Lets see this week end how i feel it
I have googles so help, and my soldering equipment include a long and narrow tip, though i dont know yet if narrow enough
But but .. 2.18v sensitivity for xlr now seems too little as my dac throw 4v ...
On one side 4v is too little Vs 6.15v
On the other side 4v is too big for sensitivity 2.18
Grrr
I have googles so help, and my soldering equipment include a long and narrow tip, though i dont know yet if narrow enough
But but .. 2.18v sensitivity for xlr now seems too little as my dac throw 4v ...
On one side 4v is too little Vs 6.15v
On the other side 4v is too big for sensitivity 2.18
Grrr
Yes it's strange at first sight, but most DAC-s reaches the best S/N ratio below their maximum output (and the best ones provides that to 0dBFS). So with a 4V XLR output DAC and 2.18V input sensitivity, the FA reaches the maximum amplifier output, if the DAC/preamp is set to approx -6dB. It's optimal if you think about it, because you already have the best S/N ratio from your DAC/preamp and your Fusionamp too, along with the maximum amp output. Not that itt matters too much, because the noisefloor of the FA is already very low and probably your DAC/preamp too.
I was messing with some plate amp configs...
Has the software always asked if you had a FIR module before you set up a plate amp? I wonder if I never noticed or if something is coming out soon?
And, in other news, one of my FA123's has gone "flashy red light of death". I listened to it two times earlier in the day. When I turned it on for a 3d time, nothing. It cost $60USD to ship the thing back from the USA.
Has the software always asked if you had a FIR module before you set up a plate amp? I wonder if I never noticed or if something is coming out soon?
And, in other news, one of my FA123's has gone "flashy red light of death". I listened to it two times earlier in the day. When I turned it on for a 3d time, nothing. It cost $60USD to ship the thing back from the USA.
I had some questions about FA253/NC252NP to what I not find clear answers in documentation
1. If I use Fusion Remote KIT, is then possible to control volume of two FA253 (left/right as master-slave) from remote, when I use SPDIF connection between left right? I need only one Fusion Remote KIT for this?
2. Is volume control with remote made in DSP and digital domain? I presume yes.
3. What is DAC full scale voltage level used on FA253?
4. Did anyone know NC252NP input buffer schematic? I had plan to modify it for use 6 dB/oct dipole slope correction as on https://www.linkwitzlab.com/images/graphics/shlv-lpf.gif
Is it possible with balanced buffer?
1. If I use Fusion Remote KIT, is then possible to control volume of two FA253 (left/right as master-slave) from remote, when I use SPDIF connection between left right? I need only one Fusion Remote KIT for this?
2. Is volume control with remote made in DSP and digital domain? I presume yes.
3. What is DAC full scale voltage level used on FA253?
4. Did anyone know NC252NP input buffer schematic? I had plan to modify it for use 6 dB/oct dipole slope correction as on https://www.linkwitzlab.com/images/graphics/shlv-lpf.gif
Is it possible with balanced buffer?
3. How big is maximum DAC full scale voltage on FA253?
4. I know it can be done on DSP but if I make this on DSP, I will lost about 20 dB headroom outside dipole correction range, additionally volume control take his part so if my signal is 16 bit digital, about 10 bit will be really present on DAC output.
Still, did anyone know NC252NP input buffer schematic?
4. I know it can be done on DSP but if I make this on DSP, I will lost about 20 dB headroom outside dipole correction range, additionally volume control take his part so if my signal is 16 bit digital, about 10 bit will be really present on DAC output.
Still, did anyone know NC252NP input buffer schematic?
There is a datasheet for the FA DSP modul, the DAC output is 2.12Vrms full scale for the NC252MP: https://www.hypex.nl/img/upload/doc/dsp/MP-DSP_Main/Documentation/MP%20DSP%2002xx%20R4.pdf
This give some information but probably mean that if in DSP is defined connection to NC252MP, output is limited in DSP software to 2,12V and no information what is real full scale output of DAC.
Any way now is possible to calculate all gains from the output to input with reasonable precision and make decision, what is better, to use low or high gain on NC252MP.
Any way now is possible to calculate all gains from the output to input with reasonable precision and make decision, what is better, to use low or high gain on NC252MP.
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So you worry about DAC output voltage in order to "overdrive" it to have added gain to compensate DSP settings that you use to equalize for bass output?
I think I'm doing something similar and my only advice is to not worry about it too much. 16-bits is 96 dB dynamic range. SPDIF usually provides 24-bit input data (144 dB). DSP devices usually have greater internal bit depth than the input signal so they have headroom to perform the calculations. It's not uncommon to see a DSP engine working internally with 32 bits or more (I could not spot the bit depth of FusionAmp DSP in their manual).
Additionally the DSP filters work quite the same way as analog crossover components do in this regard, just less noise. So just don't worry too much, we all do terrible things with our DSP and achieve fantastic results!
Calculation can be made in DSP with 32 bit, it did not improve original 16 bit signal, it remains with his -96 bit level quantization noise also on DSP output.
My initial 10 bit remaining resolution estimation was wrong. Usually 24 bit DAC had real resolution about 20-21 bits. If I take off from this 20 dB dipole cancellation compensation (more than 3 bits) remains 17 bits, so I had for volume control about 1 bit before DSP+volume control dynamic range willl be lower than my initial 16 bit signal. This is why I am considering dipole cancellation in analog domain.
My initial 10 bit remaining resolution estimation was wrong. Usually 24 bit DAC had real resolution about 20-21 bits. If I take off from this 20 dB dipole cancellation compensation (more than 3 bits) remains 17 bits, so I had for volume control about 1 bit before DSP+volume control dynamic range willl be lower than my initial 16 bit signal. This is why I am considering dipole cancellation in analog domain.
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Correct me if I am wrong, but I think it doesn't really matter whether you use analogue or digital filtering for the dipole effect.
With digital compensation (I think FA uses 32 bit DSP), the internal bit-depth of the 16 bit sound reamains intact until 96dB attenuation, the limiting factor is the analogue output stage of the DAC, because it have self-noise.
With active analogue compensation, the limiting factor is the output stage of the amp, because it have self-noise, which can be attenuated only by passive filtering.
But then with passive filtering, the limiting factor usually the environment, with its noise.
With digital compensation (I think FA uses 32 bit DSP), the internal bit-depth of the 16 bit sound reamains intact until 96dB attenuation, the limiting factor is the analogue output stage of the DAC, because it have self-noise.
With active analogue compensation, the limiting factor is the output stage of the amp, because it have self-noise, which can be attenuated only by passive filtering.
But then with passive filtering, the limiting factor usually the environment, with its noise.