SPDIF with slave clocking

  • Not possible.


    Please be aware that there is no single digital amp on the market that can be slaved, from what I know.
    Slave capabilities is more an exception than a rule, especially for complex music instruments.


    IMHO audio interface manufacturers should provide products with multiple spdif ins and outs, supported with sample rate converters. This would vastly reduce the complexity of cabling and syncing. But either there is no demand, or they cannot see the demand.


    In contradiction to the common sense todays sampling rate converters provide a perfect audio quality.

  • Thank you. While certainly not good news, it's nice to have a real, detailed answer.

    "But dignity is difficult to maintain
    stamina requires constant upkeep
    repetition is boring
    and you pay for grace."

  • Hi,


    Just wanted to bump this as I've just started using SPDIF to record digitally. It just seems like a hassle to change the master/slave relationship whenever I use the Kemper.


    Blame your interface for that.
    It is usually a good habit of a sync slave to fall back on a preselected state, rather than stop working.

  • I am afraid that your GT-8 does not provide slave clocking. It simply lacks the required spdif input or wordclock input to receive an external master signal.
    Did I overlook something?


    Hey Thanks for the reply Cristoph,


    I meant the GT-Pro, not the GT-8 (I have both of them). But it's my mistake I suppose ... my GT-Pro did work with SPDIF (didnt use it long enough since it has USB audio) when I tried it out first.


    But now that I think of it - the kemper also works with my MOTU interface, if I dont slave the interface to the Kemper. The first time I used the SPDIF interface - this just worked, and I didnt think it was such a big deal (just thought this worked out of the box), slave clocking i.e.. But as time went by - I realized that this was probably one of those magical things, which would work initially, and then the clocks would slowly drift apart or something, and then it would not work. I did see issues once in a while when I turned it on, and I heard some really strange sound artifacts. This is when I started slaving the audio interface to the Kemper.


    Anyway, thanks for clearing that up CK! I'll just have to modify my workflow a little to get the job done. Like I said, I didnt know that SPDIF slave clocking was such a big deal. Sometimes we assume too much I guess ;).



  • What I notice when connected via SPDIF and the Kemper is NOT set to be the master is that audio goes through fine as such, but there are ticks or clicks every once in a while - I guess when the two clock sources are out of sync ever so slightly.


    (BTW, must be strange for Christoph to see himself being referred to as an audio device in all these threads! :-))

  • Sometimes I really think, CK should remove even S/PDIF in the next incarnation of the Profiler because he clearly has no idea at all about digital audio in professional studio environments.

    Quote from ckemper

    Slave capabilities is more an exception than a rule, especially for complex music instruments


    This comment shows nothing but his ignorance regarding the digital sync problem. Of course it's easy to just request studio owners to get racks full of async SRCs but in reality things work differently. Professional digital outboard gear has Word Clock and/or SDI sync to a studio master clock. From effects (like Lexicon PCM 96) to all kinds of encoders/decoders (Dolby Enocders/Decoders) to instruments (Korg Kronos) and audio interfaces (Focusrite Saffire 56, UAD Apollo, RME Fireface, ...) and audio embedders/de-embedders in broadcast audio and even digital mixer consoles (Lawo mc², Studer Vista, ...).


    I like the Profiler for sure, but the digital I/O is far from being professional, no matter how you look at it.


    Or to put it in a different way:
    In a complex environment you can't have everyone be a chief (master). This will bring nothing but trouble. You need one chief (master) and lots of warriors (slaves).

  • Sometimes I really think, CK should remove even S/PDIF in the next incarnation of the Profiler because he clearly has no idea at all about digital audio in professional studio environments.


    Why not quote the whole paragraph to give the statement its context?

    Quote

    Please be aware that there is no single digital amp on the market that can be slaved, from what I know.
    Slave capabilities is more an exception than a rule, especially for complex music instruments.

  • Why not quote the whole paragraph to give the statement its context?


    Because there was no such thing as "context". First he writes about digital amps and then about music instruments. A digital amp isn't an instrument, it's an effects unit.


    I know it's a difficult topic to discuss ... and mostly it's not even worth to discuss because it is as it is and no change is expected.
    But when he starts to claim that we should reach out for a truck load of async SRCs to setup a digital studio environment, then I really have to laugh hard. It is common practice in a professional, digital studio environment that there's a studio master clock and all connected gear is synced to the master clock. Of course it's possible to use an (externally clocked) async SRC here and there. But not because this is the best solution but because some devices (many actually) don't meet the professional standards.


    Also, as far as I understand, the AxeFX can be slave to the AES input (48kHz, of course) ... which renders CK's "no single digital amp on the market" statement simply wrong.



