True (in digital sense) 24 bit assembled USB DAC ?

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Hello,

I am looking at various 24 bits assembled USB DACs - on eBay and Chinese sites like aliexpress, dhgate, etc.

Looking at the items closer I realized that though the DAC chips indeed support 24 bits (with whatever limitations), true 24 bit playback is often possible only through SPDIF input.

This is because USB to I2S converters are quite often 16 bit.

First of all, I'm glad I've discovered it. I need 24 bits not because my ears support that wide dynamic range, but because I need to do some pretty deep digital correction which needs the extra dynamic range.

Secondly, I did find some USB DACs that apparently are truly (in digital sense) 24 bit ones, e.g. the ones with TE7022L chip on them.

I've also found that there are separate USB -> I2S converters, so maybe I should rather be looking for two separate boards ?

And I would like the whole thing to cost up to $75 including S&H.

Any recommendations ? Personal experiences ? For example, regarding how feasible in reality are 24 bits on TE7022L ?

Alternatives to TE7022L in the above price range ?

Thanks in advance.
 
I've also found that there are separate USB -> I2S converters, so maybe I should rather be looking for two separate boards ?

And I would like the whole thing to cost up to $75 including S&H.

Any recommendations ? Personal experiences ? For example, regarding how feasible in reality are 24 bits on TE7022L ?

Alternatives to TE7022L in the above price range ?

Thanks in advance.

If you go for two boards, your not going to meet your budget.

TE7022 WM8761 24Bit 96Khz USB 2.0 DAC Decoder W/ Coaxial Out,The Best USB DAC-in Amplifier from Consumer Electronics on Aliexpress.com

LJM Assembled USB to (coaxial ) SPDIF I2S processor TE7022 chip support 24bit/96K sampling-in Amplifier from Consumer Electronics on Aliexpress.com

one of these leaves you with 40 dollars for a dac. BTW, I have not tried these or know the quality. Good luck.
 
Rise to 100$ and get yoursefl SMSL AD1955 with TE7022. Tweak it to death and enjoy.
The 1955 is one of the best deltasigmas out there, and TE7022 is one of the best USBs out there too (there is nothing better in sub-60$ department for USB input).
Make sure you order the TE7022 version.

Don't afraid if it sounds bad in stock condition - there maybe some flaws in design. Fixing them won't take much time nor experience (at least some of them), and you'll learn something plus get quite good device.
 
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I have an SMSL SD-1955 dac that uses the Tenor TE7022L usb IC, and have had all sorts of problems getting it to work correctly/consistently in 24 bit mode, without drivers it seems mostly to want to run in 16 bit mode. SMSL does not provide any driver support for its products to best of my ability to find such on Google. I did find that the Terralink Asio driver (win 7) will work with this device, and if you change to 24 bit mode in the driver and then restart play it will usually work in 24 bit mode. (The other TE7022L drivers I've found are all specific to the manufacturer that makes them. Source of all of this code is Galaxy Far East Corp,, the maker of the TE7022L) The windows 7 driver itself is quite buggy, and the 7022L is not asynchronous, but isochronous which means there is a significant amount of jitter present in the output. The device may function better in the Linux world, but have not so far tried it. I think you would need to run Alsa and probably force it to default to 96K 24 bit operation. There is some discussion of using this device at higher bit rates/sample depths in Linux in another thread here.

The overall performance of these entry level dacs is probably not going to be stellar.. The SD-1955 is not terribly bad for what I paid for it, but it's pretty mediocre sounding compared to my homebrew PCM1794A/WM8804 based dac driven by a Stello U3.

My suspicion is that you could do far worse than this little SMSL $100 dac, but could also do a lot better if you could build your own or buy a kit around here. The sound at best I would describe as uninvolving and rather lacking in dynamics. I have installed upgraded op-amps (OP2134) which helped. Possibly a better supply would help too, beyond that some fairly significant tweaking would be needed - there is potential, perhaps rehoused in a bigger box to get some room for a really competent analog output stage.

I just had a thought: Check out the TPA (Twisted Pear Audio) stuff here: http://www.twistedpearaudio.com/landing.aspx
Take a look specifically at the OPUS and COD dac pcb kits. Consider one of the XMOS based USB to I2S projects here - purchased piecemeal you might possibly stay under the customs limits. XMOS based solutions are asynchronous, and depending on implementation and coding may support 192K..
 
Rise to 100$ and get yoursefl SMSL AD1955 with TE7022. Tweak it to death and enjoy.
...

It's not about it. I am doing a "big" project regarding headphones auralization and virtualization, and the project is in pretty advanced stage.

Even with a shitty built-in DAC on my desktop motherboard it sounds gorgeous, and, surprisingly, it sounds even better using my laptop built-in DAC - but it has an AD192X (IIRC) chip.

It sounds even better with M-Audio revolution 7.1 which is installed into my desktop and at the moment is used for acoustic measurements.

So, this whole thing is not about tweaking it to death. It's about passing the 24 bits to DAC. I want to have an external USB fed/driven (I mean signal, not power) solution, so it will work for both my desktop and laptop.

I.e. it's the case when I want to spend some money, to connect the pieces together and to use it - I think out of the box it'll sound better than my laptop DAC.
 
...depending on implementation and coding may support 192K..

Why would one need 192KHz if source material is 44.1KHz ?

By the way, I am not interested in "bit-perfectness" - I consider it to be a religion with no practical bearing.

