Full digital amplifier with chip STA326

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Hi Kjeldsen

Thanks

“You need to change output device (Recording) to Hifi Cable as well. Realtek digital out.
Both Playback and Recording device needs to be Hifi Cable.”

Done….no change….

“In VST host you need to set-up Asio to output to Reraltek digital output.”

Done….no change….

I went to Devices – Wave – output (MME: Realtek Digital Output (optical)

No sound?????:confused::confused::confused::confused::confused::confused::confused::confused::confused::confused::confused::confused::confused::confused:

Windows PC as a FIR Audio Processor (http://www.jdm12.ch/Audio/2016_Windows-DSP.asp - yes read this...made no sense. Not sure what I need...


I did notice that on:

ASIO Stereo Pair Channel Selection - Output says NOT connected....I am assuming I should have some connection - but how????
 
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I know your frustration - I have been there :)

Here is the link.

http://www.jdm12.ch/Audio/2016_Windows-DSP.asp

You have to scroll a bit down. The first part is no relevant (APO equalizer is another piece of software), so scroll down to headline that says "Asio software" and the part you need right now is under the headline "VSTHost & ConvolverVST".

don't get confused with ConvolverVST right now. First you just need to get some sound through the entire chain. We will get you there, I promise you that.
 
You have to turn on Master panel in view (in the VST host program). Here you can turn up and down.
You can also turn volume up and down in youtube, or in the built in windows mixer (some function might not work, since asio takes over some of the controls)

If sound is too crappy, you might have different values for sample rate and bit depth in advanced sound panel (see the link provided - both Recording and Playback have too bet set identical) - also these same figures have to set-up in Asio panel in VSThost. This is what makes it bit-perfect.

If you don't care about that, you can also use the other VB audio cable - this cable does not require identical bit rate and sample frequency.
 
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Hi Kjeldsen

WOW…. thank you.

I found Master – great thank you.

Yes, I managed to get all the sample rates the same. If you mean in the Windows Sound Box – I have them both (input and output) set to 24bit 48,000

But the You Tube remains less than good….

I will have to call it a night…but thank you… I will be back and this is GREAT.

One last question: in ASIO Bridge, in the middle it says “ASIO on” and “ASIO off”

· Is it on when it says on or when it says off.
· Or do you push the button that says on and it is on, even though it now says OFF.
· There are some triangles that come and go with the button push…
· But the sound remains unchanged????

I suppose I have done something wrong???

Thanks again and hope to get this sorted soon.

You are indeed clever and very patient….
 
Hi.

I'm a bit late to the party. I'd just like to share my experiences.
I've been toying around with full digital amps - DDX320 - and multichannel a while ago.
You'll find related threads over here at DIYA.

My main issue was to get the DDX320 properly going.
"Properly" in terms of real good soundquality.
I tweaked, I had to, the device almost to death.
The input stages are of rather mediocre quality. The device resamples everything internally
to e.g. 48 or 96khz asf.asf. There's been a lot of space for improvement. And I felt
I ended up at a dead-end.

I also looked at the Minidsp products for multiple SPDIF outs. Hmmh.
Limitations all over the place. Nope, all I read didn't sound worthwhile going after a Minidsp solution.
I mean, just have a look at quality differences of 2 channel USB-SPDIF interfaces. I didn't expect the MiniDSP to play in the upper league.
Then I'm no sure if the MiniDSP "DSP" can compete with a PC based setup with high tap FIR filters from Acourate or similar.


Bottom line -- I left it alone...

... too many dead ends and compromises...

...bought an RME Fireface UCX and attached great Abletec AMS amp modules to it.

For the beginning I made use of Charlie Laubs well done (slightly adapted for squeezelite) multichannel setup instructions.

Live can be that easy. ;)

You think...

After toying around with multichannel/active speakers for a while, I realized that all the effort is not worth it.
Again. Compromises and dead-ends all over the place, plus a much bigger hole in the wallet.

I'm back to stereo and passive crossovers.

That doesn't mean though that I don't see space for improvement on my stereo setup.
It still improves continuously. But this I can manage much better then a complex and expensive
active multichannel setup.

Good luck with your project.

Enjoy.
 
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Hi Kjeldsen

All your help has been great. Thank you.

But I know I have a long way to go….

What I have found it that playing a192kHz flac file is OK at 48zHZ, but the sound deteriorates was the kHz rises. When I pay it at 192kHz is it almost unrecognisable.

Also after playing for a while I encountered a buzz…. but I cannot seem to hear if now after a re-boot. (edit - the buzz and creakle is back - say 30mins later)

There are more…but let keep them later

FYI… I have disconnected most of the equipment to make it easier. I only have:

Source – DAC-amp (only 1) to speakers.

So what is the next instalment??

Should I ask????:confused::confused::innocent::innocent:
 
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Hi Kjeldsen

Wow…. beginner…you could have fooled me. I am humbled in your presence. Thank you!!!!

