and syncrhonization - M-Audio Revolution 7.1, and > > quite possibly M-Audio Revolution allow you to use external clock > > source. > > > > I meant "M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1 > allow you to use external clock source. In theory, yes. In practice, I wasn't able to make my M-Audio Audiophile USB get clock from external source. Well it kinda works, but at some point distortions appear, and one must force clock resync or something like that by turning card off and on. I wanted to create setup similar to this, and one of the things I learned that in order to reduce jitter, you'd want to have power as clean as possible. On-board soundchips produce lowest quality signal, pci/pcie boards have much better filtering and produce better signal, but if you want something better, you have to use card which isn't powered by PSU of your PC, and doesn't suffer from problems of its signal. So if you want best digital audio, you probably should look among external cards (usb/firewire) which aren't bus powered, and use external AC adapter. Interface doesn't matter as long as card doesn't get power from it, so choose most compatible card. I picked Audiophile USB, which supports up to 24bit/96khz (though most likely you'll use it in 24bit/48khz mode). It also supports almost any sound rate without resampling, i.e. you can drive your DAC at 44100 in bit-exact mode and get highest quality possible signal, and resampling would happen only in DAC or won't happen at all (internal upsampling mode is recommended for most modern DACs though). As about quality, all I can say is that digital output from this card sounds much better than digital output from Audigy 2 ZS (both in regular or through p16v path, presumable working in bit-exact mode in the last case). As about external clock.. Audiophile usb _supports_ syncing from spdif in, but when I tried to connecting spdif out of my DAC to spdif in on card and enabled that spdif in order to synchronize, I got just noise (or very distored signal) from spdif out on card. I could get it working when spdif out on dac was set to "no signal" or signal from some of the unsed inputs, at some conditions I was even able to get it working with spdif out set to output signal from spdif in to which sound card was connected, but this configuration wasn't very stable and after a few hours of usage the distortions could start again. Switching the source of spdif out when it was used as clock source for the card also could produce some very strange results. Due to strangeness of this recursive scheme (imagine signal path: card output -> dac input -> dac output -> card input -> clocks from CARD's original output are used to drive "next" card output?) and me not understanding how and at what point re-clockings occur, but most important: completely failing to hear any advantages of this setup comparing to simplest path, I stopped experimenting with it. Now I just enjoy music and I'm quite satisfied with it ;) -- Vladimir |
|
From: Paolo S. <pms...@ya...> - 2007-11-23 15:20:35
|
On Thursday 22 November 2007 21:01, Vladimir Mosgalin wrote:
> > I meant "M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1
> > allow you to use external clock source.
>
> In theory, yes. In practice, I wasn't able to make my M-Audio Audiophile
> USB get clock from external source. Well it kinda works, but at some
> point distortions appear, and one must force clock resync or something
> like that by turning card off and on.
mmh... I guess the USB interface may be the problem source here.
Though USB could be operated in asynchronous mode, AFAIK most (all?)
of the current USB audio devices operate it in synchronous mode, with
the clock provided from the PC.
Of course, if the USB interface (from which the digital audio stream
is coming) is operated synchronously with source-based clock, then the
sound card MUST be somehow in sync with that stream clock...
The only way to "loosen" the sync with that clock and try to sync to
another one is to do some sort of reclocking, but unfortunately that
is not gonna work quite well.
In particular when the "upstream" clock is that of an USB connection
which - apart from being usually "dirt" and not so stable - is at odd
rates with respect to audio clocks.
The only real solution would be a sound card which connect to the PC
through an _asynchronous_ interface (be it USB, firewire, Ethernet,
HDMI or whatever else, as long as it's fast enough and asynchronous).
BTW, do you know if any such device exists?
> I wanted to create setup similar to this, and one of the things I learned that
> in order to reduce jitter, you'd want to have power as clean as possible.
indeed, absolutely.
> On-board soundchips produce lowest quality signal, pci/pcie boards have much
> better filtering and produce better signal, but if you want something better,
> you have to use card which isn't powered by PSU of your PC, and doesn't suffer
> from problems of its signal. So if you want best digital audio, you probably
> should look among external cards (usb/firewire) which aren't bus powered, and
> use external AC adapter. Interface doesn't matter as long as card doesn't get
> power from it, so choose most compatible card.
well, I have to disagree here. Interface would/should not matter
ONLY if it is asynchronous.
