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DAC digital output question

The little Behringer mixer that I use has a
Texas Instruments PCM2902 chip
inside of it to communicate via usb.

I had a look at the circuit for it and I was wondering if
any of you engineer dudes would know if
you could run an SPDIF signal out from it.

Here's the link for the description and diagram.

http://www.ti.com/product/PCM2902#

Thanks in advance.

«1

Comments

  • edited July 2019

    Sorry dudes,
    I've just realised that this should be in
    Off-Topic and I don't know how to move it yet.

  • Ask @Michael - he might move it :smile:

  • @MrBlaschke

    Thanks dude,

    @Michael

    Is it possible to have this moved to the correct category.
    I think this should be in off-topic.

    Thank you very much in advance

  • Find the pins on the 2902 that are for SPDIF and touch them with a speaker as the load device.
    Do you hear and noises (it will be a digital signal of course) but this would confirm that Behringer
    has it wired/configured to output SPDIF. I suspect it's not putting anything out since that might make the chip run hotter for no purpose. But you should be able to do this without any soldering.

    Let us know what the model of the Behringer Mixer is and maybe more can benefit from your research.

    Personally, I find the "hall monitors" insuring every thread is in the right category when most of us just drink from the firehose is a waste of energy. @LinearLineman is a moderator but he's on vacation and might not read this post at all since DAC digital is like "Differential Calculus" in his musical lexicon. It's not just "Off Topic" it's off limits.

  • Moved. Vacation, yes. Surveillance is everywhere.

  • @McD

    Good suggestion.
    I'll be trying that soon enough.

    If I can get that happening then I can send the signal through to my
    Ultra match Pro SRC96 so that I can upsample the audio coming out
    from the mixer which is consistently running at 44.1kHz.
    With the signal I can then upsample to 96kHz.

    It's for Mastering Purposes using an iPad as the Mastering tool.

    It will give the final mixes that I do air.

    The mixer is a Behringer Q802U.

    This is the interesting thing.

    Every digital audio interface that uses this DAC chip can then be modified.

    @LinearLineman

    Totally agree.

  • @Gravitas said:
    @McD

    Good suggestion.
    I'll be trying that soon enough.

    If I can get that happening then I can send the signal through to my
    Ultra match Pro SRC96 so that I can upsample the audio coming out
    from the mixer which is consistently running at 44.1kHz.
    With the signal I can then upsample to 96kHz.

    It's for Mastering Purposes using an iPad as the Mastering tool.

    It will give the final mixes that I do air.

    The mixer is a Behringer Q802U.

    This is the interesting thing.

    Every digital audio interface that uses this DAC chip can then be modified.

    @LinearLineman

    Totally agree.

    If it breaks your Behringer... I'll apologize in advance. A speaker is really just an external inductor.
    It might be wise to insert a large resistor in the path to avoid the speaker acting as a "short" and damaging the chip. The resistor will stop excess current that could over heat the chip. A potentiometer
    would act as a volume knob so you can slowly increase the speaker volume (assuming that SPDIF is active). It this works the next step would be to run the signal into a pre-amp to insure you're not just picking up some noise. The SPDIF signal might need a little signal boost if it's there.

  • @McD

    Totally hear you.

    I've going to do a test run on another DAC before I attempt to modify the mixer.
    A potentiometer is good advice, I would've gone for the resistor.
    I was going to use a multimeter rather than a speaker but a speaker
    is very straight forward, top tip.
    The multimeter will come in handy if I need to boost the signal.

    The Ultramatch Pro can accept both SPDIF and AES EBU so all is good in that regards.

    Thanks dude.

    Not going to do this until I get my next interface, bear that in mind.

    Need to do the research first.

  • @Gravitas said:
    The little Behringer mixer that I use has a
    Texas Instruments PCM2902 chip
    inside of it to communicate via usb.

    I had a look at the circuit for it and I was wondering if
    any of you engineer dudes would know if
    you could run an SPDIF signal out from it.

    Here's the link for the description and diagram.

    http://www.ti.com/product/PCM2902#

    Thanks in advance.

    You're lucky. About 12 years ago I did exactly such a project, a USB 2902 codec with digital i/o:

  • edited July 2019

    @rs2000

    Dude, that's you???

    It's because of your page that I actually know that the PCM2902 has SPDIF out.

