#audio-tech
1 messages Ā· Page 181 of 1
sweet, this looks decent, thank you. when you say 30mW power, is that like coming out of the headphone port?
30mW is what your motherboard can deliver to the headphone
https://headphones.com/pages/headphones-power-calculator this is a tool you can use to calculate how much power your headphones need
it needs about 63mW, but that is like the minimum, its better to at least have double available for the best sound
which the Fiio KA11 gives you, at 200mW
I see. does that make the sound louder too, or just provide more power to make a wider range of sound?
you could get it louder in total, but you'll still control the volume via windows
you'll still have full control over how loud you are playing
okay, I'll grab this then. thank you for your help!
didn't realise proper headphones were this much of a hassle to get right š
the pay off will be huge tho
what will actually change? more bass and anything else? also would there be any sound degradation if I bought a USBC extender in the future? like female to male
the extender shouldn't cause noticable degradation, as its just power and digital data that is transfered over it
the bass will be noticably fuller, and you have more headroom to EQ
beyond that, compared to your pc's headphone jack, I think you'll notice it sounding better in general
nice! glad you like it. It sure is impressive for it's size, also kind of indestructable I've dropped it a lot
iām dumb. i accidentally bought the first gen. no wonder it was 80 bucks
i might return and buy two 2nd gens for stereo
Dang, think you'll like the 2nd gen it's more balanced, clearer treble, better codec, and less distortion
i can get two for 220 usd
with the weight, i can easily pack 2 in my luggage
solid price, I haven't tried 2 but it should open up the soundstage a lot
the only thing is, how useful will stereo sound be while traveling
i already have a proper 2.1 in my home so i wouldnāt use these
whats the best 60-80 dollar cad headphones for gaming
might be better to get 1 unless you want it for parties or something
thatās what iām thinking
is it also worth 40 dollars more for the 2nd gen
i get a 3 band eq and 5.3
for 220 i can get a refurb soundlink max
anyone have thoughts on the edifier MR5s?
they seem like a pretty straight forward upgrade from the MR4, still having trs, rca, and then adding xlr and LDAC support for inputs
how big are those
Once again the Sennheiser cable has started to disconnect on the right ear piece, it's starting to drive me insane... I take good care of my stuff, yet it keeps doing this with regular intervals, it's starting to get rediculus
Starting to wonder if it's about time I start maybe looking at new headphones at this point if the cable is gonna keep doing this to me every few months :/
the max was kind of a step down tbh, returned mine. Too boomy
the EQ is pretty decent, filters seem to be in the right place (to my ear only)
mines on -2 bass, +2 mids, +1 highs
What's your guys' opinion on the Beats Studio Pro for consumer headphones?
not as bad as the other beats, not as good as other headphones for the price
the 1000xm4 is 42% off on amazon for a bit more
What stands out to me in this review is that they say it's not comfortable and doesn't fit well. If you have a best buy near you it's probably best to try it on for yourself before committing to that purchase. Probably not a good headphone to wear with glasses on if that applies
beats
studio
What???
people like what they like, accept it, move on. The point of this hobby is to enjoy music. I feel like the community has lost sight of that
what does studio mean that would make it mutually exclusive with beats
bro you produce musical beats in a music studio
I did some searching.. it's not worth trying to fix if the socket is broken
other than cleaning the contact points or sending it back it's probably better to get a new one. If the cable is the issue try hart audio cables
jcally jm20 max costs 20
itās less clunky
still overheats
but itās less clunky
bit late now š either way it would take weeks to arrive from aliexpress
thats the only place that sells it in my area
for an extra tenner i dont mind something similar arriving tomorrow
well, it is just better imo, the form factor, you can carry it in a medium sized wallet no problem unlike the fiio
im not planning on taking it outside my room
and i have a secrid wallet which can't really fit anything in it anyway lmao
also if you get an unit with old fw i recommend updating it
you may run into some heat related issues in the long run
was not aware these things had firmware i thought it was electronics only lol
but yeah i'll do that
tbh itās expensive
even on sale
firmware is way more common than analog electronics now tbh
I 'member the days of analog control systems, where we had RLC cards you slot into things to adjust the controller.
