PCM5102 Audio DAC - A quick look at Aliexpress' best!

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I keep getting requests to make an Audio DAC board for use in DIY Monitor Controllers for project recording studio's. As the product definer at TI/BurrBrown, I worked on the PCM5xxx series of DAC's, which in this case, fits the need quite nicely!
First things first, see how cheap they are on Aliexpress, and see how I can improve on a basic module from China.
This one is pretty good, but there are certainly areas for improvement! Comments below! :)
(and please forgive my video skills, i'm not much of a videographer!)

Опубликовано:

28 сен 2024

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Комментарии : 45

@GeorgeTJ 5 лет назад

Amazing that this specific chipset is used on a rather expensive DAC/Amp, the Audeze Deckard

@Mr.Leeroy 6 лет назад

Are you actually working on DAC board? In case you do, how much info would you publish about it, if any? I just really would like to see your designs because I got myself into D/A audio converters and could really appreciate any knowlegde. You may not be a videographer but you present your way of problem solving very impressively. That's rare on YT. You could easily grow an audience.

@expataudio 5 лет назад

Hi Leeroy, thats really kind of you. Have a look at the ohter content I've put on recently, I've been trying to design a mini headphone mixer for a raspberry pi recentlyy (for a gaming application). more of the same stuff I think.

@0io1 Месяц назад

Most of the software controllable pcm5xxx devices have a dac mode choice: New hyper-advanced current-segment architecture Classic PCM1792 advanced current-segment architecture Since you already talked about randomizer I want to know what's the exact difference between two modes guesses: 1)pcm5xxx running at 128x osr , pcm1792a defaults at 64x 2)pcm5xxx have two types of sdm, even switchable segment 3)randomizer changed also a question: is the modulator almost same as pcm1792? (eg. only msb and lower 18 bits modulation via 3rd-order 5-level sdm shown on pds)

@nikolajkynde9714 6 лет назад

Thank you for a nice run through. :) I see similar boards using only one voltage regulator. What is the pros and cons of separate voltage controllers in your opinion?

@DafyddRoche 6 лет назад

Nikolaj Kynde ams1117’s are so cheap, it doesn’t really matter, but mainly, people put them in defensively to try and drop crosstalk between the digital interfaces and the analog references in the device. Remember that any noise on the analog power rail gets coupled into the output (divided down by the psrr). What many forget to consider is that at DC most regulators are great, but with higher frequency noise items, some can look like a straight pass through! I personally think an LC filter followed by a regulator is good for the analog power rail, then another separate for the digital and charge pump should do nicely.

@fabiuh991 Год назад

I just got this board and managed to get output from it fed by a jab5 board. At the moment the dsp program is set so the i2s output comes from the onboard BT device through the i2s bus as well i believe. However the analog output i get is super loud and distorted. So does the i2s signal bridge need to be attenuated somehow? Thank you in advance

@sonovoxx 6 лет назад

Yup. I was one of those yelling "5 volts!" at my screen! Nice tidy breakdown there. Not sure what your end spec will be, but I use 4 x input pairs wired to my monitor controller, with choices of outputs to 3 x feeds (main mons, near-fields, and headphone amp). The 4 input feeds are pre / post mastering chain, an A/B channel (compare mix with similar material), and an aux input (CD player, etc.). My switching setup is all analogue (thought you'd appreciate the spelling), but my digits rely on ADAT lightpipe, and I've often wondered how easy it would be to decode 'any' pair of the ADAT chip output as the input to a DAC. Switching between any set of inputs (called PEC/Direct switching in the film industry where they would match levels on the optical track [Photo Electric Cell] with the magnetic tracks) can be up to 3 times a second when you're chasing down the last bit of the mix, so needs to be done silently and quickly - gapless, but not instant (click!) - which was where I decided I didn't have enough time to figure it all out in digits. 😁

