wordclock and spdif?
wordclock and spdif?
as the final hardware in my studio chain , i have a api a2d , i record the result into my daw, giving me the benefit of a good converter. My problem is i had been running this via spdif, now i use wordclock, and i find scope either sees the clock from spdif of word, the problem is ......i record into my daw from spdif, so how could i use the wordclock, and the audio from spdif?, instead of using the clock from spdif?, it won't give me audio if i don't use spdif, there is a conflict of clocks i guess within scope eg..its one or the other
right
thats why immanual, i may have to get a wordclock that outputs spdif along with bnc, cause scope won't let you choose both spdif and ext bnc, it will only let you choose one. Audio comes thru spdif and clock, too bad just the audio wouldn't come thru
you can only have one sync source! you misunderstand how this works.
my pm to you:
i don't understand the problem....
if you have an external clock, connect a bnc to the scope syncplate and another to the api. set scope to slave to bnc and the api to external sync. done. go ahead and connect sp/dif and data will flow beautifully.
if you don't have an external clock, but have a syncplate, set scope to master, connect a bnc from the syncplate to the api, set the api to external sync(this may be automatic) and connect sp/dif, it'll work. OR set scope to slave to sp/dif and don't connect bnc.
sync is about clock timing for the playback of the samples. sp/dif is a data path for the samples. sync can be either along the data path or via wordclock bnc.
i hope this helps.....
-g

my pm to you:
i don't understand the problem....
if you have an external clock, connect a bnc to the scope syncplate and another to the api. set scope to slave to bnc and the api to external sync. done. go ahead and connect sp/dif and data will flow beautifully.
if you don't have an external clock, but have a syncplate, set scope to master, connect a bnc from the syncplate to the api, set the api to external sync(this may be automatic) and connect sp/dif, it'll work. OR set scope to slave to sp/dif and don't connect bnc.
sync is about clock timing for the playback of the samples. sp/dif is a data path for the samples. sync can be either along the data path or via wordclock bnc.
i hope this helps.....
-g
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yes garyb that's true and jeff uses (i just read an old mail from him) : Lucid GENx192 Ultra
http://www.lucidaudio.com/index.php?Show=216&Show1=217
it seems it doesn't have superclock...anyway,if he has another one now ,my chain is right.preamp had to be set always as master.
eric
from lucid website :
Can the GENx6 / GENx6-96 / GENx192 generate Word Clock or Superclock synchronized from an AES signal?
Yes, the GENx6 / GENx6-96 can synthesize a Word Clock or Superclock output derived from the Word Clock, Superclock or AES input.
The GENx192 can only generate a Word Clock signal from an AES signal. Superclock was not included on the GENx192 since recent Digidesign gear does not have Superclock.
http://www.lucidaudio.com/index.php?Show=216&Show1=217
it seems it doesn't have superclock...anyway,if he has another one now ,my chain is right.preamp had to be set always as master.
eric
from lucid website :
Can the GENx6 / GENx6-96 / GENx192 generate Word Clock or Superclock synchronized from an AES signal?
Yes, the GENx6 / GENx6-96 can synthesize a Word Clock or Superclock output derived from the Word Clock, Superclock or AES input.
The GENx192 can only generate a Word Clock signal from an AES signal. Superclock was not included on the GENx192 since recent Digidesign gear does not have Superclock.
thnks guys
hopefully i can clear this up, ..i have rented a genx 696, it only has clock and superclock, ..the clock eric is referring to (genx192) has spdif out, i guess my question out there to anybody is. ....if scope in samplerate settings is set to ext bnc, and everything is slave ..how come i cannot hear any audio from spdif , being the last thing in my chain from api, unless i put that connection in scope to spdif in samplerate settings,
becuase something's not connected properly. the outputs from the genx must be connected to both scope and the api's bnc connectors. the output to scope must be wordclock. the output to the api must be superclock and it must show a lock to ext(an led) on the front of the unit. set scope to slave to ext bnc. verify the samplerate in the scope settings. connect the s/dif output of the api to the s/pdif input of scope. it MUST work.
eric is right, of course. if you do all this and it doesn't work you can alway just clock to the api via s/pdif. it will be very good.
eric is right, of course. if you do all this and it doesn't work you can alway just clock to the api via s/pdif. it will be very good.

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api needs 3vpp and 20-26khz from superclock to work
the manufacturers data for api a2d has to have a 20-26mhz on bnc and minimum 3vpp for the unit to turn on, for some reason tech support for api said its not working because the generator might not be right for it. , they recommended a mutek i clock, ..but said the api clock is one of the best crystal clocks you can get.. ...In other words its on the downstream, that final stage of conversion that everything tightens up on , and no sample drift is possible if you use your converter clock, ..where conversion is compromised is with and ext clock in competition with the converter clock, ..Api support then told me that the ext clock does help on the d to a side , and that is where someone will hear , say monitoring, , WoW! what a difference when i buy a wordclock, they are just listening to their monitoring environment, not the audio they are recording within their DAW, ...so i do not need a wordclock, api support saved me money,
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hi all,
http://www.planetz.com/phpBB2/viewtopic ... 607#172607
a way to understand how some customer's support works and if wordclock is only a myth
eric
http://www.planetz.com/phpBB2/viewtopic ... 607#172607
a way to understand how some customer's support works and if wordclock is only a myth

