#audio-tech
1 messages · Page 169 of 1
the dongle shouldn't sound any different between different computers, unless maybe the OS is messing with it somehow?
he doesn't mean volume
yeah i dont mean volume
Mega quality diff between 3 different machines using the same setup
could be messy sampling rate
Maybe apple has now "anti different OS" like Android diff
i think the macbook built in output might use the same dac as the dongle but a slightly stronger amp
or just usb meme negotiation
it should sound almost identical but be louder
yeah
Like, I would say my desktop gets 85% of the way of what I hear from the macbook
Maybe less
it sounds squished
like sample rate mismatch
I tried literally everything I could think of
what's the sample rate of dongle
removed drivers, downloaded mobo specific drivers
sample rate mismatch shouldn't really do anything as long as the sample rate on your output is higher than the source media/file
41k
just use 48 khz on the dongle
ok thats what i'd expect
some DACs are set up for one sample rate. so its possible that it samples up or down
maybe some weird audio processing going on on the windows pc
But it’s definitely worse on my Desktop, better on my wifes desktop, better yet on the mac
or gae usb voltage negotation
describe the diff
i think upsampling is supposed to be essentially lossless (with the tiniest bit of added noise), but downsampling is supposed to be worse
so that's why you just want to use 48 khz on the output
Clarity, sound stage. The difference between sounding less clear than airpods pro vs full sound stage and SUPER clear tremble
actual proper bass not just rumbles
it sounds like it’s compressed
i really would suspect some DSP accidentally being enabled somewhere in windows, or with some program you have installed
Like my m50x’s sounded better
does like treble etc gets turned down when massive bass hits?
no
ok so no ducking
I checked all configurables, toggled on/off everything
nothing changed
Updating drivers helped a lot
but still 85% or so
drivers ALL up to date?
like ALL drivers?
Yeah
even if its for the fucking printer 
i can't really remember what all you mentioned already
but did you try linux on the same windows pc (where you're having the issues)? i thought i saw something about linux mentioned
No, but I wanted to switch to linux due to having arguably easier time resolving driver issues
And window is just dumb with drivers
The motherboard is the asus x470-f
although i don't think there should even be a driver at all for the apple dongle
it should just be generic usb audio
It does showup as generic, so yeah
so that part shouldnt really be a big deal
Driver would make it x10 price xD
not possible
bcs. its digital
it takes the shit directly from the CPU
So is it possible my dongle is just borked?
no
if it works on other devices perfectly fine?
i can't really imagine this being anything but a software issue
i don't use windows so not really sure what to try
so, I do have voicemeteer potato, nvidia broadcast and virtual audio adapters installed
it can be Software / Firmware
which I wondered about the extra overhead
but driver refresh should ahve resolved those issues
And those aren’t tied to anything
Yez
i know that Voicemeeter doesnt really like AMD
stereotypical amd cpu issues (it's probably not)
idk if they fixed it
indeed it does not
It works for a lot of things, just not all
It can’t do high samplerates
So if you drop say your mic quality down it’s fine
$500 mic running 16bit 41k
and thats fine
could be software, ive faced this before with windows when connecting to a hub/directly to board usb c. somehow volume is lower than connecting to the phone so i just reset my windows and problem solved (for now)
thought my usb dac was cooked so i got another one out of spite and it still does the same shit
This is a fairly new install, less than 2 months
if u want we can hop on a call to try an troubleshoot
no way to find out unless you try but i guess this aint worth the time lol
might as well just cope with it
its not that bad yea?
Fun fact, it is nearly 1am for me and my typical workday starts at 5am, so maybe not tonight- but I’ll definitely take-up your offer another day/night
yeah go to sleep
maybe its all placebo 
I could just use a secondary boot, I just am fairly certain it’s the onboard chip being literal dog
yeah could be that also, or usb being usb
sometimes good sometimes shit
when my non audiocaring wife noticed a difference when I showed her (without mentioning it) I knew I was cooked
And a child!
nice life
twenty three year old reproducer
I could just steal my wifes motherboard. Same generation of AMD

bro is 2 years older and is Level 100 Mafia Boss
Im just a Level 1 Crook in life...
lmao not like she would notice
i just have a work
and life with my parents bcs its cheap
dont think it would change something
if hes lucky, its just windows, if hes not, its the board
I tried on her PC before the mac, that’s how I knew it was my desktop not the headphones
t the..
Windows 11?
Well, if I’m on an old bios and latest audio drivers it’d screw with something surely
yeah
made it worse
usb has a standardized voltage so you wouldn't be changing that
Windows 11 had 2 or 3 times issues with Audio
yeth
not that you can in the first place
Only after installing the official x470-f board drivers did it even reach 85%
prior was more like 45-50%
(Just raw windows audio drivers, no mobo specific)
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get that
gud dac
I’ll send my pj over
its R2R
dac cost more vs my whole audio setup combined
I could probably upgrade everything but my GPU for that
oh cool it's only 1.147,93 that means it's only 1 and 14 tenths euro
eh could get snowsky melody
i dont personally like it, but its a decent dacamp usb
but perhaps solving the actual issue would help, if you ever find one
I picked up a cx usb c dongle which runs at a higher power than the stock apple dongle
which should theoretically clean up the power moreso
I would bet my balls it’s purely just very dirty power
if that doesn't fix it then it's definitely software
idk if there's such a thing as dirty power
windows being gae part 150million probably
Prior I ran my m50x’s off my massdrop ctrl keyboard’s secondary USB-C port
which helped those specific headphones a crapton
didn’t do jack all for the ft1s
Another reason to switch to Linux
unfortunately might be worth
All my development environments would actually run properly on Linux instead of using a secondary system to do tests lmao
Or WSL. Second bane of my existence, windows being the first.
the dongle is a tiny amp/dac. if it sounds different on the macbook then there's something happening on the software side of things that's changing the sound. buying another amp/dac isn't gonna do anything. also fwiw using the apple dongle on any computer bypasses its internal dac/amp. plugging the headphones directly into the 3.5mm out on the macbook would sound even better than the dongle
what are you listening to music through?
could be the quality settings or an eq in the app you're playing from
Hii, i just lost my cmf buds pro 2, whats the best earbuds i can get for £50 max uk? Or £62 if they come with tracking as i would need to buy a smart tag 2 so i dont loose them again
Spotify
They’re downloaded at the highest quality Spotify will let me
what store would you prefer?
This was what I assumed, but if that were the case then updating drivers shouldn’t have changed anything unless the headphones were plugged directly into the 3.5mm jack?