  • You will have to call dozens of manufacturers being ignorant or having no clue. :)
    And you will need a truckload of SRC's to have multiple digital instruments or amps connected to one audio interface, unfortunately.


    The component costs of SRCs would only be $15 each for a manufacturer. You would save the additional cable for the sync clock, on the other hand.


    The Korg Kronos allows to be slave, but on 48 kHz only.
    The GT-Pro has no slave capabilities, nor does the Axe Fx!

  • The Korg Kronos allows to be slave, but on 48 kHz only.


    Nothing wrong about 48kHz. I would rather have the Profiler support 48kHz.
    If you want to see a big production facility with 48kHz digital audio only and probably the most complex sync configuration across multiple studios, give me a call when you happen to be near my location.

    ... no slave capabilities, nor does the Axe Fx!


    I don't own an AxeFX and I have no plans to get one, but what I read from Cliff is different to your claim:
    http://forum.fractalaudio.com/…xe-ii-aes.html#post601502
    And the AxeFX manual (page 20) suggests the same I wrote in my prior post.

  • In contradiction to what Cliff said, it was the Axe manual that gave me the impression that it does not support a slave clocking of the internal sample rate. Every Spdif receiver will sync its individual clock to the incoming Spdif stream and send an error message if the clock is not valid. But that does not mean that the overall sample clock will be synchronized at the same time. For a correct slave clocking it is mandatory to have an additional switch to set the system clock source (internal / external). This is usually independent from the input source selection. How will I synchonize my digital output to my DAW, when I still want to use the analog input for my guitar?



    Btw: Lightbox, do you use one or two toslink cables to connect your Kronos to your DAW?

  • Why would you invite me to see a full 48 kHz studio? To teach me that 48 kHz is the preferred sampling frequency in the audio world? I know many pro users that definetely would not like me to learn your lesson :)

  • This is usually independent from the input source selection. How will I synchonize my digital output to my DAW, when I still want to use the analog input for my guitar?


    Again, I don't own an AxeFX so I can't verify. But from what I understand there's the Instrument (guitar) input on the front panel ... and in addition there's 2 analog stereo inputs on the backpanel plus 1 AES/EBU input and 1 S/PDIF input. Maybe these back panel inputs can be configured without any effect on the "main" instrument input on the front panel?

    Btw: Lightbox, do you use one or two toslink cables to connect your Kronos to your DAW?


    I'm smart enough to use 2 cables, don't worry about that. ;)

    Why would you invite me to see a full 48 kHz studio? To teach me that 48 kHz is the preferred sampling frequency in the audio world? I know many pro users that definetely would not like me to learn your lesson :)


    Maybe to teach you a lesson about the existance of movie/video production/broadcast audio. I haven't been talking about one small semi-pro studio. Maybe you remember that I already told you where I work in my main job. But it looks like you just don't want to accept the fact that there's more in the pro audio world than 44.1kHz. This general ignorance is one of the main reasons why my motivation to deal with you (and to keep maintaining the wikpa.org Wiki) has come to an end.
    Everything I ask for, try to explain, request ... everything is treated like I'm a complete idiot from the beginning. I'm completely fed up with this attitude. :(

  • In contradiction to what Cliff said, it was the Axe manual that gave me the impression that it does not support a slave clocking of the internal sample rate. Every Spdif receiver will sync its individual clock to the incoming Spdif stream and send an error message if the clock is not valid. But that does not mean that the overall sample clock will be synchronized at the same time. For a correct slave clocking it is mandatory to have an additional switch to set the system clock source (internal / external). This is usually independent from the input source selection. How will I synchonize my digital output to my DAW, when I still want to use the analog input for my guitar?


    Btw: Lightbox, do you use one or two toslink cables to connect your Kronos to your DAW?



    Pretty much every digital fx hardware box I have ever had allowed for slaving, from Lexicon to Eventide. My H8000 has 4 AES i/o, ADAT i/o, spdif, and analog TRS/TS. I can run analog in and then route out to any number of the digital and/or analog outputs simultaneously, while being either the master or slave at any clock rate. It basically has a mixer and I can mix and match any combination of digital and analog paths.


    I am not sure of the technical limitations of why the Kemper can't do it, but it's definitely not a "normal" thing in a studio environment. No big deal really, I got great converters and can just use it analog (but it would be nice to go straight out digital to bypass a extra conversion step going D/A then back to A/D). It's more that I just need more configurable i/o on the Kemper. If you ever come out with a Kemper Pro (hint) I suggest adding more i/o options. Two true stereo fx loops. And AES and SPDIF that can be slaved. Oh and put more DSP in it so we can run dual amps. :)

    Edited once, last by Animus ().

  • Guys,


    I am a very deep technical IT developer in one of the 3 biggest high performance & highly available clustered databases on the market - just so you know ...