nowadays you just adjust Kpi and call it a day.
return to vacuum tube
Preety sure "now" is the past 15 years
At least
it depends
some industries took longer than others.
also for some time when microcontrollers weren't as powerful as they are now, it made sense to do some things analog still.
or in fabric
I've been doing it a long time lol
even nowadays there's reasons to use analog, but more and more consumer products prefer to do things with a microcontroller for many various reasons
I've built a many state machines using discrete logic IC's and gates.
once you started moving into digital filters, modulation, and stuff you started to see more digital MCU's, but still used analog for latency sensitive applications and high-bandwidth.
in radar we often had analog frontends with digital backend processing, for example.
Interesting
really the use of MCU's in consumer products is 100% cost.
that MCU costs less than a dozen opamps.
then you add in the "push out out the door," mentality, where you just get it to work and then promise to fix it with firmware
cant' do that with analog lol
in an audio sense though, why do they use FPGAs sometimes?
i can snag another gen 1 flex or 80. not sure itās worth though. i hear the stereo pairing is a pain
For travelling 1 is prolly enough id say
yeah. maybe iāll just stick with the gen 1
TikTok
233K likes, 4194 comments. ā#fyp #spotifywrapped #spotify #musicstreaming #applemusic ā
I got them for $130 from target. I was just wondering what other people though of em. Thanks for the response
What?
I think this guy is just coming to the shocking realization that studio is a marketing term
The only substantial outcome of asking what other people think of what you bought is you feeling worse
Enjoy your headphones. Don't ask other people what they think about them.
@steel escarp the DAC arrived, and i've already noticed its a lot louder now
the bass does sound a lot less flat now
they're not that bad for beats but still $130 is imo too much
They're normally like $300 som
yes, price doesn't mean quality
there are some horrendously overpriced headphones out there
including most of beats
250 too much
good
even a 40 dollar dac/amp dongle can make a big difference compared to your default pc output
yeah it sounds a lot nicer now
I went a bit far off the deep end with my dac/amp
I may have spent 4700 on the pair
but man does it sound nice
what are you noticing so far?
u liking the Hifiman Edition XS so far with your music?
yeah its really nice, infinitely better than my shitty ass gaming headphones
the sound is much more vibrant imo
and games feel more immersive
š«”
I'm pretty glad I got the Scarlett Solo Gen 4 instead of the Gen 3, having the XLR cable on the back is so much cleaner
its okay.
but really REALLY bad for headphones tho
Are there any cheap audio amplifiers worth a shot? ut to 50$?
lid on my earbuds randomly broke :(
case never felt solid in the first place, maybe I teardown the case and 3D print my own one?
rq
My dog's a d*ck puppy and chewed on my headset wire and now it isn't working. How can I fix it? It's the wire to the Seinhiser HD600 headset
It is currently seemingly a small puncture with an exposed wire.
isnt it just a 3.5mm aux cable?
Not quite, it's an aux that splits into two wires that plug into the L and R sides of the headset
The R side is what is currently broken
I looked up "sennheiser hd600 replacement cable" and got results pretty easy
any less than 30 dollars? That feels insane for a basic cable
solder it
I can try. Would it also work if I tied the exposed cables together?
idk if solder would introduce weirdness to the audio
yeah but you should cimp it or solder it instead
your headphone cables dont come out of the factory with welds on it
My dad probably has some wire crimps or something, I can try seeing if that could work.
why do you need one
i bought some IEMs and i was thining about changing my Arctics nova for beyerdynamic dt 770 pro
you dont need an amp for this
also dont but a dt770
no dt 770
Why no Dt 770?
Got them exposed enough to maybe do something with them
Idk what the tethery wire is for, protection maybe?
Maybe I am doing it wrong but touching the tips didnt seem to produce audio
Yo I bought Sennheiser HD630s, now I am having buyers regret because i got randomly recommend on Reddit that they were not that great and not comfortable, what are your guys review, while i was looking into it they seemed great some even ranking it higher than Air Max
i think that shows how bad the air max are instead of hd630 being good lol
if you dont like them then nothing we have to offer should impact that.