Amazon or argos ideally amazon tho.
update
dude after replacing a cable connected left channel to R and L and right channel to R
and could not find it out 4 times
my MOBO is perfectly fine for pushing dt990 600ohm btw 🙂
no issues with listening music now at pretty koud levels (after he fixed it finally after fifth RMA)
you can pick one of these. my personal recs is the space travel
theyre all quite decent, the space travel definitely edges your old cmf buds pro 2 tho
@broken grotto I'm technically still in the rectifier and clamper circuit part of my electronics class, but I'm doing some reading ahead for MOSFETs, and it seems the active region there is about a 1-5% variance from linearity that's expected and amplitude dependent. Could that be a valid explanation for the about 3% variance I saw in gain ratio for the amplifier I tested? I haven't technically gone over this stuff formally yet, so I'm trying to understand it better.
What about anc? How are the on board controls? (Do they have any) Do they gave apps for control and do tgey use ldac
Im also wondering if i can get something with in ear detection
I really miss my cmf buds but i got them as a gift and can barely afford new ones
anc is alright, space travel is hard to beat for their price
and they do have decent on board controls, they use ldac, and have an app
iffy but its there
Alrightt, what about ied also how come the space travel 2 arent available here?
ldac?
is this space travel 2
some correction it doesnt have ldac but that shouldnt matter
probably import
the space travel 2 are still relatively new
Space Travel 2 are better
heck yes
in the store he provided, there isnt any stock
apparently he found another so thats fine
i dont know, i also wonder the same thing
it is woth
they are really recent
It probably changed some software processing. the "driver" may also have software doing DSP
I like my stack of schiit
If anything else it gives a nice spot on my desk to plug headphones into instead of having to buy a longer cable.
(And sounds better than the pc motherboard audio.)
Do you even know for sure that that amplifier uses MOSFETs?
I haven't had a chance to take the amp apart. My fiance uses it so she'd probably be upset if I did that lol
I need to get more amps eventually
Yeah Idk if I’ll regret it but I’ll listen to the wiser than I
I do think he is right
$120 for a Modi/Magnum stack
Amp model and maker?
Monoprice liquid spark
so, it's using both JFETS and MOSFETs.
but that variance you see is still measurement-related.
you even saw that with your sine wave tests.
if it was nonlinearity, you would have seen the same with the sine.
Both saw about the same variance of 3% in gain if I'm remembering right
bruh.
do the math here. make a table.
just to clarify
now you do it for the square wave.
apologies
i think the point still stands tho lol
bottom line, at this point im starting to think you're chasing a unicorn to say, "ah-HAH! I'm right!" but, cmon , 8mV?
Is that percent difference in the margin of error
if you want me to be honest i think the values he was getting was just dumb luck
I expect on really good equipment and skilled operator you'd see pretty consistent unity gain.
I just don't have a frame of reference for electronic measurements
1-2% might as well be considered identical in my field
in other words, yes
I am fairly certain those measurement values were pretty well within the scope's margin of error.
if you've never used a scope before, sometimes (especially on cheaper ones!) they're pretty sensitive to just about anything
you can breathe on it and get an extra 10mV if you really wanted to.
Or if the room were a few degrees hotter
I am all for electronic discourse but I'm pretty sure it was pretty definitive what it was between Golden and I lol
I even was able to recreate the same thing with Multisim, which I know he can do himself.
so at this point im fairly certain it's just chasing a unicorn.
I've known biooc for many years and he has always found creative ways to be wrong in the face of overwhelming evidence.
So this is just who he is.
its fine to be wrong when learning, that's how we learn
but holy cow accept when you're wrong and figure out why you were lmao
especially without a good foundation in analog electronics like transistor power amplifiers and nonlinearities, it's not gonna be a screaming answer on google.
Golden explained it pretty well, if it was a real behavior, then it would have been documented by now, especially in the audio space.
audio is relatively low frequency. I spent most of my analog electronics stuff in radar, which is a much different beast than the low frequency audio; you're looking at X-band and higher (9+ GHz).
weird stuff happens at those much higher frequencies, but those don't often translate to audio because it's just too low of a frequency.
(for reference, take the 20kHz cap on audio for humans. A full wavelength antenna that would pick that up would be 15 kilometers long!)
some audio players use the headphone cable itself as an antenna for FM radio
Those were the good old days
I feel like a lot of people mystify audio equipment a bit too much.
Some of the best designed amplifiers are a handful of parts.
Things get more complicated when you want an amp that's a few hundred watts but other than that
and for that, it works because most headphone cables are about, idk, 1-1.5m long?
which is riiiiiiiight about where FM radio's half-wavelength is.
Is the half vs full wavelength and efficiency thing
How low can you go realistically
1/4 wavelength is pretty common.
yeah, full wavelength antennas are most efficient and it decreases as you go smaller.
a half wavelength is most common
well
full wavelengths have their issues
Other than scraping the underside of bridges?
yeah, radiation patterns and stuff like that.
full wavelengths are very good at converting current to radiation
but the downside is for something like FM, an antenna is pretty heckin huge.
electromagnetic radiation... not the other kind
the music-through-air radiation
or comms like CB radio
or HAM if you're into that.
you'll see quarter-wavelength antennas a lot for FM radio too, which are about... 2.5 feet long?
are you an audio engineer?
I see ones that are about 5-10 feet on some trucks
No I'm a chemist
I took some analog electronics classes to help with instrumentations
Never got beyond transistors, RLC circuits, and op amps.
It's really all I need. All of my instruments are photodiodes, amp meters, voltage meters, and wheatstone bridges with some fancy stuff ahead of them.
I did a lot of analog and digital circuitry along with DSP for radars
it was very systems heavy
lots of cool projects tho
do you have spotify's eq enabled?
whatever is happening is definitively a software issue somewhere on your computer. the fact that the same, dac, amp and headphones are giving you a different output across devices is all the confirmation you need
Statement that describes what science is.
like. this is literally science
you have a theory, try it. either youre wrong (mostly) or you're in the right and maybe wonder why? xD
then you find the reason why you're in the right or wrong and of that you can learn more, adapt what you have learned. apply it somwhere else etc etc
with the apple usb c?
