The New Hypex Fusion Plate amps

I maximally reduced Fusion Class D pre-amp gain by modification (removing some resistors, search back in the thread). I do not need the headroom with my relatively sensitive drivers. On top of this, I put series resistors on mid and AMT drivers. Mundorf M-resist Supreme. This way I think have a reasonably optimal gain structure for my speakers. And the series resistors also provide measurable frequency response benefits (and likely somewhat lower distortion as well).
Thanks! Good to hear there don't seem to be any obvious flaws in FA's volume handling. This gain removal in the pre-amp is a smart trick. I need a lot of headroom for my woofers so I'd be more likely to resort to individual channel attenuation.
A tip from my days when I dealt with super efficient full range speakers - Duelund resistors can't be beaten for that. I can't justify the cost of their caps but those resistors are worth every penny.
 
Lowering volume raises relative noise floor, but if you use digital input, the noise floor is as low as it theoretically can be. And the bits that you loose, are the quietest ones that you wouldn't hear anyway. If you are afraid of losing bits, the only option is to play back at nothing but full volume. Analog attenuation introduces more noise than digital, because analog components add noise among other problems.
I'm definitely not advocating for analog over digital but I think it is not so black and white. Digital, just like analog, can be done better or worse. Try comparing ESS chip in a good DAC against iTunes as the source in the same setup. Good example is Gustard DAC too, where they included both analog one step drop and digital volume control. Having said that, I'm leaning towards full digital path with the Fusions.

If you read ASR, be sure to read the whole post containing the measurements and not stop after SINAD. With comments like that you almost sound like you think that SINAD is the only measurement in the world.
Ironically, that's the point that we were trying to make. It is ASR afficionados who often want to limit the quality to just SINAD reading. Believe it or not, I sometimes read their tests and I value the results for what they bring to the picture, which to me is value added and not the core assessment. I'm yet to find the test which translates to how realistic the instruments sound or how good imaging is. Let alone how well it will integrate with my gear. It is not ignorance of science on my part, it is the ignorance of reality amongst many of ASRers. It can sometimes be amusing too, e.g. total disregard of tubes. But generally I agree these tests bring more value than sponsored articles in most magazines.
 
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I'm yet to find the test which translates to how realistic the instruments sound or how good imaging is.
I have found that with any gear that tests audibly transparent (and there is a lot of gear in that category), how realistic the instruments sound or how good imaging is totally depends on the recording, speakers, room, position of the listener in the room - and mood of the listener.
It can sometimes be amusing too, e.g. total disregard of tubes.
How many car magazines discuss steam engines? :)
 
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I actually have a RME AIO PCI express audio interface. I did not recently use it for S/P-DIF, but I do not recall it was so good jitter wise...

Still I like RME a lot, mostly for good & long term driver support. I have a nice pile of internal and external soundcards in the attic that are worthless due to lack of driver support... :-(
My RME UFX is 12 years now. Plays like day 1 and get's a few software updates the year.

At least the USB/Firewire interfaces have some anti-jitter tech built in which works extremely well. Don't know the name, just have a look on their homepage and search if your card has this tech built in too. Not sure about the built in ones. Look here:
https://www.audiosciencereview.com/...yface-pro-fs-portable-interface-review.12313/
 
Would you mind sharing your thoughts on controlling the volume in this setup. NCores seem to like a very strong signal but PC is likely to have a better digital volume in you use players like Jriver. To be clear I don't know how good FA mechanism is but some comments in this thread mentioned better sound quality when adding gain instead of lowering it.
If you go with a digital source, you would want to send the signal with the highest output possible that still leaves headroom. This is to saturate all bits. I set the piCorePlayer to 90 percent. When I used the internal alsa mixer equalizer, this headroom was necessary to avoid clippling. I have however never experienced this since I use CamillaDSP instead to give me physical tone controls. The attenuation is done with Hypex IR remote, I never manipulate the source level. My startup volume is set to -40 dB in Hypex Filter Control, which is a good low level listening volume with my 94 dB efficient speaker.
 
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That's EXACTLY the same you do here. Hypex plate amp does the volume control before D/A conversion. So it doesn't matter if you control the volume in the computer or in the DSP.

But don't worry - these amps have all their noises and artefacts so low you can not hear them any more.
With an analog signal you also have a noise floor and dynamic range - you can't hear details smaller as the noise floor! Similar is with digital - all your great 24bit information would be thrown away anyways when leveled down analog cause it's burried in the analog noise.
As long as your resolution is high enough and noise is low enough digital volume control works great!


(Actually you can hear "into" the noise floor. There is dithering. This Test CD has a level of -100dBfs recorded with dither which you can fully reproduce with 16Bit: https://www.discogs.com/de/release/8312637-Various-CD-1-Test-Disc
But that's not easy to understand ... )
 
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I can share mine. If you feed digital input, it almost don't matter one bit (pun intended) how you control the volume as long as your gain structure reasonable. Digital volume control is a multiplication arithmetic operation and it's very easy to get it right.

Lowering volume raises relative noise floor, but if you use digital input, the noise floor is as low as it theoretically can be. And the bits that you loose, are the quietest ones that you wouldn't hear anyway. If you are afraid of losing bits, the only option is to play back at nothing but full volume. Analog attenuation introduces more noise than digital, because analog components add noise among other problems.

