Indeed, the clocks are running without sync and modules must rely on internal PLLs stability. Wasn't be better to link the modules after ASRC using master module clock? How Kii link works in comparison? I still believe the company using Bruno P. wisdom and experience knows well how to handle the signals.
Thank you for your reply!
But I'm trying to understand how this would work. If I hook one up to be master and the other to be slave and add for instance the IR board then I can set the volume of both speakers/plates through the IR remote of the master. That would mean the master is doing processing at the very least volume before sending it to the slave, or am I missing something here? Trying to understand how this can be done without introducing any delay.
Control signals are superimposed on the spdif connection, not integrated.
So all volume processing is done in each plate.
As far as I understand it now, the asynchronous resampler is running from the same clock (93 something kHz) as the DACs so they are in perfect sync.
So if we take for instance a 44.1kHz spdif signal arriving at both plates/asynchronous resamplers at the same time (as is apparently the case) then it doesn't matter much if the clocks of both plates are in sync. In fact, they could be working at completely different sample rates and as long as the sample rate of the ASRC's are higher than the sample rate of the spdif signal then the maximum error seems to me to be 1 sample. (at any specific sample of the ASRC). So while this is not as perfect as having a single DAC chip running from a single clock handling all channels, it seems to me there can be no real clock drifting effect and the center image should be steady enough in practice. (Benefit though is that there is 0 crossfeed of course
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Exactly - and one sample interval, at 44.1 kHz, is 0.02 ms, corresponding to a difference in distance to the speakers of 8 mm (0.3 in).
I think the maximum error is 1 sample at the sample rate of the asynchronous resampler. In other words, if you take one exact moment in time then it is possible the ASRC of one plate has already resampled the latest spdif value while the ASRC of the other plate still has the "old" value at that moment in time. It would take a maximum of slightly less than the 93 something kHz sample rate before they are equal again. Though this can go both ways so the maximum is +- one sample at 93 kHz, so perhaps this ends up as about the same as your calculation.
I'm not sure though if this directly translates to this (maximum worst case scenario) distance effect you describe..
I'm not sure though if this directly translates to this (maximum worst case scenario) distance effect you describe..
Still trying to decide whether I'm going for a Fusion amp or for seperate (class D) amps, a really good multichannel DA converter and doing the crossover filtering on the computer in high quality.
Designing a really high quality 3-way system and want to get it as good as I can. So sorry if I'm being perhaps overly perfectionistic.
Things I personally still worry about:
DAC resolution and gain.
As I read it, the Fusion plate amp has a -102dB THD+N at -1dBFS.
But where is -1 (or 0) dBFS set? I read in this thread that full output will be updated with firmware to be at the full rated amp power for 8ohm. But firmware updated? I want to know where it is set in hardware. What is the 0dBFS output level of the DACs in Volt (or dBV or dBU) is it for instance 2V? then I can calculate combined with the amp (for instance NC252) specs how many dB down I will actually be for each driver.
I mean if the end result is that for for instance the mid driver I'll be at -60dBFS for it to output 90dB (or 70dB) SPL /1m then that would be bad as I'd have only -42dB till the noise floor/resolution limits of the DAC and having selected a driver with better than -60dB THD at that level most of that would be basically wasted.
The other thing I'd love to know is the resolution / quality of the DSP processing. The crossover filters of the Fusion plate amp DSP, what is their quality, I don't even know what spec applies to this. But if I for instance make a 24dB/oct LR lowpass, how far down is the distortion / aliasing of this filter? As I read it there are very big differences in this for different DSP implementations / filter code.
Is anybody able to shed some more light on my questions? Many thanks if so!
Designing a really high quality 3-way system and want to get it as good as I can. So sorry if I'm being perhaps overly perfectionistic.
Things I personally still worry about:
DAC resolution and gain.
As I read it, the Fusion plate amp has a -102dB THD+N at -1dBFS.
But where is -1 (or 0) dBFS set? I read in this thread that full output will be updated with firmware to be at the full rated amp power for 8ohm. But firmware updated? I want to know where it is set in hardware. What is the 0dBFS output level of the DACs in Volt (or dBV or dBU) is it for instance 2V? then I can calculate combined with the amp (for instance NC252) specs how many dB down I will actually be for each driver.
I mean if the end result is that for for instance the mid driver I'll be at -60dBFS for it to output 90dB (or 70dB) SPL /1m then that would be bad as I'd have only -42dB till the noise floor/resolution limits of the DAC and having selected a driver with better than -60dB THD at that level most of that would be basically wasted.
The other thing I'd love to know is the resolution / quality of the DSP processing. The crossover filters of the Fusion plate amp DSP, what is their quality, I don't even know what spec applies to this. But if I for instance make a 24dB/oct LR lowpass, how far down is the distortion / aliasing of this filter? As I read it there are very big differences in this for different DSP implementations / filter code.
Is anybody able to shed some more light on my questions? Many thanks if so!
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@JustIntonation,
While the Fusion amps are not the cheapest bit of kit on the market, I think they offer amazing value. I'm putting mine into hi-fi like boxes, so I still have the ability to swap out speakers (and amps) when necessary. For the price, I think they are worth a go, if they don't work to the standards you need, sell them.
The old DLCP platform (of which I have sitting redundant now) is another wonderfully good value pre-amp crossover. I believe its pretty old dac chips now, but sounded great, and the ability to plug in 6 nCore amps (any of them) and have it up and running within half an hour to me is worth a lot.
While the Fusion amps are not the cheapest bit of kit on the market, I think they offer amazing value. I'm putting mine into hi-fi like boxes, so I still have the ability to swap out speakers (and amps) when necessary. For the price, I think they are worth a go, if they don't work to the standards you need, sell them.
The old DLCP platform (of which I have sitting redundant now) is another wonderfully good value pre-amp crossover. I believe its pretty old dac chips now, but sounded great, and the ability to plug in 6 nCore amps (any of them) and have it up and running within half an hour to me is worth a lot.
Indeed - and while some people are all too keen to insist on the latest chips, the DACs are unlikely to be the weak link these days. The main shortcoming of the DLCP is the USB interface.
Sorry for the Nooobie Question, but I am not able to configure the HFD Audiosettings correctly so that a Sweep is sent to Speaker for measurement.
I connected the RCA out from the Mainboard to the FA251 RCA inputs. In Asio settings I can chose as Input my UMIK Mic but not really the Speaker output from my MB where the FA is connected. How can I do this?
I connected the RCA out from the Mainboard to the FA251 RCA inputs. In Asio settings I can chose as Input my UMIK Mic but not really the Speaker output from my MB where the FA is connected. How can I do this?
I don't believe the UMIK is compatible, this question was asked earlier in this thread.
I dont get any tone out of the Amplifier. The speaker is connected with the red and black cable and rca is connected. There is simply no output and I proved that there is an input at the rca. No ideas anymore, maybe its broken
First of all, read the manual carefully. You need to upload a filter to the amp to get any output from it. For a starting point, choose a unity filter, this is flat response filter.
Thanks, but I just found a good manual for newbies in the Net.
Hypex-PSC2.400d
The Hypex manual described that a filter is needed, but not which and how to do it.I slowly proceed ...
Hypex-PSC2.400d
The Hypex manual described that a filter is needed, but not which and how to do it.I slowly proceed ...
After few hours of testing, I can say that when using a Reciever with Audissey XT32 the DSP from FA is adding a really bad delay to the signal. It sounds like the sub is then not a fitting part of the music. I had to delete all Biquads and let the XT32 do its Job, then it fits perfectly. Nevertheless, the FA251 sounds great and is more than loud enough for my Visation TIW250XS.