yeah thats normal
I guess I just don’t feel like it’s as good as my prior headphones at the current moment with the existing setup
whereas if I replace the ft1 with the m50x’s , everything else made equal- the m50x’s sound better
but if I move to Macbook pro the ft1s beat it soundly
not even comparable. Ft1s are very nice
Yeah, but it doesn’t sound any different toggled on/off assuming a flat EQ
Yeah I think you’re right
I’m gonna try through my keyboard on a different USB port
Only tried through the keyboard with the USB-C adapter on a 2.0 port, which probably isn’t ideal
@lone flame It was caused by depop being enabled in BIOS. This motherboard shares the filtering logic for the 3.3v/5v rail to prevent/help voltage drops- essentially making all USB power not great for audio
Now it sounds as good as my wife's desktop, but still probably not as good as the Macbook pro
Enabled by default
bruh
I think in this instance it was asus but same diff
i think it might just be a volume thing
get a dB meter app
Laughs in Mac
do a test tone at 1khz

try your best to approximately put your phone or dBmeter at the same distance your ear would be
get the same volume level from both headphones and take note of what volume % this happens in
and then compare the headphones that way
a lot of people compare easy to drive headphones to hard to drive headphones without volume matching and just pick the louder one
same with dacs and amps too
It wasn’t a volume thing
also I misremembered which headphones I had prior
They were the m40xs
I feel like you just don’t believe me when I explain the differences. You’re essentially telling me what I’m hearing is placebo and I’m gaslighting myself
Don't get my wrong, I really appreciate the help picking the headphones and generally the feedback- but I feel like you just straight up dont read 80% of the prior conversation or what I'm explaining. Which drives me nuts lol
I'm happy to anwer questions (Repeat or otherwise), happy to give information to help problem solve. I am not happy to essentially be told that what I'm hearing is wrong and I just need to get over myself and what my first impressions are
its okay dark, klaus is the local clown of the channel
don't take anything he says seriously
do it regardless
also wtf i didnt know we trained nils ai model to be able to send gifs
nils is a failed cleverbot instance we decided to put in the audio chat for spreading misionformation because we found it funny btw he isnt a real person
Compare audio -> get them close to the same loudness -> Record % = use that volume when comparing the two?
yeah
Aight, will do
the cx dac/amp usb-c adapter comes tomorrow too, I don't actually expect a super insane difference now that my stupid BIOS isn't screwing me but I'm definitely excited. I had a super cheap apple knockoff one that I tried a little bit ago today and holy crap was that horrible lmao
Klaus when someone hears something
I am bias towards u-green so I opted for theirs over something probably cheaper/better
they're probably almost all the same conexant DACs anyways
hey guys, im sorta interested in buying a record player, any recs? idk how pricing really works for these, is there really a good point to getting the ones that cost thousands? i wouldn't know
anyway i am probably looking to spend less than 400, but if there is a good reason to go higher ig i could
it literally doesnt matter, just make sure it has a counterweight and decide beforehand if you want it to have a phonopreamp built into it or not
whats a phonopreamp?
95% of the sound will come from the stylus you put on the turntable so thats the only part that actually needs to be good
Don't blindly trust ANY review website, but... this is a good place to start if you want to learn what you should look for in X thing. https://www.nytimes.com/wirecutter/reviews/best-turntable/
turntables provide sound in a way where bass is very silent and highs are loud, a phono preamp makes that into a normal sound signal
just go on ebay
buy whatever turntable that you think looks nice
with the condition that it has an adjustable counterweight
and maybe an integrated phono preamp, in which case a tt made more recently might be better
if you have a speaker amp there is a good chance there is already a phono input
yeah you dont need an integrated phono preamp
make sure whatever you buy either has no phono preamp in it or has an option/socket where its not active
actually fuck that it doesnt matter
if it does have a phono preamp you cant turn off you can just use literally any of the other rca inputs
great, so all that matters is that it has an adjustable counterweight then?
yeah
what you need to research is just what stylus you want
and maybe some accessories to help you set up the cartridge/stylus in a flat manner
theres a difference?
yeah thats the entire difference
i thought they all jsut used piezzos
and its a huge difference
no
there is also the entire low compliance vs high compliance tonearm debate
but you dont need to get into that
does the type of music i listen to matter?
ok im gonna watch some yt this clearly is a bit too in depth for a discord rec lol
and within those there are different quality ones
go ask in sonic visions
there are some good tt nerds there
you can also ask them for turntable advice but it largely doesnt matter what you get
but they might recommend some bang for buck tt with good suspension and easy to replace dustcovers etc just for convenience's sake
the sound that's produced by the little needle/stylus that rides the grooves of the record is extremely quiet. a preamp is boosting the volume of that signal while introducing as little electrical noise/white noise as possible. that signal then runs into an amplifier before being output by speakers or headphones
Thoughts on a Aune ar5000 + Fio k11? I don't really have a budget, I'm just looking for an upgrade from my He400se + apple 3.5mm to type c
I'd go for a Fiio FT1 Pro
question is
what do you expect form a upgrade?
its not like. investing more = better sound
its more likely. try around. find out. with luck you find what you like
I was not a fan of the he400se stock, I did an eq on it. Stock just sounds thin. Vocals sound like they're trapped under instruments rather than their own thing. The EQ for me adds thickness to the bass and makes the vocals pronounced.
So I guess I'd like something with more bass and better with vocals.
(This is my eq graph for the he400se) ^
Hey everyone, I need some advice on managing and EQing a pair of headphones. I’ve got Sennheiser HD600 open back and HD620S closed backs, and I want the 620S to sound as close as possible to the 600s. I’ve never done EQ before (in this scenario anyway) and I’m not really sure where to start. Someone suggested using Equalizer APO with the Peace GUI and maybe starting with an existing HD620S preset, then tweaking by ear. But again, having not done it before seems like alot of stress as it's quite important.
On top of that, I want an easy way to switch between the two headphones. Right now they both plug straight into my PC (Having to swap them out as I need to) I’ve heard people mention using a USB plus headphone jack setup, a passive 3.5mm switch box, or a DAC and amp with multiple outputs, but I’m not sure what would be best or which DAC/amp to get.
So basically just looking for how to EQ the 620S to get closer to the 600s and what setup would make switching between them simple without constantly unplugging. Any advice would be appreciated.
I dont think ar5000 fits your criteria there
I would consider ft1
Hmmm. If you're into vocals, maybe one of the Sennheiser would be better.
I'm always an advocate for the 6xx, but it doesn't have more bass than the 400se. It is warmer, but not bassier. But really good vocals.
Hd560s or hd550 might be worth checking out.
Sennheiser's sound? Would something like the hd490 pro be good? i just want something that would be a good gap from me he400se. I guess the best option would be to abuse amazon's return policy and try a few.
Seems like the hd490 has a little more bass
490 comes with different pads. One of the pads adds a decent amount of bass.
But that 2khz dip I think is what you're complaining about with the vocals on the 400se
And it's also on the 490 pro.
Sundara might also be an option. It has better bass extension and doesn't have the 2k dip that most other hifimans have
Better headband too
Ok I will look into it, thank you.