They still to be delivered
Try them out for their return period. It took me a bit to get used to my HD600's from Hyper Cloud X II's
š
wait for them to be delivered and try them out.
Hello recently I got a blue yeti nano and I have Turtle beach stealth 600 headphones, however I want to upgrade. Does anyone have any suggestions for a good pair of headphones (doesnt need to have a mic built in) that has really good audio directionality. I want to use them to play games like war thunder where I need to know where the sounds coming from. Please try and keep them under 120 usd.
I would also want them to be wireless
its gonna be hard to get something good and wireless at that price, if your headphones still work, I'd stick with them
any significant upgrade is out of your budget
Well I would like to still have options because i can save for it
HyperX Cloud III S Wireless is actually not bad
but its like 130 EUR
actually its on sale for 120 usd
Is the audio quality better then the stealth 600
they are a pretty big upgrade over your turtle beach
The HyperX Cloud III S is one of the best stealth releases of 2025. It takes everything great about the Cloud III Wireless, but adds Bluetooth, better sound quality, and more! In this review I cover every detail to help you decide if it's worth the upgrade!
Purchase Links:
HP Direct (No affiliate) - https://hyperx.com/products/hyperx-cloud-ii...
screw it ill get them
make sure you get the correct model
the Cloud III wireless S specifically
@sand grove
The Cloud III S Wireless marks a new evolution of Cloud III that features a variety of wireless options such as 2.4GHz, Bluetooth, or Instant Pair with select OMEN laptops. It also offers extended battery life and enhanced customization. With HyperX as a proven leader for comfort over decades, th...
These right
that should be it
there are multiple variants it seems?
seems cosmetic
cool there is a white option, if you enjoy that
Probably wouldn't stay too white for very long lmao
yeah... white is often a mistake for gamer setups
apparently there cloud alphas are on sale too
wouldn't get those
I looked at the measurements
the Cloud III S seem to be the most coherent out of all of their lineup
in terms of how natural/balanced things will be, tho with a bit of enhanced bass, since its for gaming and all
Alright ty
overhyped
logitechs are less...
name a better logitech then
I think I would take the cloud iii s, seems to have significantly better bass extension
and seems to isolate sound better
the logitechs have existed for much longer, i'm not saying the III S are bad but they cost 1.5x more
read somewhere the hd 560s āscales well with a good ampā i know nothing about audtio tech what does this mean
some people believe that there's hidden performance to unlock from some headphones by giving them a fancier amplifier.
but headphones aren't that complicated. the 560s is remarkably easy to power, so I think the people that say that just like the way that those headphones sound with that amplifier.
there's no reason to believe that theres a hidden dimension to a headphohne remaining uncovered until the right amplifier comes along.
@wispy gate earfun uboom x is not good. just listened to it on several eq profiles
how do 4 drivers sound worse than 1
a much bigger casing with more power
and it sounds bad
virtually no upper bass
i can make 50 drivers sound worse than 1 no issue. its remarkable easy
and not as much clarity in the vocals
intentionally sure
Bad tuning
Bose has insane engineers, world class
i hate how they appeal to normies who love low end boosted bass
a lot of bt speakers have such a v curve
yea that makes sense
even with 10 bands, i canāt amplify the upper bass
at least from their marketing materials š
still canāt get a lot of things right on their ābudgetā headphones
Their headphones are just fine
Apple and sony have billions to throw around still no eq still cant do anc like bose can
the airpod pro (3rd gen) has pretty good anc for what it is

Bose is
in the Audio world
Genelec has World Class Engineers
Bose ANC is impressive
Replies with a speaker brand š¤”
Bose released the first ANC headphones ever in 1989, Genelec makes speakers for people who think money is a personality
@steel escarp listening to all my music again has been fun lol
genelec does make good stuff, but theyāre ridiculously expensive
Ikr, everything just sounds good on them
if they were 70$ then up to the hype, but <120$?
also depending on the reviewer that 9kz can be up to 12db louder than 4khz
nothing horrible but may be worth considering
Considering its for gaming, the boosted upper treble is useful
are you sure about that
did you see some of the things they've done with dsp and weird full-range drivers
its close to the preference bounds so
guys I need best headphones under $300 that can be wired and bluetooth at the same time, I was thinking sony xm5 for $250. Use case daily drive, light gym, hardcore gaming, work meetings, any recomendations?
