AES/EBU to two PC's? - Page 1
Forum section: electronics / projects

author: AaronD
author_rank: Frequent Contributor
timestamp: August 10, 2025, 07:47:49 pm
content: I have two PC's that need to receive the same stereo soundtrack. One streams with video and records the stream locally, the other only records audio. Different software, different destinations, different integrations, etc., so they do have to be separate machines.The digital console that produces that soundtrack (Behringer X32) has most of its digital outputs tied up with other functions, so I'm presently splitting an analog RCA out to a USB line-in (Behringer UCA202) on each PC. But even after inserting a set of isolation transformers (Jensen JT-11-FLCF), I'm *still* getting some low-level noise (see Note), even when the console is set for dead silence on that output. I have 6 of them now, from trying one set of 2 just to isolate the source...and a bunch of other combinations when that didn't work...and then another set of 4 to fully isolate everything...still no. So I suspect that at least one of the converters itself has a problem.The console also has anAES/EBUoutput that is presently unused, and I can set it to be a copy of the same RCA analog outs. So, is there something affordable to get that into 2 PC's, while avoiding a ground loop for the remaining analog functions? There seems to be no direct off-the-shelf solution, that isn't stupidly expensive for what it does, but I did find an *incomplete* chain of intermediate standards that looks somewhat promising. Is it really as simple as just slapping two "fancy packaged LED's" (standard connectors) and an appropriate resistor, all in series, across the AES data pins and calling that "2x TOSLINK", to be received byan off-the-shelf USB thingfor each PC?---Note: It sounds to me like a low-frequency SMPS or digital comms, which I've had a ground loop pick up before. The solution there was to isolate the USB side of an embedded sound card, hence the idea here to isolate the audio.

author: moffy
author_rank: Super Contributor
timestamp: August 10, 2025, 09:04:01 pm
content: AES/EBU from what I remember is an expensive to buy standard, perhaps if you just used a good ADC to toslink(split the output into two) and use optical to connect the Toslink to PC you will get around any ground loops, maybe even the low level digital noise if it doen't appear this way.It's best though to indentify the source of the noise by disconnecting things until it disappears, otherwise you can spend a lot of money for no solution.P.S. AES/EBU has a lot more bandwidth than standard Toslink

author: macboy
author_rank: Super Contributor
timestamp: August 10, 2025, 09:22:38 pm
content: AES/EBU and S/PDIF are almost the same. Many S/PDIF receivers handle AES without any issue. So you have a good chance of it just working. The AES/EBU signal is differential and is coded to be DC balanced. So you can likely just capacitively couple it (avoiding ground loops) and feed the two lines into the two inputs of a comparator to recover a single ended signal. Feed that into a pair of Toslink transmitters. Toslink transmitters are very inexpensive to buy or easily salvaged out of discarded TV boxes or DVD players.

author: langwadt
author_rank: Super Contributor
timestamp: August 10, 2025, 09:31:25 pm
content: https://www.gear4music.dk/da/PA-DJ-and-belysning/Lambden-Audio-LA507-Toslink-S-PDIF-AES-EBU-Converter/6Q1F?_gl?

author: AaronD
author_rank: Frequent Contributor
timestamp: August 12, 2025, 12:42:37 am
content: Quote from: macboy on August 10, 2025, 09:22:38 pmI have some extra RS485 chips from a DMX project. Good up to 10Mbps. Would one of those work as the comparator?I suspect that the entire project then, could fit on the back of an XLR connector: one RS485 receiver driving two active Toslink transmitters in parallel, all powered from a USB phone charger. Floating supply, ground-referenced to XLR pin 1, optical out. No ground loop there. Toss some MLCC's across the power pins, and call it a day!