Hard to accomplish
Or lemme say. Nearly impossible
Firstly. HD620 is 500series chassis
HD600 is dirrerent chassis. Larger ear cups more space inside etc
Also it's closed vs open
So Acoustic impedance diffs alot
You could archieve a somewhat similar tuning
But there are toany factors changing the sound...
You can use AutoEQ to use the hd600 as a target
But at that point why not just pick an actual target
Yep I see
I get what you mean about picking a target, but for me the 600s are my reference that’s why I want the 620s tonality to be closer to them. Where’s a good place to start with that, like what software and how to actually do it? Also, what DAC/amp would make sense for this setup, and how can I make that the EQ I apply to the 620s doesn’t affect the 600s?
There are dozens of videos online on how to use autoEQ. Headphones.com and crinacle have the best videos imo so start there.
There's a database called squig.link that has AutoEQ integrated it into and it's what most people use. You use this website to create an EQ profile for a headphone and export it to your software of choice.
The software for PC is called equalizerAPO with peaceUI. Its clunky and annoying. There's probably a few other softwares that can do this too.
If you plan on getting a DAC/amp, it makes sense to get one that had hardware level EQ built into it. Something like a qudelix 5k or a qudelix t7i would be a good buy. They have an app that had AutoEQ built into it so it'll be as easy as opening the app and switching profiles.
Topping dx5 ii also has hardware EQ with profiles and has way better specs than the qudelix options.
Because theres no real way to have a headphone amp know what headphones are plugged into it, you will have to switch between profiles somehow. Or just buy two devices that are set to that eq at all times.
I was recommended to get the atom amp+ and pair it with my apple dongle? Or should I just get the fio k11. I'm going to go for the hifiman sundara for my headphones.
Hmm. I was less aware than I thought I was then. In that case I might just return them, seems more hassle than anything to have them. It wasn't a necessity just a bonus.
In that case, what amp/dac should I get for the 600s? 😂 Sorry for being such a pain.
Either is a good option. The atom amp is pretty good. It's downright one of the best amps for high sensitivity stuff like iems. The dongle will be holding it back in terms of power, since it only has 1 volt output. Should still be plenty for the sundara.
K11 is also really good value and has a digital volume knob, which means perfect channel matching across the entire volume range.
If you think you'll be upgrading soonish or listening to sensitive stuff like iems, then get the atom+dongle. If you want something simple that just works, get the fiio.
How much do you wanna spend and what are your priorities?
I’m not entirely sure what an amp/DAC does or what I should look for in one. I just know they’re supposed to be beneficial. I produce Dancefloor DnB and I might add studio monitors in the future (not sure if that changes what I should get). I don’t have a fixed budget, so I’d be happy to hear a couple of suggestions at different price points. Maybe something that’s also future proof if I do get monitors later.
Do you need an amplifier? How much power do you really need? And what misconceptions exist about hard to drive headphones in general?
Headphone Power Calculator: https://headphones.com/pages/headphones-power-calculator
Benchmark 'Think dB not percent': https://benchmarkmedia.com/blogs/application_notes/interpreting-thd-measurements-think-db-no...
What is a Digital to Analog Converter, and why do you need one? And how do they even work?
What is an amplifier?: https://youtu.be/--LEDxMs4yk
Read more headphone, IEM, amp, and DAC Reviews: https://Headphones.com/TheAudioFiles
Glossary of Audio Design and Measurement Terms: https://headphones.com/blogs/features/the-glossary-of-audio-measurem...
If you're doing pro audio, your needs might be different compared to the consumer hifi stuff that most people in this chat focus on.
So you'll probably want xlr pre-outputs on the back for studio monitors
But you can get away with rca -> dual mono TS cables. Most studio monitors should support that. So rca should be fine.
I don't do pro audio so idk what else you'd need.
But realistically, what you have now should be fine for the hd600. It's not hard to power and its high impedance means its pretty agnostic to whatever you plug it into.
At the high end, if you get a minidsp studio or a rme adi-2 they're really flexible and will be really useful for doing DSP to make sure you get the most accurate sound for your music.
If you just want something simple, get a fiio k11.
Thank you. That clears things up. One thing I’ve noticed is the 600s feel a bit lighter in the low end compared to the 620s, though overall I really like them. Would getting a DAC/amp help give an all round improvement, or would it sound more or less the same? You mentioned the Fiio K11 would that be a good neutral choice, or should I look at something a bit warmer if I want a touch more presence
The 620s does have more bass and more bass extension than the 600. So you're right in thinking so.
Getting a DAC/amp will not change the bass on the hd600 all that much. So it will be about the same.
And the actual role that a dac/amp will have on the acoustics is not huge enough to fix something like the bass rolloff on the hd600. You need EQ to do that.
The difference an DAC/amp will make is adding a subtle smoothness to the sound. Taking away some of the harshness. So it will probably be an all around very minor improvement.
And as for warmth, neutrality, and presence I'm not the person to ask for that. I pick my audio source equipment based on functionality and convenience. Ideally they shouldn't play much of a role in the tonality of the music.
Okay thank you. The bass thing hasn't been a major just something I noticed im comparison. My mixes seem to be fine regardless so it's whatever. Referencing exists anyway. I take it I can just leave a dac/amp for now then
If you're hitting the volume levels you want, and you don't hear any noise, cracking, popping, or distortion then yeah a fancier DAC and amp would be a luxury.
No need to buy something you don't need if what you have works fine.
i'd recommend checking out some audio interfaces rather than headphone amp/dacs. you get 1+ headphone outputs, speaker/monitor outs, and minimum 1 analog input for a microphone, guitar or synth. there's also an added benefit of dedicated drivers that'll increase performance in your daw, assuming you're on windows or linux
i don't have any specific recs. just avoid 3rd gen or older scarletts
Thanks guess Ill have to try look into it and see if it's worth grabbing one. I forgot about audio interfaces tbh so depends what it does and if it's worth it.
I ordered this today. Is it worth it?
do you like how it looks?
Typically you're supposed to ask that question before you buy it.
But I guess you'll be able to let us know after it arrives!
most people ask questions for affirmation of what they just bought
that's why half the time it's pointless telling someone if it's a good buy or not, they've already bought it lol
I was attempting to give a subtle bit of criticism for exactly this
it's just how people are, it'll happen until the end of time lol
Won't work sadly
i know. but it's free and worth trying just to see what happens.
something will happen. and it will be a learning experience.
but you could've gotten real speakers for this price
did you not have desk space?