If the interface is synchronous, then the card has to lock itself
to the interface clock and, unfortunately, IMHO/IME no reclocking
and/or resampling will ever be able to really clean up the mess.
For the USB, the standard bus clock frequency makes things even
worse... don't know about firewire.
Ciao,
Paolo.
--
Skype: Paolo.Saggese
http://borex.lngs.infn.it/saggese
You can still escape from the GATES of hell: Use Linux!
|
|
From: Sergei S. <ste...@li...> - 2007-11-22 21:11:23
|
On Thu, 22 Nov 2007 22:00:29 +0100 Paolo Saggese <pms...@ya...> wrote: [snip] > > If the digital audio folks have had a lesser simplistic approach in > the first place, we would not have had to wait some 20 years before > gettin' a barely acceptabe sound out of a CD... > [snip] In a sense, one _can not_ have acceptabe sound out of a CD. That's because of (44100 / 2) is very close to 20kHz, and antialiasing LPF has insanely steep slope, causing huge phase distortions, thus severely distorting signal envelopes and creating preechos. With 48kHz the problem is much less severe, and that's why professional equipment uses at least 48kHz sample rate. Think of analog era tape recorders - the decent ones had bias frequency of 180-200kHz - bias frequency is a raw moral equivalent of sample rate. So, the true improvement comes with higher sample rate - that's why SACD. Whatever ... --Sergei. |
|
From: Darrell B. <drb...@ya...> - 2007-11-23 01:54:18
|
I have the M-Audio Audiophile 24/96. I purchased the card a few years ago on the basis of recommendations on this mailing list. I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. The card does things in a non-standard way. For example, you need to use Envy24Control as the mixer. Other mixers don't work, or don't work properly. Volume controls in some apps seem to work, others don't. If you search the archives of the list you will find that the only people who seem to have this card working well are alsa developers or those who understand every last line of source code in the alsa project and know all about the sound daemons of linux, such as jackd, artsd, and oss. You will find many people like myself that have asked for help and example configuration files and get no responses. Don't figure on installing this card, running alsaconf, and having a sound system that fully exploits the capabilities of this card, it won't happen. While sound in linux has come a long way in the decade since I have been using linux, it is no where near plug and play. Be prepared to learn all about the intricacies of alsa, and the sound applications. Note that many of these sound applications don't seem to get along well together. Perhaps at some point in the future, you will be able to install an M-Audio Audiophile 24/96 card, and some super sound configuration application will detect and configure it so that you can record a track in Audacity, have KDE play a warning beep, and watch the whole thing in Baudline, all at once. For now that is my dream that seems to be out of reach. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. Darrell On Thursday 22 November 2007 02:39, Rene Herman wrote: > I know the M-Audio Audiophile 24/96 and Audiophile 192 can sync the master > clock with their S/PDIF in and both are great cards, also analogue, and not > expensive. The 24/96 you'll even be able to locate 2nd hand by now for +/- > EUR 50 or so. Check eBay -- I see them starting at EUR 0,99 ;-) > > As far as I'm aware, both card function well on Linux. I have neither card > myself, so ask around a bit for any possible specific trouble though. > > Rene. |
|
From: Vladimir M. <mosgalin@VM10124.spb.edu> - 2007-11-23 08:45:20
|
Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: > I have never been happy with this card. While it works okay for playing basic > sound, getting it to do anything more sophisticated is pure black magic. For > example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. > Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of > the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? -- Vladimir |
|
From: Gene H. <gen...@ve...> - 2007-11-23 17:57:50
|
On Friday 23 November 2007, Vladimir Mosgalin wrote: >Hi Darrell Bellerive! > > On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: >> I have never been happy with this card. While it works okay for playing >> basic sound, getting it to do anything more sophisticated is pure black >> magic. For example, I have never gotten full duplex to work. > >Well now, when pulseaudio era came, hopefully it's not true anymore. > >> Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth >> of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. > >How does that matter when all is needed is digital output? > >Anyway, in analog mode, are you sure there is no option to switch off >bandwidth filter? No one in their right mind would want to do that as the aliasing would drive you up a wall. The other delay distortions the filter might give are 100's of times more tolerable to listen to. -- Cheers, Gene "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Pie are not square. Pie are round. Cornbread are square. |
Thanks for helping keep SourceForge clean.
X