S/PDIF Sync still not working...anyone?

[shasha]

Even if the rate is the same (i.e. the mixer is 48 kHz) it's not synchronous and therefore requires sample rate conversion (SRC). In this case the SRC would, for example, be 48.001 kHz to 47.999 kHz. The point being the clocks aren't locked.

Wow - thank you for that explanation. I'm using E-MUs 1820M SPDIF w/ my Ultra, telling the E-MU to get its clock externally (Ultra). Like other users I assume naively that it's best to stay in the "digital domain".

With my setup I sometimes hear really high-end cracklies/distortion that drives me nuts (just monitoring - nothing to do with recording/DAWs, etc.), as if two different sample rates are being used. Now I think I know what might be going on.

Thanks again.

I am so glad that you guys talked about this on here. I was about 2 clicks away from ordering a new soundcard because my current one doesn't allow monitoring of the SPDIF signal to analog out and when I was able to record enable a track in my DAW I heard the same high end crackles and pops. I thought that it was the Ultra being pushed too hard (it was all of 1 day old at that point). I had to solder a new analog jack onto my soundcard (main reason I wanted to go SPDIF to begin with) and it sounds PHENOMENAL! It's so amazingly clean that I couldn't possibly be happier....

But of course if I was looking at new soundcards you know that I was trying to be happier anyway. :lol:

So is AES/EBU not recommended either?

 

[jesussaddle]

I was wanting to try digital on the axe fx myself. I'm not going into a digital mixer. Just directly into my audio interface that has a SPDIF (and adat). So in my case, do you recommend I just avoid digital out on the Axe fx?

Cuz at least one or two people have said that once in the mix there wouldn't be a noticeable improvement, or that the analog out produces warmer tone. Cliff was speaking about a definately cheap SRC, and other mediumly cheap ones. I'm not sure what the chip in my RME rates as. Plus, does anyone out there feel that, in an isolated, not "in the mix" environment, digital is "more there" or somehow clearer, like say in the high end fx like reverbs? Second, is there actual a way to get improvement in the jitter issues using the AES out on the AXE into one of the RME fireface 400 digital inputs, using a home-made converter cable as mentioned? I am fine with staying analog, but is there ANYONE who is benefiting by staying digital, somehow? (You don't need ta answer this second, but maybe in the next year..)



(I have an RME fireface 400. Side issue - does anyone have this interface? with this problem: RME has some weird noise in "analog input 2", relative to input volume, not with the axe fx, but even when no signal is coming into the interface inputs - I don't know what this is - When I switch sample rate it stops and then starts up about 5 seconds later, I played with other settings but nothing works - if I bring down either the playback or output strip for input 2 the noise is killed, but I will be contacting RME).

 

[grape]

I use an RME400 with S/PDIF IO and an Ultra. I had to start out this way due a to a lack of available cables. I also use it on a mac and most of the time it syncs most of the time with FF400 as the master at 48KHz. Unfortunately it loses sync when using other apps like itunes or anything that tries to run at 32KHz. I don't know the exact cause but the jitter becomes pretty bad when running other apps at the same time but that's ok. I don't need to run anything except TotalMix and AxeEdit or Cubase, TotalMix and AxeEdit. If you clip the AxeFX hard on the digital input (at least I think it's the AxeFX that's crapping out) you can also lose sync.

After reading everyone comments over the years I wanted to try analog. Between the FF400 instrument in and the front panel of the AXEFX, I haven't noticed much difference. I'm not hearing the "special sauce" on the Axe input. Therefore, I will most likely continue to use Input 3 on the RME for guitar input. For Axe Output, I made my own cables but unfortunately did not look at the actual analog IO of the Ultra hand. I made 1 FT TRS 1/4" cables not realizing that the 1/4" outs are unbalanced on the Ultra so I connected the outs unbalanced with those cables as well as some 2 foot TS cables. I A/B-ed the analog out vs S/PDIF and S/PDIF is without question better. I had slight phasing with both sets of cables but I haven't done further testing to determine which component is at fault (must be FF or Ultra b/c I tried different cables). I also haven't made XLR->TRS cables yet to test the balanced outs. At this point, I don't expect analog to be any better. I have a feeling that most people prefer analog b/c it softens the tone for them instead of having to eq it - especially in FRFR setups (I only go direct). I also have a feeling that many have tried S/PDIF without an accurate clock which leads to really bad sound.

PS I haven't noticed an issue on FF400 analog in 2. This is not to say I don't have the same issue, but I haven't noticed it.

 

[cporro]

The reason SPDIF is a lousy format is because you have to recover the clock from the data stream.

thanks cliff. this is interesting. so you're saying the format is flawed and prone to un-syncing? does adat have the same issues?

so is the jitter and popping caused when things unsync or is it there even when spdif is in sync? i mean, is there any warning when things have gone wrong?

i only care about the sound. running a RME fireface 800. but have not gotten a spdif signal to show up yet. i have gotten the analog out through my RNP to sound very nice. i believe your advice here is to use the analog outs and a decent pre-amp.

i was skepical about the fractal at first. but i have to say it's the first non-tube amp i've ever played that sounds like a tube amp. amazing and a super good value if you consider what nice tube amps and mics will run.

 

[shasha]

On a slightly related note I just installed a new DAW into one of our studios at work and tried to use SPDIF into the console. I tried using one of those inline converters and even a rackmount box to convert to AES/EBU. Couple things I learned along the way.

I cannot seem to get the audio to pan correctly using SPDIF. In the application I can see it shift from left to right just fine, but coming out of the SPDIF connector it's like it's a summed signal, it just lowers in amplitude, but doesn't pan correctly.

But the great news is that it's even more jittery and poppy than my old cheap soundcard I had at home was when I tried to connect the AxeFX to it. I did upgrade to a new audio interface at home and it worked, but it still sounded and worked better using analog (even better with the better interface). And it wasn't limiting me to using 48kHz for everything either.

I know that some people are insistant on using digital for everything when possible. My experience with SPDIF so far is that it's absolute hit or miss at best and at worse it's unusable. And the stuff I was using at work is definitely not pro-sumer junk, it's commercial broadcast gear and everything else is AES/EBU or balanced audio and works flawlessly.

So my point is that if you've got a SPDIF issue don't think that it's an AxeFX issue, it's a format issue IMHO.