BTW, the on-board balanced output really is quite bad in comparison. With my simple output stage the speakers seem to disappear, while the dynamics and high end of the on-board buffer sound pretty smeared and artificial.
Looking forward to your report of a multichannel set up. Please keep us posted as you implement additional boards.
can you do it in a blind test? Maybe have a friend switch the outputs?... Doesn't even have to precisely match levels...
Been using amanero+1021 with PC and iphone for a few years now without problem. Do note that you don't get volume control with amanero at least on iOS. Also, if you switch between phone and PC, it's a little glitchy and you might need to switch twice for some reason.
I think it is the same in Windows. Applications which use the asio driver otoh can control volume.
Tried it last years ago.The software VC seems awkward to me and i also prefer to leave the computer concentrate on playing and not do unneeded calcs. Same goes for any other form of dsp.
That's not true. The default windows driver gives you volume control, so do WASAPI and ASIO. I also use EqualizerAPO for non-fb2000 audio applications and that works well too, though one could argue the non-exclusive windows audio drivers aren't the most well-written
Having done more experiments I don't think it is merely a problem of the frequency the DAM is syncing to.
I have now tried 2 different sound cards with various clock sources, 4 different ways of connecting the AES/EBU signal (ground/chassis/shield connections) as well as switching the voltage of the AES/EBU signals. I also tried things like adding a small ceramic connecting pin 3 of the AES/EBU input to chassis...
Most of these changes won't affect the clock frequency itself, but they still create a reliable and reproducible sonic difference.
They do affect the digital signal before the input transformer, so there might be different levels of (HF) noise and some of them change the waveform and amplitude of the digital signal. This might translate to an increase or decrease of jitter.
Clocking one of the soundcards to the other one should theoretically increase clock jitter vs the internal clock (according to the manual it's <1ns externally and 800ps internally) and this very audibly worsens the sound quality.
Every one of these different setups translates to characteristical sonic changes, with different frequency bands affected, clarity, focus, room, background vs foreground etc.
It feels more like tweaking in the analog domain...
I also wonder about the DAM1021's S/PDIF input... The DAM1121's input looks familiar and can be set up to accept AES/EBU inputs.
The suspicious thing is the, roughly, 100s period of the effect.
I see that with SPDIF input and the square wave (as linked above), as well as with I2S input and measuring the DAM-clock frequency directly. Therefore my guess that this might be correlated.
At least with I2S, the input jitter should be sub-ps and thus not the primary problem.
I hope that I can soon do some measurements on how the DAM clock reacts on changes of the input clock in a more controlled framework. But I fear I will not be able to make any statements on the effect on the sound.
I see that with SPDIF input and the square wave (as linked above), as well as with I2S input and measuring the DAM-clock frequency directly. Therefore my guess that this might be correlated.
At least with I2S, the input jitter should be sub-ps and thus not the primary problem.
I hope that I can soon do some measurements on how the DAM clock reacts on changes of the input clock in a more controlled framework. But I fear I will not be able to make any statements on the effect on the sound.
Soeren, does the DAM1021's input circuit need to be adapted to accept (balanced) 110 ohm AES/EBU input?
Yes, the level need to be lowered and terminated to 110 ohm.... I can do a drawing later, are too busy right now....
Hi Soeren,
when do you think will you get to drawing an AES/EBU circuit? I wonder if termination might be an issue with my build now... The RME sound cards allow for switching voltage between consumer and professional, but I doubt that beyond that any impedance matching happens...
when do you think will you get to drawing an AES/EBU circuit? I wonder if termination might be an issue with my build now... The RME sound cards allow for switching voltage between consumer and professional, but I doubt that beyond that any impedance matching happens...
It is also moot if one doesn't use a cable with the right impedance. But i don't see why RME would not change the impedance, it is a trivial circuit.
I'm using a standard microphone cable on one output and a 110 ohm cable on the other output. But these cables are short (2m), so I don't think it should really matter that much, should it?
I also don't get the DAM1021's S/PDIF input circuit.
First, there are no specifications for the transformer in the manual. But these should determine the surrounding circuitry.
And second, with the 330ohm termination resistor between the + and - input, wouldn't there be a difference in DC bias between the secondary windings - which should be avoided because it magnetizes the transformer, resulting in higher distortion, wouldn't it?
I also don't get the DAM1021's S/PDIF input circuit.
First, there are no specifications for the transformer in the manual. But these should determine the surrounding circuitry.
And second, with the 330ohm termination resistor between the + and - input, wouldn't there be a difference in DC bias between the secondary windings - which should be avoided because it magnetizes the transformer, resulting in higher distortion, wouldn't it?
Yes, I've seen that. But I'd like to avoid having to build that for 16 AES/EBU channels if I can... Like Soeren wrote yesterday, there should be a way to simply adapt the input circuit.
My pulse transformers were cannibalized from old Digidesign converters, they are meant to work with AES/EBU lines.