@lean grove
I have heard good reviews on from people here in the server and someone recommended the brand anyways
lol
How many more times you wanna fail that?
most people's standards of good audio ends at whether it succeeds at creating sound waves
Then why ask us if it's good
I got a Bliss
Why would you do that
For my Susvara
I thought my Violectric V222 was endgame. I was dead wrong
The improvement is very substantial
Any time you need someone to tell you that you're wrong let me know. I'll do it for a lot less than $3000
I hear things on that amp in the far away stage that dont even exist in violectric v222
Nah its real
I heard like 5 other 3k+ amps. And thought it didn't matter
But Bliss is different
Sound stage is like 40% wider
That's impressive considering that the headphones are in the exact same spot on your head
yeah headphone mount position and seal make huge difference
Its 3W of power vs 10W
I've got some class D speaker amps I can recommend that are more than 10x that
Imagine a 400% soundstage improvement
You'd be able to hear things from the recording booth next door
yeah save for a small house/room dedicated for speakers and you’ll have much better experience
instead of spending so much for amps
plus real estate doesn’t lose money unlike amps, because the cheapo manufacturers are improving by year
Class D amps are not good for low level signals
I want a personal experience. Not speakers.
I have speakers. I cant play them loud or the neighbors complain.
foam exists ^ ^
10 watts isn't a low level signal.
OH you're talking about the susvara. Doy.
Yeah that thing only needs like a watt
Maybe less depending on what you're listening to
pretty sure if you build your own room for audio listening it’ll be the best
Yeah easily less than a watt
Ye I got it for Susvara
i guess he’s listening at ~125db
Yeah to produce sound it needs that much power.
But not to sound good
It needs more for that
i don’t think it’s good for da ears
even if it’s for short time
You dont turn it up
You just need an amp that has the headroom
Oh in that case I have some class D speaker amps with tons of power I can recommend you.
Speaker amps are not made for headphones
You can even hook them up in parallel with the amp you have now so you have an extra 90 watts of headroom
The noise floor is completely different
are you sure you just don’t have your headphones little off on your head, like for my ears if i angle the headphones forward and pull them forward i hear much more treble for a lot of headphones
Also designed to run under a different load
Imagine how much better they'll sound that way. All this extra power that you're not using just never being used
The amp itself is also way better
Oh well why didnt you say that
The engineering that goes into holo products is 10x what the competition does
I thought you just bought the amp for power reasons.
Nah. I'd get a topping then
If you want I can charge you 10x what Holo costs
If you make it 10x better
they don’t add any sound signature to the sound right
Bro. Legit only the cost of the parts already adds up a looot
Nah. Even Holo couldn't do that.
Bliss is mostly neutral to my ears
I am not buying it to tune the sound
If you get the chance to listen to a Bliss with Sus, I'd recommend it
Btw the holo dacs are a meme
Even tried the may
Send them to me I'll give it a listen
Oversampling literally beats any nos dac
Ok buddy
I'm not your buddy, mate
Lmao
Also.
More Soundstage from a dac doesn't sound real to me
the Holo Bliss is an amp
Does it has stuff to improve that
Like the AMP from Zähl
some sort of crossfeed?
Ye
no I don't think so
I was just able to hear very quiet details in the background way clearer on my Violectric amplifier
Like the Zähl H1 has the best crossfeed I have ever heard
I think its not quite called crossfeed but something else that they have
That's mostly Placebo
I did an A/B test
Blindly etc?
I legit was not able to hear a lot of stuff with my violectric, even when knowing its there
Also most people don't know how to A/B correctly
I volume matched
Measured?
no, by ear
Yeah no that's not really how it works
and for funs, I turned the violectric up quite a bit more than the Holo Bliss
Can be 2+dB of a difference
How did you A/Bd them.
Blindly
So it's invalid
A/B has to be blindly. Because Placebo does ALOT
In your head you expected that it sounds different
we are comparing a 1200 euro amp to a 3250 euro one
And that can cause lots of shit going on in our heads
I thought it was gonna sound like ass from what I heard from some reviewers
Funny that it doesn't mean it's better
Price does not correlate with Sound Quality
generally not
Some 👍
3250 euro for an amp that's objectively inferior to the topping a70 pro which is only $500usd 🥀
bro, an SMSL SU-1 has better SINAD than most dacs
Tbh the A70 and D70 has lots of stuff happening to measure that good
yet it sounds like shit
tho if we talking dacs, Holo audio literally makes the best measuring dacs that I know of
I'm not your mate, pal
Cyan 2 treble rolloff 
I did A/B cyan 2 to Chord Mojo 2
TIL non-oversampling dacs exist in 2025
with a switch
Only if you have inferior digital files.
A/B has to be blindly otherwise Placebo will get triggered with that switch too 🙂
Wasn't that Centauri?
With the shit performance to 16 bit 44.1khz files
no
no you can use HQPlayer to upsample
they possibly fixed it
Afaik any NOS DAC will have treble rolloff at 44.1khz
they released quite a few firmware updates
literally
onboard sucks though i think that one is pretty fair
Apple dongle
except on apple computers
the apple dongle is so ass
yet another apple W
Only if you have low impedance headphones like some loser.
shut up
No
I only use my 2000 ohm Sennheisers
i love my iems that can be driven by a potato batteyr
in europe we dont get as much output power on the apple dongle as in the USA
tube that mfer up
Sounds like someone's shouting at you at all times but I can confidently listen without worrying about output impedance
I did but the tube amp had the worst ground noise leakage I've ever heard
lmao
Made it past my fancy measure baiter DAC
Rip
average lowfi tube amp
Yeah xduoo ta-26s
pretty soon we're gonna see people using tube amps in 2025
:supershocked:
people are tho? they are great with dynamic headphones
that was the joke i was making
I should get a ta-66. And a jds labs synapse
Imo depends. Some tubes sound ass on some headphones
sennheisers are good with tubes
nah not necessarily
so is dt880
motherboard analog audio outs are the lowest latency solution for a pc
KSC75 with tubes
a lot of new boards are coming with usb based audio tho so we're latency cucked regardless
simply run everything at faster clocks
Ok guys back to work bye
just get a NOS, no decoding, just sound go brrrr
crazy low latency
why can i not run my usb at 10 GHz
This is the only tube amp I need
just get a Feliks Echo 2
latency cucks out
latency cucks when the 480 hz monitor has 2ms of latency
if only Cayin tube amps were not so noisy
latency cucks when the human reaction time is 200 ms
TWO HUNDRED CHERRY MX SWITCHES
some nerds are going back to using pcie sound cards in their main rigs in order to get lower latency
man
it's pretty but it's a scam
in what sense? the component cost alone already justifies the price
you are getting your money's worth
how so?