@expataudio 6 лет назад

Keith, you'd definitely need some extra brains for the digits. I'd be tempted to go two fold. First, use an off the shelf ADAT decoder IC. It'll make sure that all of your clocks from your ADAT will be sync'd, then it's a simple case of selecting which data pin you want from the decoder into your DAC. I'd follow that with a variant of the DAC that has I2C (That's PCM5122, PCM5142, PCM5242), and do rapid volume ramps between the inputs, on zero crossing, so that you don't get pops and clicks on switching. One of the nice tricks in that family is that if you ground one of the clock pins, the device goes into "Ramp to GND" mode, where an internal oscillator takes the last bit of data (even if it's at max amplitude) and gradually ramps the output voltage back to GND. Better to do that, than slam the output voltage back to GND. I'll make another video soon with my PCM5242 variation of this board, that I'll actually put the footprint to a small arduino on. That will allow folks to quickly program the devices to do whatever they need, without significant effort. Thanks for the feedback! D

@sonovoxx 6 лет назад

expataudio Great stuff. I don't have enough time to devote to doing it myself (couple of decades out of electronics now), but I'll be following this with interest... 👍

@mofared 5 лет назад

How to expand this interface with Sure adau1701 dsp

@volkerking5932 2 года назад

What you're not mentioned is that is Sigma Delta DAC. It make every time a oversampling with the data that come in so this is in normal 8times the clock of the input signal. 48kHz sample *8 = 384kHz output sampling. The trick is that it make between sampling points new delta sampling points and make a higher (calculated pseudo) resolution but also a higher noise bandwidth and then they need not so good output filters (output filters are filtering the sampling frequency that you can't hear it) because you can't hear 384kHz with your ears. Best Regards Volker from Vienna Thumbs up Nr.160.

@expataudio Год назад

Hi Volker - I apologize that I never replied when you first posted this. Internally, the part has varying interpolatiora to push the data to a constant 384kHz. The modulator takes over from there. The design of the interpolators are a major part of the magic sauce - I’m sworn to secrecy to my deathbed on that one! :)

@weakbit633 Год назад

@@expataudio don't worry - we; you and I shall tell that to the viewer and the world. Also we should mention that a Interpolation sound better but is not the near analog so that should be but this is at the boarder frequency from about 24kHz minus Sampling Filters. Thank you for watching.

@Simone-qr6mq Год назад

Why isn 't the resistance applied to measure the maximum output voltage declared on the datasheet? If it is, where can I find it?

@expataudio Год назад

Hi Simone - it’s been a LONG time since I worked on that part (but it’s still my baby!). The spec for linenout calls for 10kohm impedance. I believe the abs min load is 1K ohm. The line drivers inside were never specified for headphone loads, as the charge pump and the line driver would have needed to be much bigger.

@Simone-qr6mq Год назад

Thanks a lot for the information! I purchased a similar DAC the last year, but I did return it because it didn 't work well with my old amplifier (which had a sensivity of 200 mV on 22 Kohm). However, I would like to try again, since I 'll have to wait some time to replace that amplifier. So, I 'm designing a voltage divider to match the 2V output with the 0,2V required by the amplifier and I thought that the audio source 's output impedance would 've been important. Given that the amplifier has a 22Kohm input impedance (which I will identify as a load for the voltage divider output), should I do the calculations with an input voltage of 1 - 1,2V instead of 2V?

@expataudio Год назад

That’s an OLD amp! (I bet it sounds great!). The 2VRMS specification comes from a Dolby digital spec that came about late 90’s, early 2000’s. It’s way too much for an old amp, so your idea to use a voltage divider is great. Design of your divider will be heavily influenced by the load the amp will present. That will be a bit of a challenge too!

@rajatbansal512 5 лет назад

what amp would you suggest under 80 dollar from alibaba please help

@expataudio 5 лет назад

Rajat Bansal I think any of them using a device such as TPA3116.

@rajatbansal512 5 лет назад

@@expataudio none of the big brands use tpa3116 plus it's not just about the amp chip is it it's also about the opamp used with the configuration if it was just about the amp chip we would have more insane build like O2 amp .

@expataudio 5 лет назад

IT appears you've answered your own question! :) The PCM5102 doesn't require an external opamp for I/V conversion, nor does it need to boost it's output for most amplifiers. Worst case, you have to attenuate. For differential inputs, take the big brother, PCM5242.

@Jay-fp8iy 5 лет назад

Did you do electronics engineering?