Discord thinks you are a spammer 
discord just hates me
is it true the hifiman edition xs donāt really need an amp these days for most of the sound quality? (on windows desktop)
my audio codec is a realtek alc4080
can't confidently say
but $20 dac will work for sure
no need for a $100+ dac with xs
i guess ill buy one with the headphones and test it out to see if i need it or not
a guy in here bought a Fiio KA11 for the Edition XS, and the difference was notable compared to the pc output
Yea its just such a ridiculous comparison, their desktop monitors are like $7000 and he's comparing them to headphones lmao
If we are talking who has better engineers Ill go with the people who actually invent things
Bose's new beat - http://www.cnet.com/news/bose-behind-the-scenes/
Bose spent years developing the ultimate suspension system that offered a super smooth, magic-carpet like ride. It was a technical success but a commercial flop. See why.
Never miss a deal again! See CNETās browser extension š https://bit.ly/3lO7sOU
Subscribe to CNET: http...
I like bose as a company
do you listen to music on your suspension
sometimes
You could go for Beats Solo or Studio Pro
I like mine
imagine
depends how loud you listen, really
they are inefficient/quiet headphones, and I have killed a headphone jack by running it at max
make sure whatever you plug it into is rated for 16 ohm loads if you're gonna play it loud
invented a system that cannot be incorporated into anything
yeah
soft suspension isnt magic
and electromagnetic suspension essentially achieves that
statement was to be derogatory due to counting it as soft suspension
anyways the technology is really bad
So is your argument, bose has made a lot of impressive advancements in multiple fields
Objectively they have good engineers, sorry if that hurts your ego
Snobs are the worst part of this hobby.
cant make good without making bad first.
sincerely, former product R&D
anc yea
else nah
bose is one of those companies that will let their engineers do wild stuff for no reason
it's actually a great company to be a part of.
they are really good at anc and wireless transmissions
else they are kind of a meme
they are a meme even in audio with some of the shit they made
Then why are you saying they have bad engineers
bro got warshipped
Gets his personality from hating on things smh
bose 802
From 1 product that isnt a headphone š¤”
what headphone would you recommend from them?
overpriced anc meme 1
or overpriced anc meme 2
maybe overpriced in ear meme X pro
Qc ultra is fine with eq
mid headphone becomes less mid with eq
good to know
how much does this set you back again
they aren't a good company to buy anything from nor has their engineers achieved anything in regards outside of anc and transmission
Im sure if you change the topic enough you can feel like you're right about something lil man
Moving goalposts again - anc and transmission dont count anymore
People really put up with you irl?
saying I am moving goalposts meanwhile 70% of your messages being ad hominems
š
we live in a society
šŖ
he might just be drunk tbh
You could take all that genelec money and pay some people to hear all about what brands you hate
But Im not wasting my time with someone who acts intentionally obtuse, cheers
wow why are you moving goalposts
i never said we should talk to people about stuff we hate??
are you building a strawman??
omg
get your clown ass out of here
rename yourself to wikipedia.org/wiki/List_of_fallacies
Celebrating 60 Years of Sound. Explore Decades of Innovative History from the 1960s to Today.
what a rich history
Damn we clowning in Klaus? Lemme join in. Gimmie a sec to read the messages.
Bose spares no expense to make some of most weirdly well engineered mediocre products.
Like they give themselves these massively stoopid design constraints and do a real good job of making the thing, all to deliver something unquestionably mid but with great profit margins.
Cant forget how they just made expensive smart speakers paperweights this year
like hundreds of thousands of overpriced soundtouch crap just went poof
with no local option besides aux in
anyways bose is just overpriced underperforming made for the masses gear that rides on the brand these days
bose is definitely capable of making good stuff, but most of their stuff is pretty mediocre
this trend follows suit in car audio too. sometimes they can make some exceptional systems, but most of the time theyāre average
should have made it stiff maybe some autist in le mans would have used it something
and then the fia bans it a week later or something
@warm scarab tbf, it's probably more bad company culture and business side that make Bose so trash. Their engineers probably know what they're doing at least
i dont know man, when have they invented something unique?