author: cncjerry
author_rank: Unknown
timestamp: August 13, 2025, 03:53:27 am
content: I assume for any USB links you are using USB optocouplers? I've never run video over them, only audio, but I assume they run at the same speed no matter the data. Ground loops are tough to fix and people on this board will go nuts if you suggest to lift the grounds one at a time until you find the culprit. But considering not that long ago, nothing was grounded, and the macbook pro m4 I am typing on isn't either. In fact, running your finger lightly along the macbook while touching other metal feels like there is a low voltage potential between the two. Could also be a fan in a piece of gear.For AES routing and switching, I use a ZSYS AES/EBU router and I am a big fan of them. 8 ins and 8 outs, any to any routing. I think it is an AE8.8 and they make SPDIF 16x16 and others. No DAC in them of course. Seems clean, no splitting issues, no problems with impedance, etc. They can be had for cheap. If you find a parts unit, it is usually because the power supply in them are crap, but you can swap-in an adjustable brick, or repair the supply like I did to two of them just by soldering the transformers.Jerry

author: AaronD
author_rank: Frequent Contributor
timestamp: August 13, 2025, 06:19:27 am
content: Quote from: cncjerry on August 13, 2025, 03:53:27 amI've only had the one so far, and its datasheet looked more like a very short distance (all on the same chip) radio set. Whatever works.Quote from: cncjerry on August 13, 2025, 03:53:27 amEach version of USB has its own data rate (progressively faster), but that's the only determining factor. Sending less data simply creates more empty time slots.But this is entirely about audio. The cameras come in on their own dedicated links, and they (mostly) work just fine.Quote from: cncjerry on August 13, 2025, 03:53:27 am$300 on eBay? Ouch! For a big-budget show that throws around so much money that it can't afford to sit down and actually solve something itself, sure, just buy that and move on. But this is for a no-budget church, and it appears that a prototype PCB and a handful of cheap parts (see my previous post here) might have a shot at working for well under $100 and about a month or so of lead time.Quote from: cncjerry on August 13, 2025, 03:53:27 amYes. I ended up with an IEC-input analog EQ (among other things) from a church upgrade / closet cleanout. They said it wasn't reliable, so I opened it up and happened to notice that the board-mounted mains transformer was arcing to the board. Reflowed those joints, good as new! It's now the master EQ in my home rig.---Anyway, I'm kinda hoping for some commentary on using an RS485 chip to receive AES/EBU with practically zero cable length, just because I have some already and this is a one-off. Blatant mismatch of official standards, but I'm not seeing anything in the actual specs to say it won't work. Anyone else see something wrong with that idea?https://assets.maxlinear.com/web/documents/sipex/datasheets/sp490e_sp491e.pdf

author: alex_
author_rank: Contributor
timestamp: August 13, 2025, 07:01:47 am
content: You can find some off the shelf AES splitters (correctly matched transformers)https://fast-images.static-thomann.de/pics/atg/atgdata/document/specs/255363_data_sheet_v3_en_online.pdfIf you have sound cards in these two PCs you should be all set. AES does not have ground loop issues if all manufacturers do their job properly, if there is residual noise try the ground lift switch.If you want TOSLINK, which is very equivalent, there are plenty of cheap TOSLINK splitters on the market, so you just need a powered AES to TOSLINK in between.