it doesnt do anything for you that a cheap dacamp couldnt
it's pretty yes but it's essentially overpaying for art
really well
i dont think a cheap dac can do r2r nos
this is not a dac
fiio k11 r2r
tho it measures like shit
not even an SNR of 80
XDDDDDDDD
It's not a scam any more than a Ferrari is a scam. Or a rtx 5090. Or any luxury item.
ferrari kinda is a scam 😔
those have appreciable objective performance characteristics
overpaying for a dac or amp is not comparable
cope
i think ferrari is kind of a scam yeah
latencucks need not apply
with dacs and amps you're paying more for measurements you will never be able to perceive outside of a graph
if you had infinite money that would make money worthless so you would have zero money
gota buy 50 other ferraris before you can buy the one youwant
this guy is stupid or something i swear
no you can get good measuring stuff for relatively cheap
and you cant do anything cool with it either
its not what you are paying for
the things id do for an r34
Why are you buying a Ferrari when a Ford fiesta can get you to work just as quickly?
Or why are you paying $4000 for a 5090 when you game at 1080p?
There is a real difference but any moron can come in and look at it and deem it unnecessarily expensive.
only if you spend it all
only if you hide the fact you have infinite money
i will pray that you guys escape the hamster wheel eventually
If you could go to Ali express and buy the exact same Holo bliss for 1/10th the cost in some no-name Chinese PCB, yeah it's a scam.
i would as long as they didn't blacklist me
It's literally just gray
i dont think holo stuff is a scam because of what you pay for the shit inside
I'm almost retired from hifi

or porsche
you couldnt get what holo may has in terms of hardware from anywhere else
why is everything about "good measurements + cheap?"
why not pay more for something just because it looks nice, or you like it, or something along those lines?
porsches are actually made well
yeah you can overpay for art if you want to, i bought a premium 5090 model because it's pretty
pushback against the boomer mentality or something
porsche makes actually driveable sports cars, mclarens lose their worth while being on the road
I got a new hobby to drop money on.
fr
you went from reddit to reddit
wow good on you
buy garbage quality because "paying more for something," is bad or smth
klaus is jealous because he's too obese for that bicycle
i cant believe we lost a man to fucking cycling of all things
I didn't buy it for measurements or looks. but for the sound.
this is why i say you get scammed
there is no appreciable sonic difference
but measurements cant quantify how it sounds
if you wanna buy it because it's pretty then go for it
that is where you are wrong
but youre deluding yourself and wasting money like an addict
could be worse
its got like 50 pcm filters stacked man
then invent a new measurement to quantify how it sounds :trol:
you'd have to be deaf to not hear a difference
have you tried it
i dont even know what device in particular he's talking about right now
what is it called and has someone measured it?
Not even cycling. Bikepacking.
I'm gonna slowly go long distances with a bunch of shit on my bike.
I've got a 100 mile trip planned in a couple weeks.
at least you are in a place where you dont bother others while on your bike
why does it need to be measured just use your ears
measurements only indicate something isn't absolute dogshit
i hope nils gets stung by some weird mosqiuto
they literally dont tell you anything about how good it sounds
you are not supposed to hear your dac or amp grey that's the entire point
@steel escarp i have a joke
all it's supposed to do is convert from digital to analog and then amplify without altering the sound in any perceptible way
i honestly can't tell if klaus and nils just hate each other or if it's mutual insults
what do you call a cyclist that goes over long distances slowly with a lot of stuff on his bike @steel escarp
🤡
Been the most fun I've had in years. Genuine fun. Climbing up a thin mountain trail with nobody for miles.
nothing ever happens and nobody hates anyone
a delivery guy, idk
bikepacking 😂
i hate everyone
my gif picker is a mess
keep yourself safe
i couldnt find it in time
disagree
nils the gigachad explorer
it alters sound though
there are actually cases where you want it to
lmao
first and foremost assuming we have perfect D/A isnt true
and also, just electrically it will have characteristics that modify the output.
second of all you dont need perfectly converted D/A to enjoy it
we should hunt down gray
troo
he has gone unpunished for too long
converting digital to analogue perfectly is impossible. you require an infinite amount of sinc functions to decode the signal
also this thing is expensive, not just literally but they are more than happy to call out some components used
grey said the same thing as me
so ill change my opinion instead
we have perfect A/D
but the question is whether the missing data is physically perceivable by human hardware
for reference, the caps they say they're using, Mundorf, i think are nearly $100 a cap.
the closest you can get to converting digital to analogue perfectly is by using extremely high taps filters. via HQPlayer. and for using that you'd ideally feed it into an R2R dac that supports NOS. but most NOS dacs measure really badly due to resistor tolerances. So if you also want clean. you'd have to go for a Holo R2R dac
yes but there's an appeal to chasing coloration through amplifiers. adding harmonic density to sound is virtually always gonna sound better to our ears, even though it's technically inaccurate. music is recorded, mixed and mastered with coloration/distortion in mind the entire time to produce a more pleasant result and amps are just an extension of that from the listener.
regardless, i think there's better means of reaching that coloration than relying on overpriced amps
There have been several studies on this. Current conclusion is that its still unclear what exactly we can percieve
44.1kHz contains everything you need, under the Nyquist Sampling Theorem. It's more the implementation that matters.
yes, but there is no perfect implementation
you can never get the 100% correct transients back
there's no such thing as a true transient, is the point
and afaik it's not believed to be perceptible
only if you use an infinite amount of sinc functions
but that would be impossible
but literally no cheap dac uses even close to a good amount of taps
idk the exact specifics or if i can explain it right
everything between your audio file and your ear hairs is supposed to be a linear system, so it's a sum of the individual frequencies that create the sound wave
we only have ear hairs that can perceive up to a certain frequency
so replicating anything above those frequencies shouldn't be possible to perceive
so it doesn't matter if we can't recreate a perfect transient because we just have to recreate the components of a transient below the maximum frequency we can hear
a few hundred taps would give you about 100dB of stopband attentuation, which is basically where it stops mattering
even in Radar we never gave a hoot past a few hundred taps.
someone let me know if this is accurate because i would like to reuse this explanation
I am not talking about accurately reproducing information above the nyquist frequency
Within the relevant bandwidth you can.
transients aren't some special magic signal, we can already recreate the parts of a transient within the 0-20 khz domain
yes, but those transients are incorrect
for it to be incorrect that would mean you have to be talking about information above the nyquist frequency
you can't reconstruct signal below the nyquist frequency properly without an infinite number of taps
well, real quick because I have a feeling about something rn: describe to me what the "impulse," you are talking about is in the time domain.
wow he had that on standby
I am talking about an issue in the time domain, yes
not the frequency domain
we start with the assumption that you can't hear anything above 20 khz or so
so we can just remove all the information above 20 khz
you can perfectly recreate that signal, that has the supersonic information removed, by sampling it at the nyquist frequency
I wanted to verify that lol
and that's proven
that's not just something we assume is true
im unsure what was said but take the opposite of what grey said and youll probably be correct
you need an infinite amount of sinc functions to reconstruct it 100% accurately. otherwise the output is taken at the sampled points, and then the stair stepped output is smoothed. but the waveform is malformed then
Again. Grey. This isn't the case for a bandwidth limited signal
it is
It's literally not.