But not much you can do when the average middle manager has an IQ lower than their own gpa
They probably aren't tasked with inventing things though
okay then what makes these engineers special
The company culture is probably aimed at making them find the cheapest way to complete a need
then how can you come to the conclusion they are good engineers
is what im getting at
I didn't say they're good
im saying general argument wise since that was the goalpost
not that you've personally said it
Just that id wager it's probably the management side of the business that makes the company so bad
yeah sure
a lot of bright minds everywhere but you can't say they are very bright without results
@warm scarab these are the results that matter
it even sucks in the trash!
is anyone here an expert on those stupid 2012 sonos play:1 speakers? i also have a sonos connect.
also DMing me would be easier, im trying to connect it but it keeps failing, and the sonos app cant find the sonos connect so i cant connect to that
bose was still a major force in getting a lot of low frequencies from small drivers and small enclosures
they also somehow got a lot of high frequency use out of specialized full range drivers
alongside dsp shenanigans
I respect these achievements and their positive effect on the bt speaker market mainly
it just doesn't mean that such a bose product can replace real speakers
I wonder if soundcore is a good buget brand
got my lttstore order, but i already have DT 900 Pro X, would be this be worse or equal to it ? (mostly in terms of gaming, especially shooters)
Can you return them in case you don't like 'em?
Whats wrong with your 900 pro x
absolutely nothing
im from lebanon, at most i could sell the Sennheisers
What do you fellas think of the Bose wave cd player music system?
I think the performance of those two headphones are similar. The big difference would be preference
yeah, thats what ive been seeing online
might just open them and try them for like a day or two
just use spotify
hell nah
why not
because youtube music/youtube premium is a way better deal
also isnt spotify right now like $15 a month ?
okay
valid
i thought you'd try to justify buying the cd player
lol no, better off just downloading high fidelity audio files
It's also got one of the lowest quality streams on the market
#teamtidal
oh, do you already have them?
try them
yupp, just tried it for 1-2 hours, dont like it honestly
feels cheap compared to the DT 900 Pro X
i find them light but not necessarily cheap
though the pads do like to pick up every bit of dust
hifimans or oneodios is what i find cheap and behave cheap for example when you extend and descend the headband slider few times the plastic parts can start to chip and leave this microplastic litter
like hifiman likes to use some of the cheapest lego feeling plastics ever
pads are more comfy on the beyers for me
it doesnāt matter much really but⦠it is to consider
although the clamp can be a bit strong when you first get it
the pads on the 1.8m cable version of 560s are not good just ok
do you have the 1.8m or 3m version?
the 3m version also doesnāt have the best pads but probably lil better than the 1.8m version
padding is also different on the new 1.8m version
also the 1.8m version can sound a brink different than the 3m but itās not significant, but from reviewers it seems to be at times worse than the older 3m ver.
Guys
What mic should I get?
Live streaming and recording and content creation and gaming
Budget 100ā¬
Samson q2u or a atr-2100x with a boom arm.
tidal ui is ass
i have the q2u. great budget mic
donāt get a shitty boom arm. it will fall apart
Kinda clean. Hope u never switch dacs tho
Because they sound good, are pretty easy to use, and have both XLR and USB so you can upgrade later on with a dedicated audio interface.
it's 2025 bro
cd's are a waste of money
if you want to be retro get a record player, at least you can find the records for it
you mean 1.8 meter cable ? if so, yes
box literally only came with the headset, nothing else
Should i upgrade from my razer kraken v1ās after 11 years of abuse?
the newer one is sadly imo bit worse
(1.8m version is new)
well "new" it's like 3 years old now
Vmoda boom mic pro. It has the best over all convenience
Any suggestions for an audio mixer? I currently have a USB DAC/Amp for my PC (Scarlett Solo Gen 4) and want to be able to also listen to my Switch 2 audio at the same time
Don't want a capture card because of the added latency, and my PC's mic-in port has a ton of static noise
Feature-rich and delivering outstanding sound quality, the Yamaha MG06 analog mixer is well-suited for ultra-compact mixing rigs. The MG06's quality op-amp ensures that your mix sounds transparent and articulate. The MG06's crystal clear D-PRE preamps will squeeze every drop of tone from your mic...