@expataudio 4 года назад

Yes. Although I’m not sure how much I’ve retained

@steffenstengardvilladsen3740 5 лет назад

Nice walk-through of the design. Will you make a video on the more advanced DACs and why a I/V section is needed? I am debugging an the DAC-section (on subwoofer output) of a Onkyo SR876 receiver and i think i will order such a board to ensure the noise is not already present within the signal www.diyaudio.com/forums/solid-state/332666-noise-subwoofer-output-onkyo-tx-sr876.html

@NikiDorian 4 года назад

nice video, no need for the music

@cbr10009 3 года назад

hi expataudio, since you work for ti/burrbrown,i would like to give you a feedback regarding the sound quality of newer dacs.newer dacs dont sound as balanced as older philips,wolfson dacs.i have listened to products containing philips tda1311 ,1387 dac,wolfson 8772,pcm1727,pcm1742,pcm5102,dsd1793,cs47l91.i have done a few experiments with passive components and opamps,and from my experience all the output opamps/passive components around the dac chips were almost neutral/normal sounding-so difference in sound is only due to dac chip itself.i find newer dacs push voices forward and snare drums backwards.(i think voices are brought forward to enhance back ground details-but the side effect is the snare drums get pushed behind and this reduces the musicality while listening songs.Can you tell me which songs you all use while designing dacs?bcoz if you listen to certain songs that have forward snare drums(such as michael jackson's songs),you will not immediately notice that snare drums are being pushed behind.I would request you to listen to products containing above dacs,if possible,to better understand what i mean to say.I was searching for a dac that could replicate the sound quality of old philips,wolfson,burrbrown dacs.But newer dacs dont sound the same,so i had to look for used older product containing old dacs.

@jonlaws4493 5 лет назад

Hey Dafydd, I stumbled across your RU-vid channel. I remembered you from the Nice and Northampton days. Good to see you doing well. Jonathan

@expataudio 4 года назад

Now THAT's a blast from the past! Hope you are doing well!

@АлександърПанов-н4у 5 лет назад

Thank you for deep review! I just order PCM2706 and PCM5102. I have intention to put them inside my russian amp with germanium transistors Аккорд 201 Стерео. I make that step to integrate in amp a DAC because I listen only a music from computer. This will save me lot problems with interference in input of amp.

@tele_visiones 5 лет назад

Thanks for the video! do you know if its possible to have 2 of this AUDIO DAC running on one Raspberry? the thing is to have 2 differents stereo outputs to use for DJING on MIXXX software. Thanks

@expataudio 4 года назад

the challenge doesnt rest with the DAC, but with the Raspberry PI. The PI would need two DOUT pins and a single set of clock signals. Otherwise, the DOUT pin would have to support a TDM (Time Division Mutliplex Mode). That makes like alot harder. you'd need to change to PCM5122 to configure TDM input modes.

@pauls-audiolabor 5 лет назад

Thank you, very revealing video :)

@slli9052 4 года назад

HI, i mhave a RPI A model , wnat to use this to be connect this PCM5102 Audio DAC, can kindly advise me , how to connect this pins to the P5 socket. thanks

@iceberg789 5 лет назад

how is the sound quality ? hifi ?

@expataudio 5 лет назад

It's a lovely sounding DAC. Architecture was initially based on the classic PCM1792A, but the digital core was upgraded to have a better randomizer. The randomizer evens out tones (moves very low level tones out into the noise floor)

@zighunt 4 года назад

Hi, I'm wondering can the negative charge pump be fed by a seperate regulator also?

@expataudio 4 года назад

it already is on this schematic. (a separate +3V3). I'm uncertain if the hardware controlled PCM51xx devices can switch off the internal charge pump and use an externally provided -3.3V.

@mohammedshalash7842 4 года назад

I used TDA1543 and working perfect so I bought this PCM5102 module and it does not work at all !! I decided to go for I2S Audio Breakout - MAX98357A from Maxim (waiting for delivery)

@expataudio 4 года назад

That surprises me! Did you check voltages around the device?

@mohammedshalash7842 4 года назад

@@expataudioThanks for your reply, Yes I did , my other option is MAXIM MAX98357A