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8ch DSP on a shoestring? (SURE WONDOM AA-AB41173)

NotInYourHead

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The Sure Wondom AA-AB41173 is essentially a multi-channel DAC with USB input. A Bravo SA9128 chip provides USB input, as well as 4 I2S outputs for the bottom board. A connector then routes the I2S signals to four PCM 5121s and provides eight output channels. You only need to provide 5V and data through a Type-C cable from your PC.

Now, the special thing here is: there's complete freedom to set up your system. You can use the I2S outputs, combined with Sure's TAS amplifiers, to create fully digital systems. On the other hand, there are 4 stereo analog outputs powered by quality PCM DACs. You're not tied to a particular amplifier, which is a problem I find with DIY DSP amplifier combos.

Even better, the icing on top of the cake is that it's only going for $55 (shipping and taxes not included). 8-channel DSP at this price point is INSANE. I think Sure Wondom made something wonderful that really scratches the itch for budget multichannel DSP.

The only 2 caveats I have are: is 5V from a USB port sufficient (and clean) to drive such a complicated circuit? And you're tied to a PC for DSP adjustments. Personally, I listen to music from a desktop setup, so no biggie, but what if you don't want PC clutter? Maybe a Raspberry Pi could be used for wireless streaming?

1770654277289.png
 
If I understand correctly, this is only a converter and not a DSP.
To add DSP functions, you will need something like an ADAU 1466 Board, for example, additionally
 
There is no DSP, only signal transport / conversion, so no PC involved unless that is your music source. In which case that may as well be your convolver too, before using this, if you even need it for output conversion.

Input is USB Audio Class 2.0 , up to 8 channels (7.1 digital stream)

48-kHz/24-bit only

It can either CONVERT (DDC function) to 4x I2S pairs

or decode to 8 analog channels (7.1 DAC) unbalanced single-ended only, max 1V only? user manual shows dip switches have four levels so I assume 2- maybe even 4V

Yes of course tremendous value - if you even need it for output conversion.


Anyone considering it, have it drop shipped to Amir for testing - ask him first!
 
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If I understand correctly, this is only a converter and not a DSP.
To add DSP functions, you will need something like an ADAU 1466 Board, for example, additionally
 
can be really nice with raspberrypi and camilladsp
 
The analog output volume is only 1 VRMS max. That's half the 2V typically outputted by a DAC via RCA. So, if you're driving speakers with exceptional sensitivity, it'll be fine, but for speakers with average or low sensitivity, you'll need relatively powerful amplifiers, even in high-gain with the drawbacks that entails (reduced SNR, increased THD, etc.).

The prices are attractive, and there are some good ideas, but there are always drawbacks (some more serious than others) with this brand's cards.
 
OK, if that is true, my question becomes what are the three other levels and what are they for?
 
Maybe they're starved of power? These PCMs are rated for 3.3-5V input, but all they have to go on is a 5V/0.5A supply from USB, and there're 4 of them.
Yes, that's what I deduced as well. But it might not be that simple. I imagine that if it were possible to draw more current from the DACs by powering the board using a method other than USB, they would have mentioned it...
 
OK, if that is true, my question becomes what are the three other levels and what are they for

OK, if that is true, my question becomes what are the three other levels and what are they for?
There are TPA3255 amplifier boards with analog inputs that drive the 3255 chip to max power with only 300 mV RMS. One in particular is the SAMP-100. I own one and have tested it using Quant Asylum QA-403. If operated with proper input signal level with power supply in the 36-40 volt range, this amp has remarkably low distortion and noise. I am guessing Wondom has those lower output levels to accomodate boards like the SAMP-100.
 