    As the mixer had a usb interface and I know about doing
    modifications I started to wonder what DAC chip was in it.

    I did some googling find the model number, looked it up
    and then ran some more research and your page came up alongside some
    dude who modified a Behringer UCA202 headphone output I do believe.

    You made that interface?

    Super Kudos.

    Respect.

    Awesome.

  • @Gravitas said:
    @rs2000

    Dude, that's you???

    It's because of your page that I actually know that the PCM2902 has SPDIF out.

    As the mixer had a usb interface and I know about doing
    modifications I started to wonder what DAC chip was in it.

    I did some googling find the model number, looked it up
    and then ran some more research and your page came up alongside some
    dude who modified a Behringer UCA202 headphone output I do believe.

    You made that interface?

    Super Kudos.

    Respect.

    Awesome.

    Thanks but no, that's not my page. I was just searching for a schematic similar to my circuit and this one was the best match. I don't have my original files anymore.

  • @rs2000

    Still very cool dude.

    So my questions are this.

    Can I run a very simple SPDIF out from the PCM2902 using two wires
    and a phono connector or do I need to add to the signal
    to make it usable?

    Would it change any of the voltages going in and out of the circuit?

    Would I need to adjust anything else around the chip?

    I've seen something similar with Cirrus Logic chips.

  • @Gravitas said:
    @McD

    Totally hear you.

    Good. I was afraid this mic was not on.

    I've going to do a test run on another DAC before I attempt to modify the mixer.
    A potentiometer is good advice, I would've gone for the resistor.

    Ideally, small signals are analyzed with an oscilloscope and it presents a high impedance load.

    The Resistor is fine just to start. You just don't want the chip to see a wire without a load. Instant smoke.
    Probably not a concern if it's a a/c signal but I'm really just taking a shot at being helpful without really
    knowing if I'm giving good advice... just my best guess.

    I was going to use a multimeter rather than a speaker but a speaker
    is very straight forward, top tip.

    A multimeter for a small a/c signal wouldn't help, I think. Maybe small wiggles but SPDIF is probably
    a high-frequency square wave that acts as a carrier for the digital audio encoding. You'll probably just hear the carrier frequency. To make it audio it might require a "bucket brigade" to effectively drop the frequency. Google SPDIF and see if it documents the carrier frequency. (i'm heading out so...).

    The multimeter will come in handy if I need to boost the signal.

    You have a multimeter with an active preamp?

    The Ultramatch Pro can accept both SPDIF and AES EBU so all is good in that regards.

    I just love acronyms. SNAFU, EIEIO ("Old MacDonald's Protocol").

    Thanks dude.

    I'm dressed as a lady today. No harm intended I'm sure.

    Not going to do this until I get my next interface, bear that in mind.

    I'll forget we ever discussed this by then.

    Need to do the research first.

    Yes. A second opinion would be in order. I remember a little test probe device that would deliver a
    pin's signal to the little "speaker" but that's probably only useful for audio circuits and digital might not even be audible.

    OK. Googling SPDIF for specs...

    S/PDIF is used to transmit digital signals of a number of formats, the most common being the 48 kHz sample rate format (used in DAT) and the 44.1 kHz format, used in CD audio. In order to support both systems, as well as others that might be needed, the format has no defined data rate. Instead, the data is sent using biphase mark code, which has either one or two transitions for every bit, allowing the original word clock to be extracted from the signal itself.

    S/PDIF is meant to be used for transmitting 20-bit audio data streams plus other related information. To transmit sources with less than 20 bits of sample accuracy, the superfluous bits will be set to zero. S/PDIF can also transport 24-bit samples by way of four extra bits; however, not all equipment supports this, and these extra bits may be ignored.

    You're going to need a scope. You could connect the signal to one of your devices and see if it shows a link.

    Q: Who was that idiot that thought he could hear Digitally encoded audio?
    A: He died trying to change the battery in his installed pacemaker. Used a 9-volt and blew the electronics
    and gave himself blood poisoning.

    Q: Really.
    A: Probably a rumor he started to slip in with a new account. I think he might be "DcM".

  • edited July 2019

    @Gravitas said:
    @rs2000

    Still very cool dude.

    So my questions are this.

    Can I run a very simple SPDIF out from the PCM2902 using two wires
    and a phono connector or do I need to add to the signal
    to make it usable?

    Would it change any of the voltages going in and out of the circuit?

