I’ve been working on an active speaker project myself, using three honey badger amplifiers per channel. I use something closer to the second option you have provided, but I simply run unbalanced RCA connectors rather than doing the balanced-unbalanced conversion.
I have to ask about how you plan to implement the design you have blocked out.
If the devices are all in close proximity then the likely interference between block elements will also be small and unlikely to be worth the extra expense and complexity. However if you want the lowest noise go all digital but realise it’s likely to add cost and complexity so make sure you think you’re getting an audible benefit.
I have to ask about how you plan to implement the design you have blocked out.
If the devices are all in close proximity then the likely interference between block elements will also be small and unlikely to be worth the extra expense and complexity. However if you want the lowest noise go all digital but realise it’s likely to add cost and complexity so make sure you think you’re getting an audible benefit.
SPDIF code and decode chips are relatively expensive. Therefore I ask if it is worth to build it "fully in digital". My little experience show that solution with analog connections are not bad.
What do you mean? At first I will project PCB next I will test it with SigmaStudio. At the end I will wite program for AtTiny to drive DSP directly without SigmaStudio ( most difficult stage because I need to learn DSP commands ). Commands to mainboard will be send via RS232 from PC or dedicated panel.
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Analog is fine if you keep the levels at the input as high as possible. Best is if you can do the V/C post DAC.
In my HT, the surrounds are also active and it comes from the HTPC analog to the DSP xover then to V/C to power amps. Inputs to the DSP are set so that 0dBFS from the HTPC is just about to clip the input of the DSP.
There are ways to do it all digitally, but they involve pro solutions and are much more expensive than a MD.
In analog version I plan signal level about 5V and next set lower amplification in lm3886 AMP.
You are essentially asking if the 4344 chip will be superior to the onboard Dac in the 1701. If I'm understanding right. For what it's worth, I'm of the opinion that the difference between most modern Dac chips is small enough that your time is 10x better spent tweaking your speakers dsp, fir phase correction, xo slopes, etc then agonizing over Dac chips. I'd use the onboard Dac. Ive never worked with balanced, but couldn't it drive the 3886 directly balanced? Check the data sheet.
@Brett
post DAC volume control doesn't seem practical to me, here's why. It's very true that the input signal should max out the processor(s) to just below clipping. Enter the older cd vs loudness is great cd's. 3-4db variance in input signal. Adjusting 4-6 channels of individual amplifier(s) eventually pooches the overall system balance between drivers. Taking note of each input level as it relates to the source, in this case, is more practical. One knob.
post DAC volume control doesn't seem practical to me, here's why. It's very true that the input signal should max out the processor(s) to just below clipping. Enter the older cd vs loudness is great cd's. 3-4db variance in input signal. Adjusting 4-6 channels of individual amplifier(s) eventually pooches the overall system balance between drivers. Taking note of each input level as it relates to the source, in this case, is more practical. One knob.
That you say makes sense. Probably implementing extra DAC and converters will take to much time and final effect will be the same as using analog connections.
Balanced signal direct to lm3886 ? I don`t think. Of course LM3886 has negative an positive input pin but all circuits i've seen use this pin for loopback only ( maybe i am in mistake ).
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