Damm again, device arrived, already nothing worth, 48kHz 24Bit only......... :(

Not right :

spec-2__127323.jpg

48kHz and 96kHz possible ;) (Bottom note)

I realy like only to buy that Digital Board (Bottom Board) for Testing but that DAC Board don't needed 105 THD+N i s not better tha an normal TV Surroundreciver :(

So i could connect the RPI bord with HDMI into my ONKYO 5007 and get the same quallity without any hassel ;)

That would be nice to use it as an USB to 4xAES OUT (7.1) with 4x Audiophonics i2S to AES , SPDIF and optical Out

https://www.audiophonics.fr/en/inte...al-interface-32bit-384khz-dsd128-p-13627.html


Oh man the Datasheet of that used SA9128 Chip says only 48kHz 24Bit....thats sad :(
 

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So is this good SQ if the low line level is not an issue?

Suggestions for Even Better quality at not too much more moneys would be appreciated.

a Raspberry Pi could be used for wireless streaming?
That's what I'd use it for, "Modularised DSP" ™

24bit 48Hz is as high as I want anyway.

Local RPi4B receiving output from my central convolver over Ethernet (AVB?), e.g. the channels (however many) comprising the F-L/R stacks

this box accepting the RPi4B USB as input, in turn delivering 8x channels SE analog out to 4x boxen on each side.

The local CamillaDSP (or Hang Loose) might be handling local EQ, crossovers, both magnitude and phase tuning, also intra-stack time of flight delays, all the stuff that does not change by location, based on as anechoic as possible measurements.

While the central overall DSP (likely RPi5) handles the DRC, stuff that changes at each new location.
 
Why i stick with 96kHz, there is one thing you need to know about DACS and that is the Outputfiltering neare the top Audible Audioband!

Because english isn't my Motherlanguage i copy and past the Reason thats Google AI says !:

"Choosing between 48 kHz and 96 kHz fundamentally changes how digital audio is converted, processed, and filtered. While 48 kHz is the professional standard for broadcast and video, 96 kHz offers theoretical advantages in frequency response and processing headroom. [1, 2]
The technical and practical differences between the two include:

Sampling Theorem and Filter Design
  • 48 kHz: The maximum reproducible frequency is half the sampling rate (the Nyquist frequency), which is 24 kHz. Since human hearing tops out around 20 kHz, this covers the entire audible spectrum. However, this leaves only a 4 kHz buffer for the low-pass anti-aliasing filters. Consequently, the output filters must be incredibly steep (brick-wall), which can cause phase shifts and transient ringing within the highest audible frequencies. [1, 2, 3, 4, 5]
  • 96 kHz: The Nyquist frequency is pushed all the way up to 48 kHz. This means the anti-aliasing and reconstruction filters can be much more relaxed and gradual. This gentle rolloff preserves phase accuracy and transient details in the audible range, delivering a theoretically purer representation of the original waveform. [1, 2, 3, 4, 5]"
I am not an BAT but on every hearing test with different DAC's i alway prefere 96kHz Musik, less agressive, more relaxed and smoth i would descripe it!
Thats my opinion.

And since 96kHz 24Bit is already an longtime Standart (even before the Marketing startet to sell 192kHz Devices with lag of providing the Audiomaterial for it til this days) on Studiorecording i would never go back to 48kHz only ;)

No one realy need 192kHz, even on Blind tests i was not able to tell you the difference which DAC was playing 192kHz and which 96kHz.

But its already not easey to make an good Blindtest, all Dacs need to setup to the same Outputvoltage and all Dacs need to provide with the same Recording in 96kHz and 192kHz from the same Source, but there not much Sources aviable let say Direstraits Money for Nothing in nativ 96kHz and 192kHz without Remastering in between!

Even if you take an 192Khz source and resample it to 96kHz for comparence you change already something bevore you feed it into an DAC i guess.

So probably the only way to test it right would be to use 2 ADC one set to 96kHz the other to 192kHz, recording an Instrument and than compare it on the different DACS

But probably using an 192kHz Source and downsample it to 96kHz is the easiest way for comparence because the different Samlingfrequency Output Lowpass filter Hits it more in the DACS converting it into the Analogue domain, than the Downsampling process only in digital domain.

I was using also DACS with updampling posebilitys to 192kHz and this sounded better than 48kHz nativ for me. But offten People say the difference, thats upsampling sounds worse, probably depends on his own preferences not sure but for me always less agressive and more relaxed.