author: AaronD
author_rank: Frequent Contributor
timestamp: August 13, 2025, 05:31:11 pm
content: Quote from: alex_ on August 13, 2025, 07:01:47 amThat appears to be a mic splitter ("Feedthrough of 48 V phantom voltage..."), but it *might* just happen to work anyway.AES/EBU is much higher frequency than audio, but transformers don't have a hard limit in either direction. They simply have increasing magnetic strength for the same amplitude at lower frequencies, and a hard maximum strength before they saturate and suddenly become short-circuit. (In a solid core, that is. Air core doesn't saturate, but needs to be much bigger for the same performance.) But when you get way outside the design range, other parasitics that either weren't accounted for or were left there on purpose as side-effects of optimizing the intended range, can become a problem.Quote from: alex_ on August 13, 2025, 07:01:47 amI think you're confusing AES/EBU (digital encoding) with direct analog. Just because it uses the same connector does NOT mean it's the same thing! Connecting a digital output directly to an analog input will result in either silence because ALL of the frequency content is well above audible and thus filtered out by the analog RF rejection, or horrendous loud noise because digital (on cables through the outside world) often runs at quite high voltage compared to average analog signals (5V for digital, compared to ~1Vrms-ish for analog line level) and its direct physical frequency content is definitely non-musical! (at least by itself; an artist somewhere may see some potential in the far distance and do something with it...)For AES/EBU specifically, I would expect silence because any random two-bit window of time is guaranteed to have equal time high and low, and that sets a minimum frequency that is well above audible.(Actual high or low levels don't matter for AES/EBU. It's the transitions that matter. Always a transition at each clock edge, so the receiver can recover the clock, and another transition in between, or not, to signify a 0 or 1.)Quote from: alex_ on August 13, 2025, 07:01:47 amThat's the expensive part...that looks like it can be made for dirt cheap. And once I do that, it's trivial to just give it a few extra outputs.

author: AaronD
author_rank: Frequent Contributor
timestamp: August 13, 2025, 09:19:36 pm
content: I have a tentative design now.Each TOSLINK out is received by one of these on a different PC:https://www.amazon.com/Channel-External-Optical-Adapter-Laptop/dp/B0CZ8DGC2CIt says explicitly that it accepts AES/EBU, so hopefully the entire system "just works".Thoughts? Still wondering about using the "wrong" receiver chip. Anyone have any insights there?

author: langwadt
author_rank: Super Contributor
timestamp: August 13, 2025, 09:37:38 pm
content: Quote from: AaronD on August 13, 2025, 09:19:36 pmI think it would work, afaiu AES/EBU is really RS422 and the is very similar to RS485

author: jonpaul
author_rank: Super Contributor
timestamp: August 14, 2025, 12:04:46 am
content: Use a 1>>2 110 Ohm splitter.Jon

author: AaronD
author_rank: Frequent Contributor
timestamp: August 14, 2025, 07:58:02 am
content: Quote from: langwadt on August 13, 2025, 09:37:38 pmOoo, nice! Actually, that chip does RS422 too (they're that similar), so maybe it's not the "wrong" thing after all...Quote from: jonpaul on August 14, 2025, 12:04:46 amI assume that's an isolated transformer? Not much point otherwise. Yes, it's digital, and therefore immune itself to minor ground loop noise, but there's still at least one other analog connection that seems to be okay so far, and I'd rather not mess that up. And yes, a ground loop *anywhere* can raise the noise floor of *any* analog connection, even if they're conceptually unrelated to each other. Been there, done that!And there's still the problem of getting AES/EBU into a computer. Pre-made interfaces that go *that direction* are surprisingly hard to find, and expensive. Compared to a TOSLINK receiver that is included in several external sound cards at a consumer-friendly price point. No need for the splitter to isolate, because the optical connection does that anyway.

author: jonpaul
author_rank: Super Contributor
timestamp: August 14, 2025, 12:33:16 pm
content: 1.40:1:1 ratio center tapped sec transformerBW 20kHz-150 MHzj

author: JohanH
author_rank: Frequent Contributor
timestamp: August 14, 2025, 12:47:48 pm
content: Converting between AES/EBU and S/PDIF interfaceshttps://www.diyaudio.com/community/attachments/converting_between_aes_ebu_and_s_pdif_interfaces-pdf.308912/

author: themadhippy
author_rank: Super Contributor
timestamp: August 14, 2025, 01:05:01 pm
content: QuoteHard to find -not really , expensive -yes but then again AES aint really aimed at your average consumer. Lynx audio was our goto for getting stuff from the midas into computers,but if ya need to ask the price.....

author: JohanH
author_rank: Frequent Contributor
timestamp: August 14, 2025, 03:15:17 pm
content: Elektor 7/1997 had a AES/EBU to S/PDIF converter (by Ton Giesberts, so it's probably safe to assume it's been built and tested).And their S/PDIF to Toslink circuit from 4/1997.

