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Another project, all digital DDX amplifier

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I managed to get my friends spare amp tweaked. It's singing now.

I still need to fix my own amp though.

I pretty much redid everything I had done on my broken amp except that I didn't touch the 3rd channel output setup (relay and coils).

I was wondering what happened when I removed the last channel 3 relay and coils on that broken amp.

I guess the amp didn't like that open environment.

HifiMediy - It would be nice if you could speculate what could have happenend when I remove all relays and leave a output coil open on the 3rd channel.

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Anyhow. I still got some questions.

What I figured now:

If I'm running 16/44.1 material I can hear into a decent noise floor.
The known flaw sounds from my experience different than that.
Still - I consider the noisefloor too high.

If I turn the volume down digitally on my Squeezebox. The noise level stays rather constant. Not what I'd call highend sound.

However. I noticed that I do not experience this -- at least that obvious -- on 24/96 material.

What I did then. I upsampled some 16/44.1 reference files with sox:

sox -D -i file4416.flac -t flac -e signed -b 24 -C 0 -o file2496.flac rate -v -I -b 97 96000

Usually upsampling is causing signal degradation. I've never achieved satisfactory results.

In this case though resampling seems to be the much better compromise.
It's really like taking a towel away.

Does anybody has an explanation at hand???

Afaik the CS8416 is doing 96k resampling on everything that hits the input.
Can anybody conform that??

Could it be that this chip is not performing that well on doing the 4416 resampling job???

I've also seen that the the 320 oversamples on 44.1/16. on 24/96 it doesn't (kind of advanced NOS setup ;) ) .

Of course this wouldn't have any impact if the CS8416 delivers constant 96k anyhow.

Any hints to understand respectively improve the situation are appreciated.

I'm still not giving up on that device. ;)

Cheers
 
It's me again.

There's a weired behaviour of the amp.

1.

If I switch from 44.1 to 96 I hear the relay clicking.

The other way around 96 to 44.1 it's not clicking.

That's when the noise starts. Do I miss a CS8416 device reset??


Am I right to assume that if the CS8416 gets resetted, the relays are activated!?!?

And can that happen while switching songs with different samplerates.

2.

I do also think when loosing SPDIF lock from the source the noise starts
after relocking. ( I have to check if that also happens during standby!!)


HifiMediy: I'd suggest to check the resetting scheme of the CS8416 and the other involved chips. I got the strange feeling that something is wrong in that area.

While searching after the issue I read somewhere that the older 8414 had in fact problems in that area (lock/samplrates/reset).
I'm not sure though if that applies to the 8416 as well.

Cheers
 
I'll keep talking to myself. ;)

CS8416 datasheet:

Reset (Input) - When RST is low, the CS8416 enters a low power mode and all internal states are reset.
On initial power up, RST must be held low until the power supply is stable, and all input clocks are stable in frequency and phase.

For how long do you keep RST low?

What happens if input clocks ( The "locked" SPDIF signal itself? ) change in phase and frequency. In cases where you loose sync to the source due to whatever reasons ( reset/powerdown of source/ hook up cable etc./data corruption/sample rate change)?

Even if the DDX is in standby the SPDIF keeps locked according to the SPDIF LED. What happens if things get unlocked (source down) during standby and than while being locked down you wakeup the DDX device and than you start playing again??

The earlier mentioned relay event when switching SR is another odd event. To me at least.

As you can see. I suspect some problems in the initialization area of the CS8416.

Let see. Perhaps there is at least some interest out here to get this device to work properly.
 
Hi folks.

Been away for a few days.

Do I understand it correctly that Hifimediy doesn't seem to be interested to answer or discuss any of the isssues or questions related to their product anymore!?!?

I mean. It's their own vendors froum.

What a pity.

Though, I don't give up on it... ...yet.



In case you guys (hifimediy) read this:

1.
If you've got a new version cooking in the kitchen, please consider
reviewing the choice of SPDIF receiver chip.
As an alternative the WM8804 seems to come with much better jitter specs.

2.
It would be nice if you could introduce an I2S header.
This way all those USB/I2S interfaces can be used to feed the amp directly without the need to go via SPDIF.

Thx.
 