OK… I will give the cable guys a go….

But is there anything else I need?? You are an awesome teacher. – Thank you.

I am supposing the next is DSP and X-overs…

Do I need ConvolverVST – what is it?

I watched a Video and I have

Engine Input – Reagate – Reaeq – Reacomp – Engine output

Not really sure what these are…do I need them?

Can I ask any more questions????
:innocent::innocent:
 
I know how to make it work, but I have not spend much time using it for real.

You don't need Reagate and Reacomp.

You can use ReaEq for both crossover and eq. I use another VST eq for that, but they are essentially the same. Right now I only use the low pass and high pass function in a standard eq. This works amazingly well. You can't choose linkwitz Riley, Bessel, Butterwoth etc - But you can make your own custom slopes.

I think I will pay little to get a dedicated VST crossover (free demo is available)

Buy DSP loudspeaker processing programs

Or I will use RePhase (new to me and FIR filters - I am not very familiar with this). I have tried it, and it works.

https://sourceforge.net/projects/rephase/

You will need ConvolverVST for RePhase

ConvolverVST

With RePhase you can also use a completely different approach than VSThost and asio bridge. This Software is called Equalizer APO

https://sourceforge.net/projects/equalizerapo/

I have no experience, but it seems to be the way to go in the future.
 
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Curious to hear how this chip sounds, I shall shortly be ordering up one of these S/PDIF input Taobao boards : https://item.taobao.com/item.htm?spm=a230r.1.14.66.ebWBam&id=528808249620&ns=1&abbucket=13#detail

That would be nice for a two-way active speaker amplifier. Several years ago I started a "Lite" version of the Active Speaker Designer for the STA32X amplifiers that was going to be free--screen shot below. It lets you read in measurement files for the drivers (FRD) and interactively adjust the filters to model the behavior of the system. It also provided a nice "Amp Control" module to tweak just about all of the registers in the chip.

To use this amplifier as a fully programmable 2-way active speaker, you would need an Arduino Redboard and attach the two I2C lines, which appear to be under the heatsink :(. The Redboard has the FTDI USB interface that ASD-Lite needs. And then add about a month's worth of Arduino programming :rolleyes:.

Hmmm....

STA328-Lite.jpg
 
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Hi Guys

All your help was great... I got some other advice that suggested I try EqualiserAPO....so I loaded this and NO sound...so I uninstalled it and my VSTHost is not working.....

Any suggestions....

I ahve tried all the settings...Now I will try to go back to having nothings...

On the ASIO$all box it say my Realtek HDA SPDIF Optical Out that used to have a triangle now has a red X....

ahhhhhhhhhhhhhhhhhhhh:dunno::dunno::dunno::dunno::dunno:
 
Hi Guys

Me again from the Aussie Bush.... Hope you can help???

I uninstalled all the new software and also ASIO4ALL and re-installed it....
WDM Device List
· Realtek High Definition Audio
· VB-Audio Hi-Fi Cable
· Plantronics D100 – my Phone
· VB-Audio Virtual Cable

So what is the correct settings.

The flac files go out of the PC via the Optical then into an external DAC.

This is what I thought I had:

· Realtek High Definition Audio
o Realtek HAD SPDIF Optical Out
· VB-Audio Hi-Fi Cable
o In: 8x 8-384kHz, 24Bits
o Out: 8x 8-384kHz, 24Bitts
· Plantronics D100 – my Phone
· VB-Audio Virtual Cable

Now:

· Realtek High Definition Audio
o Realtek HAD SPDIF Optical Out RED X

How can I get the RED X off?

Yes, I have turned the PC on and off many times

?????????
:dunno::dunno::dunno::dunno::dunno::dunno::dunno::dunno:

Here is a screen shoot after re-boot...so there "should" not be anything working on the media??? Any suggestions. My PC is strange (I think - or jsut me)...the simplest things take forever

An externally hosted image should be here but it was not working when we last tested it.
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An externally hosted image should be here but it was not working when we last tested it.
 
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Hello Neil,

A two-way FDA is what I'm after for a while !… May be we aren't too far from the solution.

I have a short question about the output filters of most FDAs : if I correctly understood the issue, these filters are optimized for 8 ohms load speakers, this explains why some FDAs are no good enough used with some loudspeakers. Do you think that these output filters can be designed to be used with headphones and if it's the case should one take into account each time the load of his headphone to design the filters ?…

Thank you in advance for your explanation,
 
I'm using some FDAs with headphones - transformers are needed because the outputs are bridged. Unless you re-wire your cans to balanced operation that is.

I did re-design the on-board filters for use with headphones. One example is I used 8.2mH and 5nF as L and C respectively. This is suited to an impedance around 1100ohms for optimum damping. With a 7:1 step down trafo (20V supply to the STA333BW) a 32ohm pair of headphones gets transformed into 1570ohms which is fairly close. I've made a blog post about this, link to the left <----
 
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