    Would I need to adjust anything else around the chip?

    I've seen something similar with Cirrus Logic chips.

    If I got you right then you need a coax S/PDIF output, correct?
    You can couple the PCM2902's output passively but you need a few components for galvanic separation anyway:

    I've done this to equip a CD player with a coax output and I've used a cheap small ferrite ring and a few windings of thin copper wire for the transformer part.

  • @McD

    A dress?

    Cool, what cut.
    Halter neck or backless or it could be both depending upon the cut.
    I prefer both myself when it comes to dresses.
    I know a little about fashion and clothing.

    Don't ask.

    Yeah, acronyms mean,
    'I can't be bothered to type out the whole bloody
    thing if you don't know then I don't know either.'

    I only remembered myself because
    I'm on forum filled with developers and programmers
    and musicians and ,'oh my good gosh', 'squirrel'.

    DcM??? who that???

  • @rs2000 said:

    @Gravitas said:
    @rs2000

    Still very cool dude.

    So my questions are this.

    Can I run a very simple SPDIF out from the PCM2902 using two wires
    and a phono connector or do I need to add to the signal
    to make it usable?

    Would it change any of the voltages going in and out of the circuit?

    Would I need to adjust anything else around the chip?

    I've seen something similar with Cirrus Logic chips.

    If I got you right then you need a coax S/PDIF output, correct?
    You can couple the PCM2902's output passively but you need a few components for galvanic separation anyway:

    I've done this to equip a CD player with a coax output and I've used a cheap small ferrite ring and a few windings of thin copper wire for the transformer part.

    @rs2000

    Precisely.

    I don't want to put in a toslink.
    I need coax.

    That's exactly what I'm talking about.

    So, I can get a passive signal which is what I thought was possible,
    the galvanic circuit thingy is the one though.

    Brilliant, one step closer to my perfect Mastering Chain.

  • @Gravitas said:
    Brilliant, one step closer to my perfect Mastering Chain.

    Sometimes it takes a terrible answer to get the attention of someone that has experience.
    You're welcome.

    Obviously, I lived in the analog age and didn't ever fix anything digital. I actually just broke
    analog things too. Fixing was never the right description of my "bread board project" years.
    I'm mostly poorly recalled classroom theory with no work experience.

  • edited July 2019

    @McD

    Thank you,

    Thirty years ago when I should've been completing my
    electronics studies, I was learning how to play drums.

    No regrets.

    I'm still at the ,'err.... where's the hot bit of a soldering iron', in regards
    to electronics but I've learnt a lot through trial and error.

    More importantly I always ask if I don't know and
    I don't mind getting egg on my face if I get it wrong
    because in those mistakes eventually I learn how to get it right.

  • @Gravitas the S/PDIF signal is high frequency (in the Megahertz range) and therefore very sensitive to connectors and cabling. Proper cables are classified as 75 Ohm types. No need to buy expensive audiophile brands, but quality is essential. The problem is that it's quite difficult to test by ear.
    If the cable is completely wrong (one line of a regular cheapo stereo cable for example) then you'll hear a faint noise and crackling similiar to a heavily used vinyl record.
    A bad quality S/PDIF cable may deliver a blurry ('kind of washed out') sound.

    I mention the cables only because the symptoms of other S/PDIF 'problems' may sound similiar, but originate from the clock driving the signal. The precision of that digital clock signal is crucial and some circuits implement sophisticated auto correction methods for this purpose. Dunno the PCM 2092 details.

    But the worst thing is that 'regular' S/PDIF connections are designed to automatically set the sender as clock master for the whole system.
    That means your 2902 would clock a $$$ interface if you connect it to this interface's digital input, which in turn defeats the clock stabilisation efforts of the interface and drop the conversion quality significantly.

    It's no uncommon that simple S/PDIF connections deliver lower quality than plain analog conversions (given it's a converter of decent quality) - and analog conversion don't influence other system components.
    (in a digital system there can only be 1 master clock)

    sorry for the long blurb, but digital systems are more complex under the hood than it seems at 1st glance.

  • edited July 2019

    @Telefunky

    The signal is going out into a DAC that accepts both SPDIF and AES EBU
    which is the pro version of SPDIF with a balanced signal to improve accuracy
    and an impedance to run long cables.