Love it.
Probably LCD would be nice on a board smaller than it or same size so we can mount it easier. I agree with the other guys.
Uriah
I love the LCD screen too! I'd take it one step farther. There's a cool 2.8" color touch screen that's got the same form factor as the Arduino. Using the Arduino as the dev platform, with a touch screen you could use your finger to adjust the volume and other parameters that act as variables. Way cool. Now I'm excited! :D
 
New ST ships

Have you heard of new ST ships :
- All in one STA382BW STA382BWS - STMicroelectronics
- New driver STA309B STA309B - STMicroelectronics
- New power schip STA510F STA510F - STMicroelectronics

STA328BW & STA309B have user programmable biquad, up yo ten per channel for STA309B, so you can thrue your minidsp away ! No FIR yet, but maybe ST plan to add FIR in the next generation.

Futur is all digital.

Hifimediy, do you plan to use them ?
 
MiniDSP have shown they can do DSP, and hifimediy that they can do amps. Maybe it would be more beneficial if hifimediy and MiniDSP made some collaboration, and made a configuration so a hifimediy amp could connect to miniDSP via I2S.

I'd say first HifiMe has to prove that they are able to supply flawless "digital" gear. My DDX sounds good, really good after the mods. However. There's a major flaw in the design and it seems to take them ages to get it fixed.
That's the way how you kill your business.

MiniDsp:

Most of the MiniDsp stuff comes with limiations (samplerates/processing power etc).

I'd prefer the DSP work to be done on a PC. I'd rather opt for a WavIO USB or ExaU2i interface feeding one or more channels via I2S.


Cheers
 
I don't see the limitations of MiniDSP, only the possibilities - and that is intuitive software and a cheap product that will meet many customers needs. If you want more then look elsewhere like the DEQX og PC based. But for me PC based is not an option. I have tried, and it is far from beeing userfriendly.

The highend segment is not that big, and the products actually exists (DEQX, HOLM, Tact/Lyngdorf and some PRO audio DSP's
 
There's new version about this amp on hifimediy site. But there's no information if the problem has been solved or not.

According to the feedback from HifiMeDiy all problems are resolved.

They've done a complete redesign.

V2 comes with quite some interesting new features.

* WM8805 interface chip ( 50ps jitter - much less then tha CS8416 used before)
* no more resampling
* better clock
* no output relais
* different power amp chip on channel 1+2
* smaller heatsink
* 2.0 operation ( you can turn off the bass channel)
* remote control
* better display (volume shown as numbers)

and more.

Looks really promising. Especially that they managed to get rid of the V1 problems.

Of course the price is 30$ higher than before. We'll find out soon if it's worth it.

Two of my proposals they unfortunatley havn't managed to introduce:

1. A pulsetransformer on the SPDIF input.

2. Full shielded output transformers a la Wuerth

That's the point where DIY begins. No dealbreakers IMO. Especially
if you run a pulsetransformer on the transmitter side or if you run
Toslink.


I placed my order already. Looking forward to the new board.


And thx a lot to HifiMeDiy for listening to users feedback.

Cheers
 
Last edited:
According to the feedback from HifiMeDiy all problems are resolved.

They've done a complete redesign.

V2 comes with quite some interesting new features.

* WM8805 interface chip ( 50ps jitter - much less then tha CS8416 used before)
* no more resampling
* better clock
* no output relais
* different power amp chip on channel 1+2
* smaller heatsink
* 2.0 operation ( you can turn off the bass channel)
* remote control
* better display (volume shown as numbers)

and more.

Looks really promising. Especially that they managed to get rid of the V1 problems.

Of course the price is 30$ higher than before. We'll find out soon if it's worth it.

Two of my proposals they unfortunatley havn't managed to introduce:

1. A pulsetransformer on the SPDIF input.

2. Full shielded output transformers a la Wuerth

That's the point where DIY begins. No dealbreakers IMO. Especially
if you run a pulsetransformer on the transmitter side or if you run
Toslink.


I placed my order already. Looking forward to the new board.


And thx a lot to HifiMeDiy for listening to users feedback.

Cheers

So I'll wait for your review before placing my order :) (they have 2 new product: DDX and new ES9023 DAC, must prioritize one of them)
 
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