    ... and while I have absolutely no clue about SPDIF master/ slave clocking and what you are talking about in detail, I know that it would certainly help to keep some of that negative emotion out of the discussion and maybe take the discussion offline - maybe you can find some consensus. This discussion is likely going over the heads of most users of this forum.


    My stress levels are already maxed out this year and health can quickly suffer from that. Let's not forget that this is a device to make music and not a medical device to save lives. Let's all stay calm and have healthy discussions, enjoy life and achieve great things like the KPA ...


    Cheers from Canada

  • Lightbox, I am sorry that my attitude feels ignorant tou you. I am trying to evaluate things by facts and asking questions.
    When we talked about the Rotary Speaker, I have presented parameters that I found more valuable than the one proposed from the competition.
    When we discussed the delicate U2 delay I asked if you can hear the delay in question, 'cause I could not.
    Those posts where not meant to be insults, but real questions, that remained unanswered in a way.


    By the fact that we only provide 44.1 you simply conclude today that I simply ignore the need for other sampling rates. What can I say?

  • I have technical problems to quote with my iPad, that's why I create new posts.


    To the OP:
    Eventide and Lexicon have a perfect implentation for digital audio!
    Can you name other companies that do complex DSP outboard equipment providing a complete implementation?

  • On vacation, but this thread got my goat. No, really, hasn't this been done to death.



    Again, I don't own an AxeFX so I can't verify. But from what I understand there's the Instrument (guitar) input on the front panel ... and in addition there's 2 analog stereo inputs on the backpanel plus 1 AES/EBU input and 1 S/PDIF input. Maybe these back panel inputs can be configured without any effect on the "main" instrument input on the front panel?


    The Axe FX, like the Kemper, can only be a master SPDIF clock source. Not the slave. Which means you would have to use 48 kHz. We all know it's the main competitor to the Kemper, I see no reason to spread misinformation to win a point.



    I'm smart enough to use 2 cables, don't worry about that. ;)


    This is just a plain rude response. I'll note that you said the Korg can only be synced to 48 kHz. What about all those other studios that use 44.1 kHz, the ones Mr CK seems to be more familiar with? Should they have a hissy fit on the Korg forums about its inability to sync to 192 kHz? FYI, there are a lot of studios that use higher sample rates and arguments against the need to do so go round and round, just like this recurring topic.



    Maybe to teach you a lesson about the existance of movie/video production/broadcast audio. I haven't been talking about one small semi-pro studio. Maybe you remember that I already told you where I work in my main job. But it looks like you just don't want to accept the fact that there's more in the pro audio world than 44.1kHz. This general ignorance is one of the main reasons why my motivation to deal with you (and to keep maintaining the wikpa.org Wiki) has come to an end.
    Everything I ask for, try to explain, request ... everything is treated like I'm a complete idiot from the beginning. I'm completely fed up with this attitude. :(



    I don't understand the link between your having to deal with Mr Kemper and updating the wikpa to this SPDIF issue. If you don't want to, don't, neither Mr Kemper nor any of the users here will hold you to it. If I remember correctly, your registration of the wiKPA site resulted in some tension on these forums because the person who actually did all the ground work to get that document up and running, i.e. viabcroce, felt that you were taking control of his content.


    As I mentioned, this topic has been discussed before and Mr CK has explicitly stated, "Not going to happen." A simple workaround was suggested through the use of a format converter like this one: http://www.behringer.com/EN/Products/SRC2496.aspx . The product costs 166 euros and will enable you to use the Kemper in a 48 kHz studio environment.


    No one has treated you like an idiot. But you are having a temper tantrum over one perceived "oversight" in the Kemper, according to you. Calling Mr Kemper "ignorant" (which is pretty much equivalent to calling him an idiot in the eyes of this "fanboi") and quoting him out of context is going to result in a war of words out here, I am afraid. The solution has been offered, I don't see what further explanation Mr CK can or should offer.


    I reiterate on Kemper Amplifiers' behalf, slaving to external SPDIF clock is Not. going. to. happen. Deal with it or don't deal with it, it's not going to change. It was a design choice based on several factors, including cost. :thumbup:


    giletti: Saving lives, lol :D

  • As I mentioned, this topic has been discussed before and Mr CK has explicitly stated, "Not going to happen." A simple workaround was suggested through the use of a format converter like this one: http://www.behringer.com/EN/Products/SRC2496.aspx . The product costs 166 euros and will enable you to use the Kemper in a 48 kHz studio environment.


    yeah but... it requires that the studio provide a separate word clock generator in order to use the KPA unit as a slave. (ref p9) manual


    otherwise it's a pretty cool unit for converting to 48k... but only does half the job from what I've gathered...

    Gettin' funky up in here..