    The DAC can be used to provide both a clocking signal or it can clock to an incoming signal.
    With the PCM2902 it will have to clock to the mixer but it can also do
    sample conversion which is what I actually want it for.
    In that mode it can take an incoming signal at 44.1kHz or 48kHz and
    upsample to 88kHz or 96kHz respectively.
    In that mode it will clock internally which is preferable for this unit and
    I will be using it's D to A's when Mastering so I don't need it's digital outs.

    Which means that when I need to run an audio signal into an external preamp
    for added warmth or compressor from the iPad? I will get the best signal possible.

    I have plenty of 75ohm cables around.

    I thought connecting the speaker would be cool anyway
    with a Mad Scientist Jacket on a great big power switch.

    Hhmmm I may get the power switch.

    Have a read here on wordclock

    https://www.soundonsound.com/techniques/does-your-studio-need-digital-master-clock

  • checked the PCM 2902 specs, seems ok regarding clock (but I didn't dig into S/PDIF details).
    If the following DAC can handle 2 clocks for upsampling, then it's worth trying if the rest of your setup is based on 96khz :+1:

  • @Telefunky

    I need all four frequencies, 44.1, 48, 88.2 and 96kHz.

    Thanks for checking 👍🏽

  • McDMcD
    edited July 2019

    I'm curious... what is the Behringer mixer model? Running wires for a signal needing 75 Ohm coax is probably "dead on arrival" to protect the signal. Again another guess pulled out of my nether real estate.

    FYI: @telefunky is probably the best (most prolific for sure) audio engineer on the Forum.
    His handle is a clue to his love of the good equipment.

  • @Telefunky said:
    @Gravitas the S/PDIF signal is high frequency (in the Megahertz range) and therefore very sensitive to connectors and cabling. Proper cables are classified as 75 Ohm types. No need to buy expensive audiophile brands, but quality is essential. The problem is that it's quite difficult to test by ear.
    If the cable is completely wrong (one line of a regular cheapo stereo cable for example) then you'll hear a faint noise and crackling similiar to a heavily used vinyl record.
    A bad quality S/PDIF cable may deliver a blurry ('kind of washed out') sound.

    I mention the cables only because the symptoms of other S/PDIF 'problems' may sound similiar, but originate from the clock driving the signal. The precision of that digital clock signal is crucial and some circuits implement sophisticated auto correction methods for this purpose. Dunno the PCM 2092 details.

    But the worst thing is that 'regular' S/PDIF connections are designed to automatically set the sender as clock master for the whole system.
    That means your 2902 would clock a $$$ interface if you connect it to this interface's digital input, which in turn defeats the clock stabilisation efforts of the interface and drop the conversion quality significantly.

    It's no uncommon that simple S/PDIF connections deliver lower quality than plain analog conversions (given it's a converter of decent quality) - and analog conversion don't influence other system components.
    (in a digital system there can only be 1 master clock)

    sorry for the long blurb, but digital systems are more complex under the hood than it seems at 1st glance.

    Hi & Good to meet more tech-savvy guys here :smiley:
    What you say is correct, theoretically, but in practice I can say that it's not all that critical.
    If it's difficult to compare by ear then usually the effect is negligible ;)
    Of course, he should really use his 75Ohm Cables not only to save the S/PDIF signal but also to avoid sending noise to the mixer's internal analog circuitry.

    However I don't get this part:

    But the worst thing is that 'regular' S/PDIF connections are designed to automatically set the sender as clock master for the whole system.
    That means your 2902 would clock a $$$ interface if you connect it to this interface's digital input, which in turn defeats the clock stabilisation efforts of the interface and drop the conversion quality significantly.

    The whole idea of clock stabilisation is to fix jitter and inaccuracies from a clock master, isn't it?

  • @McD

    The mixer model is a Behringer Q802U, basically an 802 with usb bolted on.
    Behringer bought the PCM2902 in bulk and use it in many of their usb mixers
    that have a usb interface inbuilt and for many of their low end 2in, 2out interfaces.

    The Behringer UCA202, UCA222 and the UM2 for instance.
    It's a class compliant chip out from the box so no need to write an special drivers.

    75ohm cables generally tend to keep the signal better than standard
    phono cables so there are less chances of jitter or clicks which means
    the stabilising clock has less work to do.
    Telefunky will most probably be able to explain it much better.

    @Telefunky pleased to meet you.

    The last time I was discussing digital wordclock and similar was
    with the head sound engineer from the Pet Shop Boys.

