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Opinions on Hypex Plate (FusionAMP) Amplifiers?

Kelims

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No the Hypex plate amplifiers preamp or volume control will not do this, the attenuation of volume will always be in the digital domain and as a result not ‘bit perfect’.

An analogue pre-amp upstream might do what you want. Although i’m skeptical of the audible benefits of using one.
Thank you very much for your time. That's what I needed to know
 

changer

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Not 100% sure you are asking what I think you are asking. But if you are asking if FIR capability can be added to the current Fusion plate amplifiers through a firmware update, Hypex has confirmed that this is not possible. A hardware change is required.
Yes, this is what I wanted to know. Then I hope they will update the DSP modules and keep them backwards compatible. The replacement is easy and I would buy two upgraded replacement DSP boards if this gives me the ability to apply FIR room correction within the plate amps.
 

YSDR

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Sorry for my questions but I would like to know if the DAC in hypex is equipped with this: Hardware Mixer: some DACs (not all) have an array of internal resistors they can use to change the volume. In this mode, you can change the volume while keeping the audio stream bit-perfect and avoid any audio quality degradation. Excuse my English I have to use translation. Can the hardware control the volume?
There is no bit-perfectness (not as if it were needed) with the Fusionamps, because any input to the amps is converted to 93.75kHz/24bit format, even if we not using the DSP for different filtering tasks.
Anyway, the digital volume control (before or inside the amps) works like a charm, much better than many (or maybe any) analogue volume control.
 
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voodooless

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There is no bit-perfectness (not as if it were needed) with the Fusionamps, because any input to the amps is converted to 93.75kHz/24bit format, even if we not using the DSP for different filtering tasks.
Bitperfect and DSP don’t go together anyway.
 

DS23MAN

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Looks like that might be the case. Audiophonics lists the individual components: adau1450, AK4454, AK5554, SRC4382. Edit: it’s also listed in the manual.

One still wonders why this decision was made? The higher versions of the DSP have ASRC and digital interfaces integrated. One would think that that would make for a more efficient design? Possibly @DS23MAN knows?
Because the SRC4382/4392 are the best sample rate converters on the market and hands on experience in the Grimm LS1 and the Hypex DLCP does help to. It is also a Swiss knive for routing several digital in and outputs.

 
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Gremlins

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Hi all, best wishes 2022!

Question about FA503

I use the fa503 on a 3 way speaker seas kingroy mk3 with 3 different presets for filtering provided by SEAS on a single file that i upload in fa503

I definitely prefer preset 1

How can i upload only the preset 1 in the 3 slots ( i then.select at the back slot 1, 2 or 3

The idea is that is use only preset 1 filter but would like to aplly 3 different equalizer settings for 3 different flavors

Dunno if i make myself understood?

Thanks
 

changer

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Upload the filter set. In Hypex Filter Designer, select preset 1 and choose from the menu to copy to preset 2, preset 3. Overwrite both, then save this as a filter set so you can return. After this, tune preset 2, 3 to taste.
Download the manual here:
 

Gremlins

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Nice ... thank you for tip, i didn't saw it at first
I try this evening

Good day
 

changer

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Is the DAC in the FA-modules grounded?

I can buy a SPDIF transport with galvanic isolation, the prerequiste being the DAC is grounded. I would think so, because of the IEC-60320 C13 cable, but do not know what's happening inside.
 

changer

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Hello all,
what is the correct input level for S/PDIF of Hypex FA123? I use a Raspi with Hifiberry Digi+ and when I use 100%, it currently has some distorted sounds with high level passages of a song. I mean that at some point this was actually not happening and went away, but I cannot tell for sure.
What is correct here?
 

mdsimon2

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In a digital system the max level is 0 dBFS. If you have distortion at high levels it likely means that either the source material is clipped OR you are applying some sort of DSP (EQ, resampling, etc) that is causing clipping. It is important to note that resampling on its own is enough to cause clipping, you do not need any EQ boost. If you do have DSP implemented it is a good idea to back off the volume digitally a few dB until the clipping goes away.

Michael
 

changer

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I have found out the cause is pCP's option for the 10 band equalizer ALSA mixer.

I had developed the habbit to correct "to taste", say, iteratively adopt with listening experience of the filters that I generated based on REW measurements (flat on-axis), comfortably from a 15 band EQ. When I was listening with PC, I used the EQ of my player to find the right balance and had not transferred the band into Hypex PEQ filters. I entered them as 10 band EQ on pCP, then.

But: Curiously, the distortion also takes place by enabling the equalizer alone, which routes the sound output of Raspi first into the software, then to the S/PIDF board.
 

changer

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I used to use the highest possible digital output and set the speaker level from within the Hypex DAC via a remote. The S/PDIF signal goes into the DAC.
 
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mdsimon2

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Got it, DSP with downstream volume control can be a bit tricky and most often requires attenuation to avoid clipping.

Not sure about pCP but I use CamillaDSP which will report if any samples are clipped which makes it easy to dial in the correct amount of attenuation to avoid clipping. I have a similar setup where I do DSP on a Pi but use my DAC for volume control, I have boosts and resampling in my DSP and therefore have 6 dB of attenuation to offset and avoid clipping.

Michael
 

changer

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So this is working now with digital volume set to 100%, without distortion:
Screenshot 2022-02-21 190909.jpg
As opposed to an EQ from baseline:
Screenshot 2022-02-21 183326.jpg

I am still a bit puzzled, because I had this distortion the very first time when I enabled alsa, and it went away by itself, as if there was some feedback loop. But I have dialed it down now, I hope the bits are all still full and as we all know, bits are bits, so it should be fine then .. ;)

Thank you all, one problem less.
 
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changer

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My two-way speaker has a highly efficient woofer, the FaitalPro 12PR320.
I currently run the woofer on one channel of the FA123, which is giving it 75 Watts into 8 Ohm. Will it sound any different if I change to bridge-tied mode, where it would get 250 Watts? Considering that for most listening levels, 75 Watts is enough even for peaks.
 

YSDR

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My two-way speaker has a highly efficient woofer, the FaitalPro 12PR320.
I currently run the woofer on one channel of the FA123, which is giving it 75 Watts into 8 Ohm. Will it sound any different if I change to bridge-tied mode, where it would get 250 Watts? Considering that for most listening levels, 75 Watts is enough even for peaks.
Probably you wouldn't hear any difference in bridged mode compared to normal, if the 75W is enough for you.
 

changer

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Thanks for your reply.
I rewired the speakers yesterday and I had the impression of a more powerful bass. Not as how it sounds when I disable room correction, when everything sinks in deep bass, but stronger as in engines with higher torque. I have to measure the result in the next days, could be wishful thinking too.
 

changer

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I have another request: since I now use two sources, RCA-in for watching movies from computer and S/PDIF from my music server, I need to switch between sources more frequently. Despite that the Hypex remote gives the option, I face the following problem:

I used the EQ function that is on the device panel for room EQ to my satisfaction. But when I switch between sources, the EQ function often gets disabled, sometimes this only happens on one speaker, I cannot recall whether in master or in slave. This is kind of annoying. My current solution is to use auto detect function, but I find it not as good because it always takes the first seconds of a song until the amp recognized the source.

Am I doing something wrong? And is there a way to get the EQ working, as in never being deselected, when I use multiple sources?
 
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