I often feed my FusionAmps a digital signal that is attenuated by 50 dB, that's for when the kids are sleeping and I'm still watching TV. I tried to lower it to -60 dB and that's closing to a level where the sound is barely audible at all in the listening spot. Even then I have not noticed any audible defects in the sound coming out of my speakers. The DSP is able to process and cross it over between elements without any trouble. -60 dB equals about 10 bits of attenuation.
I think you are not using your Fusion optimally this way. Fine if it sounds good enough for you, but still...

First of all, 50 dB is anyhow quite a lot. It is a loss of about 8 bits. Do you ever use the -25..0 dB range during louder listening? If not, there is too much system gain.

Secondly, I would always recommend to attenuate at end of audio pipe, not at begin (presuming clipping is avoided). By doing this on PC side you will worse SNR or more rounding errors in ASR and DSP processing...
And input is 24 bit, so with a bit more than 8 bits loss you are getting below 16 bit resolution already at start of audio pipe!
 
Thanks! Good to hear there don't seem to be any obvious flaws in FA's volume handling. This gain removal in the pre-amp is a smart trick. I need a lot of headroom for my woofers so I'd be more likely to resort to individual channel attenuation.
A tip from my days when I dealt with super efficient full range speakers - Duelund resistors can't be beaten for that. I can't justify the cost of their caps but those resistors are worth every penny.
The tweak can be different per channel, so you could do it only for mid and tweeter and keep woofer gain high. But I would first tune the speaker with software attenuation and see where you end. For me it works with same preamp gains as I do a lot boosting. But of course you need to watch out for clipping at high levels this way.

Yes, I was also considering the Duelands. Maybe ever I will try them, or make carbon resistors myself with pencil stifts. Or try the Mundorf foil resistors...
 
You can change the gain of the amp module in the plate amp if you never use full power. This will lower noise floor a few dB. In the 3-way modules the 100W amp is already better by 2dB when I remember right.

There are ome things you can do with a passive crossover you can't with active (dampen THD of a resonance, lower power amp noise). But I would not go as far as dampening the whole speaker, especially when system noise is already not audible.
 
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First of all, 50 dB is anyhow quite a lot. It is a loss of about 8 bits. Do you ever use the -25..0 dB range during louder listening? If not, there is too much system gain.
@JukkaM wrote that it is for when the kids are sleeping and he is still watching TV. Do those 8 bits really matter in that situation? It will still drown in the background noise.
Secondly, I would always recommend to attenuate at end of audio pipe, not at begin (presuming clipping is avoided). By doing this on PC side you will worse SNR or more rounding errors in ASR and DSP processing...
Usually not an issue as the DSP works with greater temporary precision.
And input is 24 bit, so with a bit more than 8 bits loss you are getting below 16 bit resolution already at start of audio pipe!
Commercial recordings, studios and domestic listening rooms don't have a noise floor exceeding a 16 bit range.
 
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I think you are not using your Fusion optimally this way. Fine if it sounds good enough for you, but still...

First of all, 50 dB is anyhow quite a lot. It is a loss of about 8 bits. Do you ever use the -25..0 dB range during louder listening? If not, there is too much system gain.

Secondly, I would always recommend to attenuate at end of audio pipe, not at begin (presuming clipping is avoided). By doing this on PC side you will worse SNR or more rounding errors in ASR and DSP processing...
And input is 24 bit, so with a bit more than 8 bits loss you are getting below 16 bit resolution already at start of audio pipe!
@Julf got me covered (y) I posted this scenario to demonstrate the benefits of high SNR and overall superiority of digital interconnects in an extreme use case. My most used range is in -30 dB ± 5 dB range, when audio oriented friends come by or when watching movies with high dynamic range it can rise to -20 and above. There was one occasion where I set it to 0 attenuation, but that was with separate mixer controlling the volume and it was a private disco night where I wanted to test the capability of my sound system, which is made from PA stuff. The FusionAmps were not the weakest link there.
 
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Commercial recordings, studios and domestic listening rooms don't have a noise floor exceeding a 16 bit range.
You can be happy when a "loudness war recording" get's 10 bit of dynamic...

Btw - the ear only has 60dB of "natural" dynamics! It "switches gear" with the auditory ossicle to get at lower and higher range. That's the reason you can't hear silent stuff for a short while after loud signals. Psychoacoustics ... I love it! (You can read about masking effects, lot's of useful information and explains a lot)
 
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That's the reason you can't hear silent stuff for a short while after loud signals. Psychoacoustics ... I love it! (You can read about masking effects, lot's of useful information and explains a lot)
Indeed. That is how perceptual encoders (MP3, AAC, Vorbis etc:) work. I guess JJ Johnston (one of the fathers of MP3 and AAC) isn't on this site - he is an endless source of information on psychoacoustics. Some of his excellent presentations are online.
 
@Julf got me covered (y) I posted this scenario to demonstrate the benefits of high SNR and overall superiority of digital interconnects in an extreme use case. My most used range is in -30 dB ± 5 dB range, when audio oriented friends come by or when watching movies with high dynamic range it can rise to -20 and above. There was one occasion where I set it to 0 attenuation, but that was with separate mixer controlling the volume and it was a private disco night where I wanted to test the capability of my sound system, which is made from PA stuff. The FusionAmps were not the weakest link there.
I prefer transparency over cover ups... but ok... ;)
 
I was mainly talking about something else, but....

IMHO seeing everything as added noise is an oversimplification. The calculation errors in ASR chip can be data dependent. Same for PC side, but probably here at least some dithering is done to solve this, but at the cost of even more noise...

Anyhow, everybody is free to use their Fusion amps in whatever preferred way of course... ;)