    @rs2000

    Clock stabilisation is supposed to do that but hi end DAC designers
    expect that their machines will be the Master Clock so
    less attention is given to external clock.
    In the article that I posted up from SoundonSound the reviewer
    noticed that all of the DAC's performed better when clocked internally.
    Still a good DAC will smooth the digital signal which will produce
    a much better digital to audio signal.

    Getting to know you guys so thank you for the introductions.

  • @Gravitas said:
    @McD

    The mixer model is a Behringer Q802U, basically an 802 with usb bolted on.
    Behringer bought the PCM2902 in bulk and use it in many of their usb mixers
    that have a usb interface inbuilt and for many of their low end 2in, 2out interfaces.

    The Behringer UCA202, UCA222 and the UM2 for instance.
    It's a class compliant chip out from the box so no need to write an special drivers.

    75ohm cables generally tend to keep the signal better than standard
    phono cables so there are less chances of jitter or clicks which means
    the stabilising clock has less work to do.
    Telefunky will most probably be able to explain it much better.

    @Telefunky pleased to meet you.

    The last time I was discussing digital wordclock and similar was
    with the head sound engineer from the Pet Shop Boys.

    @rs2000

    Clock stabilisation is supposed to do that but hi end DAC designers
    expect that their machines will be the Master Clock so
    less attention is given to external clock.
    In the article that I posted up from SoundonSound the reviewer
    noticed that all of the DAC's performed better when clocked internally.
    Still a good DAC will smooth the digital signal which will produce
    a much better digital to audio signal.

    Getting to know you guys so thank you for the introductions.

    Let's not forget that the PCM2902 is far from being a high-end codec so I doubt it would be worth to put too much effort into the project B)

  • @rs2000

    It's mainly for fun and the fact that we can do it.

  • ... Clock stabilisation is supposed to do that but hi end DAC designers expect that their machines will be the Master Clock so less attention is given to external clock. In the article that I posted up from SoundonSound the reviewer noticed that all of the DAC's performed better when clocked internally. Still a good DAC will smooth the digital signal which will produce a much better digital to audio signal.

    That article was quite puzzling as it contradicted the common assumption that a solid external master clock could 'upgrade' almost any converter.
    (SoundOnSound generally is a trustworthy resource)

    The PCM 2902 (from today's pov) is an outdated device, yet the designers applied a lot of what's mentioned in the 1st sentence of the quote within the chip.
    @rs2000 's example circuit is a really neat solution which I'd expect to deliver 'good' sound if provided with a healthy analog level, as it does 16bit conversion only.

    My Boss ME-5 has an even older PCM54 that propells this late 80's pedal's reverb into another dimension - one of the rare cases to directly observe converter performance as the same exact design exists without this chip in the RV2 and RRV10 devices.
    The PCM53 (different design, though) converters of the Lexicon 480L are still legendary.
    Bottomline: it's not all about precision, but also about sound.

  • @Telefunky

    That's what I thought when I read the article and saw the results.
    Hence my next thought which was that less attention is given
    to external clock sources.

    That's why the PCM2902 is such a fun chip.
    Everything is already in there though it's only 16bit.

    I'm looking forward to trying @RS2000 's circuit.

    Though the DAC that I'm going into is a Behringer Ultramatch Pro SRC24
    I like the D to A's it has.
    When it upsamples to 96kHz the soundstage widens considerable,
    details are clearer and complex signals are rendered more accurately.

    I was going to modify it by replacing op amps, a better internal clock to reduce jitter,
    removing the capacitors for better ones etc, etc
    I decided to leave it as is.

    Totally agree about the sound.

  • @rs2000 said:
    ... What you say is correct, theoretically, but in practice I can say that it's not all that critical.
    If it's difficult to compare by ear then usually the effect is negligible ;)

    the typical problem within this scenario is that one usually lacks a reference source.
    Differences are in fact easy to spot if such a reference exists and the signal can be switched for A/B comparison.

    Tbh (after several years) I have few ideas about IOS performance in the digital clock domain.
    Sound output is great (unquestionable), but I fail to align digital streams at the sample level which I can do between a Mac Pro Tools and Windoze DAW easily.
    It's not a true problem because sound output is ok, and whatever Apple is doing there, they seem to do it very right... considering they apply a lot of digital audio nogos with their CPU clocking stuff. o:)

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