....
bro
mate
.
chat should try mirroring 1% of my intellect
that figure shows the concept of sampling rate. that plot isn't even about a "number of taps," that's sample spacing.
Did anyone notice how it's the exact same sine wave despite one having more samples than the other
Crazy
ehhhhhhhhh. once you satisfy the Nyquist condition, the difference comes down to how accurately the filter is (with finite taps) reconstructing the bandlimited signal.
Lmao
I am not even talking about the passband ripple. I am saying you get phase distortion.
whenwe know the signal is band limited, we don't need any of the information between those samples at the Nyquist frequency because it can just be regenerated
keep shifting the argument bro
bro, the actual line is not what would be outputted. its how its sampled at recording in that image. the first line would be distorted
its still about how you cannot get an 100% accurate transient
due to the phase distortion and ripple
so you are now arguing that an insufficient amount of taps cause phase distortion?
which would only completely disappear when using infinite sinc functions
I was from the start
No. It wouldn't be. There's a single perfect solution for the line that travels through all those dots as long as you're below the Nyquist frequency.
Which it obviously is.
your hardware has to generate that line based off the samples it has
considering nyquist states you can perfectly recreate a signal if your sample rate is 2x i dont think your arguement stands
dacs aint shit
apple dongle is acoustically transparent
this assumes ideal conditions: the signal must be bandlimited, and the sampling must be uniform and precise with infinite precision
perfect reconstruction is limited by factors like noise, finite precision, or non-ideal filters
I just want to point this out, https://en.wikipedia.org/wiki/Finite_impulse_response
In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or response to any finite length input) is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely (usual...
mathematically its possibly, practically its not
oh i see golden
is isn't the whole criteria. It says you CAN perfectly reconstruct.....IF you PERFECTLY and infinitely attenuate everything after the nyquist frequency with an infinite-coefficient filter as @steel escarp was describing.
In practice we don't have infinite compute power so this isn't possible. Practically all DAC filters will either roll off early, attenuating treble, or will fail to attenuate fully by the nyquist frequency at all, often both.
Most DACs use 128-1024 tap (coefficient) filters. Which are "fine", but don't technically adhere to nyquist theorem.
Better DACs like Chord, Ferrum etc will have filters that do both leave <20khz content untouched and fully attenuate by 22.05khz by using higher coefficient count filters which requires much more compute power. (Hence the more powerful FPGAs in Chord DACs and the powerful SERCE module in the Ferrum WANDLA).
Then there's objectively much more accurate options like the MScaler with 1 million taps. Or HQPlayer which has filters up to a few million taps as well as quite advanced noise shapers which will also then give you higher dynamic range in the audible band.
Right now objectively the most accurate tool is PGGB, which uses the maximum number of coefficients mathematically possible with the number of samples in the file, AND an absurd noise shaper that has well over 600dB of dynamic range in the audible band even with 24 bit output. But it takes several minutes to process one file
Grey isn't saying that though. He's saying you need an infinite amount of sine waves to recreate the signal.
phase distortion still affects transient accuracy, even for passband frequencies not near the cutoff
no infinite amount of sinc functions
mathematically
Technically this is true. An actual 'perfect' reconstruction WOULD require infinite coefficients.
However in practice, infinite compute power isn't a thing, and there is also an additional limit in the number of samples we have available in the source audio itself.
So in practice, we can get as close as possible to a perfect reconstruction using something like PGGB. But then that can't be run realtime.
So instead you can use something like HQPlayer, but that still uses a lot of compute power, and more than could be put into a DAC itself in most cases.
pc sized dac when 😈
maybe one day. someone with enough brain power will invent a new reconstruction method with 100% time domain accuracy
phasure was on to something
We're talking about bandwidth limited signals.
he knows
yes
Infinite bandwidth isn't in the question
Lemme phrase this from my perspective of DSP doing Radar. In radar, we care about the stopband attentuation and phase linearity. 100dB was "good enough," because it cuts out thermal noise floors and anything further gave no practical gain, it's buried in system noise at that point. In practice, when we hit 100dB the filter was "good enough," and attention shifted to hardware limitations.
FIR Filters can be designed to be Exactly linear phase, which causes no distortion of phase pulses, only a uniform time shift. The tap count determines the transition width and stopband depth, not phase accuracy.
IIR filters achieve a sharp roll-off with fewer coefficients, but phase response is inherently non-linear, which does distort our radar pulses and makes matched filtering harder.
That's why we used FIR filters when phase was a big deal.
infinite bandwidth and infinite coefficients aren't the same thing
Why would you need infinite coefficients for a limited bandwidth.
If the Nyquist theorem can recreate the signal perfectly
to get perfect time domain accuracy in the phase
otherwise you get phase distortion and ripple
The thing is there's more to it than stopband attenuation, particularly in audio where we have such a narrow band between the audible band itself and the nyquist frequency.
You can design filters with shit tons of stopband attenuation but incredibly poor in-band response or attenuation by the nyquist frequency itself. And you can also design filters with essentially perfect attenuation at the nyquist frequency but with terrible stopband attenuation
Stopband attenuation itself is very rarely the issue. Most even quite basic filters will have >100dB of stopband attenuation, the issue in audio is almost always when, and how steep the attenuation is
Are you talking about ringing artefacts
Or is this something else
something else
ringing isn't really that much of an issue if you are not near the cutoff
yep, im not gonna disagree, audio is much lower frequency than radar (radar is 9+GHz, enough to give you cataracts if you're right next to the transmitter : D )
hehehehe one transmitter we had was 5kW.
anyway.
The issue would be much less a concern if we used higher sample rates. Even just 48khz.
44.1khz leaves us with only 2.05khz to attenuate by ideally at LEAST 100dB.