author: alex_
author_rank: Contributor
timestamp: August 16, 2025, 12:08:18 pm
content: Quote from: AaronD on August 13, 2025, 05:31:11 pm

author: jonpaul
author_rank: Super Contributor
timestamp: August 16, 2025, 12:47:02 pm
content: Just search AES/EBU splitterAvailable for decades.j

author: AaronD
author_rank: Frequent Contributor
timestamp: August 16, 2025, 04:13:18 pm
content: Quote from: jonpaul on August 16, 2025, 12:47:02 pmFor cheap? Compared to an RS422 receiver chip, phone charger, XLR connector (all from the junk drawer), prototype PCB, and two of these?:https://www.digikey.com/en/products/detail/cliff-electronic-components-ltd/FCR684217T/25672006And there's still the problem of receiving it on a PC. TOSLINK inputs are both cheap and easy to find (ordered 2 for $17 each, which is less than the analog RCA inputs currently in use), AES/EBU inputs seemingly are not.

author: jonpaul
author_rank: Super Contributor
timestamp: August 16, 2025, 10:29:23 pm
content: TOSLINK bad jitter.Passive transformer splitters used in most professional, high end and broadcastj

author: AaronD
author_rank: Frequent Contributor
timestamp: August 17, 2025, 03:22:14 am
content: Quote from: jonpaul on August 16, 2025, 10:29:23 pmReally? This whole thing is a bit-for-bit passthrough, or rather, an edge-for-edge passthrough because that's really all the protocol cares about. No (data) buffering, and no modulation. Where would the jitter come from?The differential receiver chip from XLR?The active TOSLINK transmitter? (which is essentially an LTP with an LED in one leg, or possibly simplified to just a current source and a shorted LED, either way to keep the supply current from bouncing with the bitstream)The fiber itself? (I doubt that: speed of light across negligible distance compared to the bit length)The TOSLINK receiver? (a photodiode with signal recovery...)And what does that jitter actually affect? If the output of the entire process is an evenly-timed series of correct samples, it doesn't matter if the individual bit times wander within that.Quote from: jonpaul on August 16, 2025, 10:29:23 pmProbably because they can afford to have the obvious match - an XLR connector labelled AES/EBU - on everything at a couple hundred $$$ per input. At that point, a transformer split is also obvious.

author: jonpaul
author_rank: Super Contributor
timestamp: August 17, 2025, 06:34:28 pm
content: AaronD: Many thanks,AES/EBU, SP/DIF, AES2id are Manchester encoded, where the data is phase encoded and the clock rate is 128 x FS.A PLL is used to recover the clock for the DAC , reconstruction.Thus a non ideal RX sig will cause jitter in the recovered clk at the RX ckt, the RS-485 PHY.Please see the extensive papers from 1980s, 1990s on Digital Audio clock jitter effect on audio, by our old friendsDr Stephen HARRIS and Dr. Julin DUNN (RIP)The papers are at the AES.ORG and AP.COM sites.Read and learn!Have fun!Amicalement,Jon