eric
well, Eric's story is about clock for sync (nice story btw), but I can't figure out Wired's bottomline - so here's my attempt.
restricted to audio quality a clock does indeed only matter at the conversion stage,
but that applies to input and output
for simplicity there's one unit generating precise wordclock for both conversion stages, as it's very unlikely that a studio never handles analog input
an electronic musician who generates all sounds within his DAW and only uses pre-recorded stuff (samples etc), gets along with only the output converter clocked properly. A high quality unit might be capable of doing this internally - it depends on the specific circuit design.
for all others the most 'precious' moment is when the analog signal enters the converter and is 'time-marked' (so to say) by the process.
a sluggish clock signal that is a bit in advance sometimes and a bit late a few milliseconds later, which drifts - or in other words 'jitters' - simply puts a wrong 'time-mark' on the 'sampled' value.
to refer to the infamous sine graph: a 'rising' signal (in time, not pitch) slightly before the top of the curve is marked as 'falling' if the clock is a bit late.
if that sample is played back later, it's always interpreted as 'beyond peak of the curve', so the waveform gets distorted, pushing up the THD + noise figure.
On simple designs this can compete (numerically) with a well driven tube amp stage, but it doesn't even remotely sound as good...
I sometimes get the impression that the word 'jitter' is used to describe sync problems and all kind of droputs (like the clicks Eric mentions) on DAW processing.
cheers, Tom
btw Eric, I would notice those clicks - does it make me an audio engineer ?
in Moloko's 'Sing it Back' there's one by a bad cut right before minute 2
restricted to audio quality a clock does indeed only matter at the conversion stage,
but that applies to input and output
for simplicity there's one unit generating precise wordclock for both conversion stages, as it's very unlikely that a studio never handles analog input

an electronic musician who generates all sounds within his DAW and only uses pre-recorded stuff (samples etc), gets along with only the output converter clocked properly. A high quality unit might be capable of doing this internally - it depends on the specific circuit design.
for all others the most 'precious' moment is when the analog signal enters the converter and is 'time-marked' (so to say) by the process.
a sluggish clock signal that is a bit in advance sometimes and a bit late a few milliseconds later, which drifts - or in other words 'jitters' - simply puts a wrong 'time-mark' on the 'sampled' value.
to refer to the infamous sine graph: a 'rising' signal (in time, not pitch) slightly before the top of the curve is marked as 'falling' if the clock is a bit late.
if that sample is played back later, it's always interpreted as 'beyond peak of the curve', so the waveform gets distorted, pushing up the THD + noise figure.
On simple designs this can compete (numerically) with a well driven tube amp stage, but it doesn't even remotely sound as good...

I sometimes get the impression that the word 'jitter' is used to describe sync problems and all kind of droputs (like the clicks Eric mentions) on DAW processing.
cheers, Tom
btw Eric, I would notice those clicks - does it make me an audio engineer ?
in Moloko's 'Sing it Back' there's one by a bad cut right before minute 2

so much of that stuff...
as eric pointed out in his other post, if you have a number of devices that all need to use the same clock, for example, several different ad/das a dat machine and the computer(
my studio), wordclock is good because everything has the same, good clock that way, no sync problems. for just recording one mic pre, however, the api has a good clock so why not use it?
wordclock is not magic, a clock is a clock if it's accurate. i still think the techs explanation did little to clear up the mystery. if the api's clock is so good, why not use it for the mixdown clock as well? wouldn't that give the benefit on playback that the tech says wordclock is for?
in all actuality, it's possible that the apis clock could sound better than the card's. try it. it doesn't matter which you use as long as it's the same one for all devices.
as eric pointed out in his other post, if you have a number of devices that all need to use the same clock, for example, several different ad/das a dat machine and the computer(

wordclock is not magic, a clock is a clock if it's accurate. i still think the techs explanation did little to clear up the mystery. if the api's clock is so good, why not use it for the mixdown clock as well? wouldn't that give the benefit on playback that the tech says wordclock is for?
in all actuality, it's possible that the apis clock could sound better than the card's. try it. it doesn't matter which you use as long as it's the same one for all devices.
and it doesn't matter on which type of cable it's transmitted - a BNC cable doesn't make it more precise...garyb wrote:...wordclock is not magic, a clock is a clock if it's accurate.
...
in all actuality, it's possible that the apis clock could sound better than the card's. try it. it doesn't matter which you use as long as it's the same one for all devices.
cheers, Tom
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like for andred,drop out and clicks are not the same you are talking,mine (or about clock) are error on a/d during x sample..but,like him,you really don't know my job,so...no more polemics for me..astroman wrote: restricted to audio quality a clock does indeed only matter at the conversion stage,
but that applies to input and output
for simplicity there's one unit generating precise wordclock for both conversion stages, as it's very unlikely that a studio never handles analog input
cheers, Tom
btw Eric, I would notice those clicks - does it make me an audio engineer ?
in Moloko's 'Sing it Back' there's one by a bad cut right before minute 2
80% of pro studios are made like minet's very unlikely that a studio never handles analog input
signal to analog ssl or neve...eq ,fx,are on analog signal and routed to a/d digi or apogee,to protools..output protools routed to the d/a to go to ssl.somme of studio like my friend is all numerique,oxford to pt via aes...but i don't like this.i prefere ssl a/d d/a and conv a/d for mix ,or mixing on analo.
eric
Eric, there's no polemics at all - whatever you call dropout or click in that session isn't my cup of tea.
Wired's concerns are about audio quality only, and I have correctly explained why a clock has to be precise for that purpose
both your quotes from my previous post are out of context (and don't even have a relation to what's discussed here), why ?
cheers, Tom
Wired's concerns are about audio quality only, and I have correctly explained why a clock has to be precise for that purpose
both your quotes from my previous post are out of context (and don't even have a relation to what's discussed here), why ?
cheers, Tom