But thats the Lowpasfilter goes to the uper range on 96kHz and don't need to be that steep there, is physicaly an Fact and you could also mesure the differences easely!

And the possebility is there for using this samplingfrequency so why still using 48kHz only today ;)
Every cheepest DAC could pay up to 192kHz, even my cheep SMSL SU-2 for 99 bucks lol with an SInad above 120dB

At the END don't forget that youre Roomsetup and Speaker used, makes 90% of the Soundquality at Home and probably only 9 % left for different AMP's and 1% for different DACS or so ;) LOOOOOOOL
 
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Why i stick with 96kHz, there is one thing you need to know about DACS and that is the Outputfiltering neare the top Audible Audioband!

Because english isn't my Motherlanguage i copy and past the Reason thats Google AI says !:

"Choosing between 48 kHz and 96 kHz fundamentally changes how digital audio is converted, processed, and filtered. While 48 kHz is the professional standard for broadcast and video, 96 kHz offers theoretical advantages in frequency response and processing headroom. [1, 2]
The technical and practical differences between the two include:

Sampling Theorem and Filter Design
  • 48 kHz: The maximum reproducible frequency is half the sampling rate (the Nyquist frequency), which is 24 kHz. Since human hearing tops out around 20 kHz, this covers the entire audible spectrum. However, this leaves only a 4 kHz buffer for the low-pass anti-aliasing filters. Consequently, the output filters must be incredibly steep (brick-wall), which can cause phase shifts and transient ringing within the highest audible frequencies. [1, 2, 3, 4, 5]
  • 96 kHz: The Nyquist frequency is pushed all the way up to 48 kHz. This means the anti-aliasing and reconstruction filters can be much more relaxed and gradual. This gentle rolloff preserves phase accuracy and transient details in the audible range, delivering a theoretically purer representation of the original waveform. [1, 2, 3, 4, 5]"
I am not an BAT but on every hearing test with different DAC's i alway prefere 96kHz Musik, less agressive, more relaxed and smoth i would descripe it!
Thats my opinion.

And since 96kHz 24Bit is already an longtime Standart (even before the Marketing startet to sell 192kHz Devices with lag of providing the Audiomaterial for it til this days) on Studiorecording i would never go back to 48kHz only ;)

No one realy need 192kHz, even on Blind tests i was not able to tell you the difference which DAC was playing 192kHz and which 96kHz.

But its already not easey to make an good Blindtest, all Dacs need to setup to the same Outputvoltage and all Dacs need to provide with the same Recording in 96kHz and 192kHz from the same Source, but there not much Sources aviable let say Direstraits Money for Nothing in nativ 96kHz and 192kHz without Remastering in between!

Even if you take an 192Khz source and resample it to 96kHz for comparence you change already something bevore you feed it into an DAC i guess.

So probably the only way to test it right would be to use 2 ADC one set to 96kHz the other to 192kHz, recording an Instrument and than compare it on the different DACS

But probably using an 192kHz Source and downsample it to 96kHz is the easiest way for comparence because the different Samlingfrequency Output Lowpass filter Hits it more in the DACS converting it into the Analogue domain, than the Downsampling process only in digital domain.

I was using also DACS with updampling posebilitys to 192kHz and this sounded better than 48kHz nativ for me. But offten People say the difference, thats upsampling sounds worse, probably depends on his own preferences not sure but for me always less agressive and more relaxed.

But thats the Lowpasfilter goes to the uper range on 96kHz and don't need to be that steep there, is physicaly an Fact and you could also mesure the differences easely!

And the possebility is there for using this samplingfrequency so why still using 48kHz only today ;)
Every cheepest DAC could pay up to 192kHz, even my cheep SMSL SU-2 for 99 bucks lol with an SInad above 120dB

At the END don't forget that youre Roomsetup and Speaker used, makes 90% of the Soundquality at Home and probably only 9 % left for different AMP's and 1% for different DACS or so ;) LOOOOOOOL
Very glad you finished with the truth! Room setup and Speaker used are the bulk of what determines sound quality..
 
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