author: AaronD
author_rank: Frequent Contributor
timestamp: August 18, 2025, 03:48:38 am
content: Quote from: jonpaul on August 17, 2025, 06:34:28 pmThis isn't quite clear yet. The only RS-485 is the raw-input (*) AES/EBU from the digital console. For the purpose of this project, I assume that it's absolutely perfect at that point because I can't really change it anyway.(*) As far as I'm concerned here anyway. It's actually a fully mastered soundtrack, but that's outside of this scope.Receive it immediately at the console's output connector (no external cable at all) with an RS-422/485 PHY chip, and feed it to two active TOSLINK transmitters right there next to it. No other active electronics at the TX end of the fiber, and no more RS-485 after that ever. The RX end of the fiber is an off-the-shelf USB sound card that claims to directly support AES/EBU by name.So where is the messed-up RS-485 RX that you mention?Quote from: jonpaul on August 17, 2025, 06:34:28 pmDo you have direct links? I'm not finding them.Anyway, I see a clear distinction between the eye diagram (at roughly the bit level) and the samples themselves, where jitter actually matters. As long as the sample clock comes out right, the bit clock can be all over the place within that. So the eye diagram for a serial protocol really isn't all that useful in a discussion about audio quality, unless the bit-jitter has enough cumulative low-frequency content to affect the sample clock, AND that's not accounted for with a suitably-sized RX buffer, local clock, and asynchronous resampler.Don't nail down the time that each sample actually finished arriving, but know that those exact magnitudes are supposed to be evenly spaced. So pile them up in a buffer, call that buffer "equally spaced", and resample *that* between a now-jitter-free RX clock and your own local clock. I'm pretty sure every computer operating system does that already for its audio I/O, to support the myriad of sound cards and all the different ways that they work. Or for a standalone DAC, which I *won't* have in this rig, you might replace the actual resampler with a very slow-adjusting...erm...not really "PLL" because it's based on the buffer fill and not the incoming clock...but at any rate, that oscillator adjusts slowly enough that nobody's going to hear it, which is why the buffer still exists in this version.If that oscillator starts at zero with only a basic control loop, then yes, it's going to take forever to stabilize. Maybe an entire pop song if the TX was shut down beforehand, instead of playing silence or "live mic" noise like the console will. But nothing in the theory says you can't use fancier controls (**), and pre-charge it to the nominal rate specified in the bitstream. This *is* a DIY forum, after all, but I'd be surprised if a "premium" device doesn't already do that. Maybe even some "basic" ones.(**) Like a full PID, that somehow takes an analog value from the buffer fill level; or a microcontroller that knows that inherently, and adjusts its own voltage-controlled-oscillator with a DC-coupled DAC and bypass-able analog lowpass (short out the R's of an N-order passive R-C?), so it can jump to the correct value and then smooth out the already-slow LSb wiggle from there; or an all-digital version of the same rough idea (remember that only the whole sample clocks matter: sub-sample clocks can be any wacky pattern it takes to end up with the correct even output, then actual-PLL up from *that* to get an I2S bit clock or whatever the audio DAC wants); or whatever...---Back on topic...mostly...I actually need to delay the audio by a few hundred milliseconds anyway to line up with the cameras, and an audio-only recording doesn't care at all. So even a big RX buffer by real-time digital audio standards doesn't hurt me at all. I'll just reduce my additional A/V sync delay by that much. Don't even need a prediction: just record some video, see how far it's out of sync, and tweak the setting accordingly. Record another test to make sure it's right, and THEN go live.

author: jonpaul
author_rank: Super Contributor
timestamp: August 18, 2025, 12:32:29 pm
content: AaronD: there is no "messed up RS485"ANY digital RX has comparators, diff amplifiers, an other circuits that have BW, risetime, overshoot, compression as the inut levels change.Thus the eye pattern of the RS485 can be used to see the potential recovered signal jitter.In a Manchester encoded signal like digital audio, the edge jitter results in clock jitter, thus noise in the recovered audio.TOS LINK OPT is not immune as the same principles apply regardless of the transmission media, copper wire, light, radio.Search extensive notes on OPT transmission jitter.Enjoy,j

author: AaronD
author_rank: Frequent Contributor
timestamp: August 18, 2025, 09:04:43 pm
content: Quote from: jonpaul on August 18, 2025, 12:32:29 pmYes to all, except for the part I cut out to respond to separately:Quote from: jonpaul on August 18, 2025, 12:32:29 pmYou're still mashing two different clocks together and calling them equivalent, when they're not: bit clock and sample clock.The eye diagram, and the recovered clock of a Manchester signal, is the bit clock. As I keep saying, that can be all over the place with no effect whatsoever on the magnitude or timing of the actual samples.Force thesamplesto even intervals, as you know they were originally, and you instantly remove all jitter and its associated noise. Easy as that. Except it requires a buffer to do that, and its inherent delay, which for some reason people either don't think of or assume is bad and try to avoid.For a mic to a speaker in the same room, it's definitely bad. So those rigs operate on single samples and shove each one all the way through as soon as it comes in, leaving only the converters' latency. But for a recording, or a stream that has an A/V sync delay anyway, absolutely put a buffer in there to perfectly solve this.