48khz would double that to 4khz
88.2khz gives us over 20khz of space between the audible band and nyquist frequency and so even really basic filters would be totally fine
okay actually this image is bad
It looks like regular distortion but the distortion harmonics are offset by a bit.
surely there has to be an image that shows the actual analogue signal (within the audible band) that is produced from lower sample rate vs interpolated during reconstruction
however, @steel escarp is talking about phase
This is a typical filter response (the fast linear filter in an ESS DAC). It has plenty of stopband attenuation (also keep in mind here that 0dB on the Y axis is normalised to the level of the white noise on the FFT which is about -60dB. So technically this is actually more than 150dB of stopband attenuation.
BUT....it doesn't fully attenuate by 22.05khz.
woah
my matlab script nailed that lmao
Instant attenuation at 22.05khz
yeah, which more comes down to implementation. that's a high quality DAC filter right there.
my concern is the argument about phase
DAC filters are typically either linear or minimum phase, which means that any behaviour that isn't perfectly minimum or linear phase will be as a result of quantization noise. But that's where stuff like Chord's modulator, or PGGB/HQPlayer's noise shapers come in.
I give up. I cannot find a picture that shows off the phase distortion I was talking about
The way I check for time domain accuracy is by generating a 1fS signal which is a sum of multiple sinusoids (different frequencies and phase), then running it through PGGB to upsample to say 16fS, comparing the results directly with the same signal generated at 16fS and finding the maximum absolute error. This is all before noise shaping. The current PGGB at 64 bit precision is accurate to the 10th decimal digit. I.e., the maximum absolute difference between the upsampled signal and the signal generated at 16fS rate is 1e-10 or 33 bit accurate. At 128 bit precision the accuracy is within 28 decimal digits or 93 bits and at 256 bit precision it is 55 digit or 193bit accurate.
Which is nuts
pretty sharp way to test that. But, what about consumer dacs?
the more "normie," ones.
Consumer DACs it's not possible to test the modulators themselves since you're bottlenecked by analog domain noise.
Even if you have a beyond-physics perfect modulator, no getting around johnson noise.
You could if you want technically run a days-long FFT but to get to proper low levels it's going to literally be days, and you'd need to set something up to even run a billions/trillions of points FFT as well so that's not something I've done 😅
Some manufacturers like Chord are pretty proud of their modulators so they've released some stuff. This is the Chord DAVE 1-bit modulator resolving a -300dB signal for instance
sure. im framing everything from radar DSP which i know is a different beast, even though the same theorem apply. Most of the radar DACs i made were aimed for 10-14ENOB.
But the filters themselves are of course pretty trivial, so I measure/test those whenever I measure a DAC
I just have a very hard time believing in the "phase distortion," being as audible as he believes, is all.
NOS dacs sound softer for a reason. besides the treble rollofff
many things are inaudible
That's something which is a bit tricky. Personally, I'd argue it IS audible, even though logically it's a bit hard to explain why when you're looking at such incredibly tiny differences. But I have successfully ABX'ed between different modulators where the differences are all below -100dB, and plenty of others have shared their experiences with the different modulator options on dCS, L&P, Sony DACs, or in tools like HQP, questionable as sighted impressions are ofc
why do we need a filter at all? won't our ears create the sharp lowpass necessary?
no
Without a reconstruction filter you get -3dB treble rolloff by 20khz
howdoes that happen
gosh i love signals, systems, and DSP.
Because there aren't enough samples at those high frequencies.
Again, the big thing people forget is that nyquist theorem is based around using a reconstruction filter. Without it, you aren't just 'not removing the aliasing', you're also not adding the extra bits at high freqs that SHOULD be there
you get sinc droop
put that in a textbook
a good reconstruction filter accounts for it
nyquist isnt a real frequency though
ok?
If it were important you'd think there'd be more graphs of it.
there are just mirrors of audible frequencies and inaudible ones around it
huh
that sinc droop is quite a bit bigger over hundreds of MHz in Radars, but our pre-comp in DSP fixes that.
the term nyquist frequency means the frequency of the sample rate / 2
its a real frequency
if you're to lowpass a simulation of this in music software, you can lowpass below nyquist and preserve the lower frequency
I wasn't wrong tho
rare W
but were you right
why would our hearing be different from this lowpass? and why would putting an electronic lowpass slightly above 20k cause any intervention
Sampling theorem.
If a signal is bandlimited to a maximum frequency f_max, then it can be perfectly reconstructed from it's samples if and only if the sampling rate, f_s is greater than twice the maximum frequency.
this is already the case
why would that play any role between ear lowpass or electronic lowpass
my ears are bricks
to prevent aliasing, you use the 20-22kHz region as the transition state for the brick wall, thus preserving the nyquist theorem and preventing aliasing.
i rub rocks together for music
placing the transition at a lower frequency just steeply attentuates those above it.
I didnt know u made rock music
no. the theorem only actually requires total filtering to the opposite extent that frequencies exist in the audible band
eg if you can hear to 16khz and nyquist is 24, you must completely filter things out by 32khz
nyquist is for any signal
the ear accomplishes this
nobody hears to 24.1k when nyquist is 22.05
so what makes it not shooting ourselves in the foot trying to electronically lowpass?
I have a demonstration of this on my pc somewhere
bro clearly does not know what hes talking about
I am well-versed on pcm audio
i have a pc somewhere
it quotes this because an ideal lowpass is a guaranteed way to give a perfect reconstruction
if you consider brickwall filters per analog electronics perfect enough, then ideal lowpass is not the only way
For example, let's assume we're digitizing a radar pulse with a maximum bandwidth of 40MHz.
- By nyquist, we need a sampling rate of 80MS/s. This means our sampling frequency is 40 MHz. (it's real.)
- We assume the radar pulse spectrum is strictly 0-40MHz and nothing above that.
In reality, real radar pulses aren't strictly band-limited. There will be leakage above 40MHz. If we sample at 80MS/s without filtering, those higher components alias back into the baseband and corrupt the signal.
this is why audio interfaces are good about lowpassing their outputs. they'll be sent back to the inputs/adc's
it is not the same as a filtered adc or in this case our ears
what we do (and what I did,) was insert an anti-alias low-pass filter before the ADC.
- Passband flat above 38MHz
- transition 38-40MHz
- Stopband > 100 dB attentuation above 40MHz. This is the guard band and filter steepness requirement that turns the theorem into a working system.
ideal lowpass doesn't exist. The only mathematical way is a brick-wall low pass. everything else we build is just an approximation.
so why would we need an electronic lowpass if our ears also lowpass?
I think I may have a misconception about the nature of the ear's lowpass, or if humans have the audible band affected by ultrasonics
that's what I'm trying to learn
there's no aliasing. you just don't hear it. anything above Nyquist aliases back into the baseband.