I mean, not only why one would need 192KHz if source material is 44.1KHz, but also why would one need 88.2KHz or 176.4KHz if source material is 44.1KHz ?
 
if you can stretch a bit ..... check out ODAC . it is $99 plus shipping .
NwAvGuy: ODAC Released

Thanks for the link. I especially liked


LESSONS LEARNED: If this project has taught me anything, it’s that getting much better than 16 bit (96 dB) performance can be challenging. The first version of the ODAC, despite following the reference design, only had about 98 dB DNR. That’s about the same as the FiiO E10. The photo to the right shows a few dozen assorted surface mount parts that were laboriously swapped out one at a time and measurements repeated dozens of times using the dScope. Some improvements were far from intuitive. Audiophile preferred polyphenylene capacitors performed worse than less expensive types. Additional filtering on the digital power supply dramatically increased jitter. Chasing down the last few dB of dynamic range the chip is capable of proved to be especially challenging. When it was said and done, the DNR went from 98 dB to over 111 dB. That’s a huge difference and something the design-by-ear crowd would have never achieved.
THE AUDIOPHILE WAY: All too many small or “boutique” audiophile manufactures and DIYers seem to just slap trendy chips on a board, listen to their creation expecting it to sound good (so that’s what they hear), and call it good. Many don’t even follow the reference design. Instead they include a bunch of “audiophile upgrades” expecting better performance—and they hear what they expect to hear even when it’s not true. But the ODAC demonstrated those upgrades often make things worse. So instead of getting even 98 dB DNR like the first ODAC revision, those following audiophile myths and designing-by-ear probably would have ended up with something even worse. Unless you’re making the right measurements, you really have no idea what you’re getting.
.

But I think it's too expensive for me. I mean, there will be experimentation in DSP and analog part outside of the DAC, so I do not know yet what I'll need in the end.

As I wrote earlier, I just know I need spare bits in order to implement digital correction.

...

An assembled board meant to be used with transformers as primary energy sources: Smallest WM8741 TE7022 24Bit 96Khz USB DAC Decoder Digital To Analog Converter | eBay .
 
hmm, so you are after 24 bits, but do not know or care for the benefits of higher bandwidth oversampling with delta sigma dacs? not aware of the fact that resampling will lose you bits of real information and fill it in with aliases? strange. prepared to spend 75 plus power supply, but not willing to spend 99 for one that doesnt need one? i'm afraid you present too much of a moving target to help you and your target is somewhat unreasonable.

just so you know, most of the modern 24bit dacs will also use integrated oversampling filters to push the aliasing and other unwanted products above the audible bandwidth.
 
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hmm, so you are after 24 bits, but do not know or care for the benefits of higher bandwidth oversampling with delta sigma dacs? ...

I am eager to be educated on benefits of running 44.1KHz source material through upsampler (say. 4x upsampler, so we'll have 176.4KHz, but if you insist, it can be 192KHz) and then feeding the DAC proper with upsampled signal.

But you'll have to present some mathematically and physically substantiated arguments.

I.e. you'll have to show that benefits may exist from mathematical point of view (but I know they don't exist) and that they may exist from physical point of view (which can be under certain circumstances, but you really have to show that the circumstances are in place).
 
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Why would one need 192KHz if source material is 44.1KHz ?

By the way, I am not interested in "bit-perfectness" - I consider it to be a religion with no practical bearing.

I mean, not only why one would need 192KHz if source material is 44.1KHz, but also why would one need 88.2KHz or 176.4KHz if source material is 44.1KHz ?

Glad you really don't mind a little imperfection along with your music.. :D

I am fortunate to be able to purchase and download quite a wide range of the music I like in higher resolution formats such as 2496, 24192 and DSD. I figure it makes sense to play them back in their native format. Suffice it to say I am sold on 24 bits and sample rates of 48kHz or greater.

I hope that you weren't serious that it is up to Qusp to prove anything to you. The papers exist, you can find them, read them and get from them what you will.
 
thanks for that Kevin, you saved me the trouble, I couldnt believe that someone, in reply to a post mentioning he was too hard work, asked me to do his research for him and provide it to him in a reasoned presentation as well. I made a part reply but bit my tongue.

nowhere: this is the basis of oversampling dacs designed over the last 20 years, which make up the majority of the dacs available today. I was suggesting dual clocks, its the clock speed and its relation to the oversampling filter that determines whether the oversampling rate is based on 22.1x or 24x integers, not the dac chip

to avoid this process (oversampling) you would have to go out of your way to buy a NOS dac (stands for non oversampling, not new old stock) which coincidentally also mostly happen to be New old stock items because nobody really makes a modern equivalent and certainly not at the ADC stage

it seems to me that you are being too stubborn in your process and are 'throwing the baby out with the bathwater' by erroneously lumping well proven science, which has undergone a great deal of research over the years, not least at MIT, with snakeoil, or other science of dubious benefit. My guess is this is because it doesnt fit with your experience, which I have to say by the sounds of it has some pretty gaping holes in it.
 
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yeah its odd, normally when talking about this subject, the people opposing oversampling are saying that it may exist mathematically but they dont believe it sounds good. I have to make clear i'm talkin about 2 different but related mechanisms here, upsampling and oversampling.

upsampling allows the oversampling filter to work at a higher bandwidth (as it will run at higher speeds again) to push artifacts further out of the audible bandwidth. upsampling being digital, oversampling being at the boundary of digital to analogue conversion in the filters and oversampling allows these filters to act in higher resolution over the waveform, so that any ringing or filter rolloff is at higher frequency, with faster a action. What it doesnt do is create a higher resolution audio waveform with more aded detail in the music, in that part you are absolutely correct. (well actually some fractal type algorithms are being played with, but thats another story)

thats how I understand it anyway, i'm sure someone will fill in some gaps, I dont have the time to go do the research and clean up my wording to be more succinct, but I think I got the gist
 
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