author: jonpaul
author_rank: Super Contributor
timestamp: August 19, 2025, 01:06:14 am
content: AaronD:Jitter URLs, papershttps://benchmarkmedia.com/blogs/application_notes/12142221-jitter-and-its-effects(John SIAU of Benchmark Media is an old friend and a top expert)https://lavryengineering.com/pdfs/Clock_Jitter_and_Clock_Accuracy_for_Digital_Audio.pdfhttps://www.audiosciencereview.com/forum/index.php?threads/digital-audio-jitter-fundamentals.1922/https://www.by-rutgers.nl/PDFiles/Audio%20Jitter.pdfhttps://www.tnt-audio.com/clinica/jitter1_e.htmlENJOY!Jon

author: macboy
author_rank: Super Contributor
timestamp: August 19, 2025, 02:08:03 pm
content: Bit to bit jitter doesn't matter at all as long as it is not so excessive as to cause receive bit errors. The only jitter that has any potential effect on the audio is the word clock coming out of the SPDIF receiver IC. That clock is at 1/128 of the bit clock and it is generated from a PLL whose filter should remove most jitter. Most manufacturers of the SPDIF receiver ICs spent a great deal of effort to ensure low jitter on the recovered clock and therefore on the word clock.While it can be demonstrated both theoretically and practically that word clock jitter can (slightly) affect audio quality, this does not imply that Toslink adds jitter or that it is any worse than electrical connection.

author: AaronD
author_rank: Frequent Contributor
timestamp: August 19, 2025, 06:08:01 pm
content: Quote from: jonpaul on August 19, 2025, 01:06:14 amThere's a lot in there aboutsamplejitter, which is absolutely true, but I still see no direct connection between that andbitjitter, which is what we're talking about on the wire or fiber. Again, pile a few randomly arriving samples in a buffer, then pull them out at even intervals, and you've instantly and perfectly removed all jitter and its associated noise. I think one of those references called it "re-clocking", but didn't go into much detail about it. Do you have something specifically about that logical connection alone? (not that it's particularly relevant, see below)Quote from: macboy on August 19, 2025, 02:08:03 pmAgreed. And I would still be surprised if any modern computer operating system didn't buffer and resample what comes out of that receiver IC, which kinda nullifies the need for that IC to clean anything up at all. Just present the correct magnitudes, in order, regardless of timing, and the OS fixes the timing.In fact, it's REQUIRED that the OS do that, at least for USB and other buffered communications, because it does not actually receive single samples. It receives a chunk and then nothing, then another chunk and then nothing. It MUST provide its own (effective) sample clock, internal to the OS, because the original clock is completely destroyed and replaced with a single number for the nominal sample rate.At no point inside the OS, does an actual sample clock exist. It's all buffers - lists of magnitudes, in order, nothing more - with a single number attached for the nominal rate. Only in a DAC, do we have an actual clock again, and single samples, and that's only on the same PCB that has the DAC chip on it. (likely even the same one big chip that does everything, because that overall function is so common) That PCB (or single chip) also receives big-chunk buffers and a nominal number, anditproduces anewclock according to that number, and pulls samples out of the buffer according to that new clock.So any discussion about jitter noise should be confined to that device alone...which I don't have (or at least care about) except for my headphones, and I'm pretty sure the sound card is good enough for that. If not, then it's still equal to what a good number of viewers are using, so I need to make sure the audio mix still works with it. So again, it's staying put.But if a cheap S/PDIF receiver (perhaps meant for a standalone DAC with that input) already cleans things up, yeah sure, use it for a sound card too. We use chips all the time that have features we don't need, and don't hurt anything to have.