yeah felt like it, they're selling it for 99 pounds on the UK website
what i saw yesterday
Discover the unique acoustic performance of our HD 560S audiophile headphones. Zero distortion ā Rich bass ā Clear treble ā Now available from Sennheiser!
for anyone interested, seems like a great deal to me
Doesn't matter anyways
yeah
i just remembered some people here thinking it was false advertising and not lossless
No.99999 of companies using misleading/downright false advertising/claims
huh
anything for a few extra $$$
In their defense "lossless" could mean that the formatting they use sounds to people like raw uncompressed files
I didn't
#audio-tech message
I guess this?
Probably yea.
I think it's fair for golden to mention that spotify is technically not lossless, since its competitors can run in exclusive mode just fine to bypass the os mixer
Yea
you could say other OSes not protecting your speakers is worse
wait really? 
I don't think macos or linux do, I imagine for most hardware you'd damage your ears far before damaging the drivers
Isnāt Lossless 100% not 97%? Nvm, didnāt read the second to last paragraph
š
Anything smaller? I honestly don't need half of the stuff this thing has xd
Not sure what you're looking for, your mixer has 2 inputs
I don't know of any with 3
PC > USB > Scarlet
Switch > 3.5mm or 6.35mm > Scarlet input
you can just google what you want
plenty of 2in1out mixers
Yeah I tried this but noticed a ton of static noise. I'll try a different cable next before going with your rec.
I found a few that I was skeptical of so I thought I'd ask here. Wanted to make sure I wasn't buying eWaste
you can compare it and return if it messes up your audio
Might be a long shot but anybody know why since I got my wave xlr my audio in shadowplay clips is like double audio
like it sounds layered or something
I have a pair of usbc iems and they don't quite reach my pc, will using a usbc extension cable have any noticeable effect on audio, specifically latency?
no.
Is it going thru stream mix also?
It might be recording both stream mix and desktop audio
Yay
i just bought some zigaat arete x Freshreviews IEM's, and every reviewer said to buy a apple 3.5 to usb C DAC to get the most out of these IEM's.
When i plug mine in my apple DAC i get very bad audio quality and static background noise,
if i plug the IEM's straight in my gaming PC's Motherboard i get clean sound but now i dont know if im missing out on better quality if i would buy a differrent DAC?
curse of tidal only half working for me but downdetector has yet to give me any definite answers so i have to suffer on soundcloud instead(its not that bad but meh i prefer 2 working platforms)
oh also i recently got some mdr7506s and b&w P5s
the 7506s are really good at bringing fowards detail and theyre pretty well made but id still take my mezes over them overall but they definitely have some really nice qualities
the p5s cost someone like 300 new and they have a dead driver and have a "leather and aluminum" construction but the main filler material is still plastic
ill get the driver fixed at some point though but the working one does sound good but i wouldnt buy them for 300 but some people are insane i guess
guys Linus & DMS have B&K 4128/5128 HATS measuring rig and FR targets, they could've put the data to Squiglink.
i bet theres many gaming peripherals that LTT had measured other than IEMs and Wireless Headphones/Earphones TWS
we must end the graphing tyrrany
how exciting
audio reviewer starter pack:
bk5128
earplugs
a muzzle so you can't talk about the sound
truth enhancement serum for when you're given headphones for review
a degree in not listening to anyone
There's a lot of challenges with squiglink, particularly if you're using newer standards like 5128
Biggest being that people continue to directly compare gras and 5128 data when you can't do that
But also the hptf variation measurements shown by the headphone show, DMS and LTT are much more beneficial than single line squigs
wait how did you learn this
I thought the last time they did a headphone video was the bengoo headset
I suppose it's cool data to have, but I'm not sure how practical it is. don't people wear headphones almost the same way each time? and why would it have any implications when the response( when turned into an eq) already contains serious errors with what parts are smooth or have sharp peaks/dips
"almost the same way," does NOT mean the same transfer function.
here's an example of a transfer function you might see,
where:
x is seal quality,
y is pad compression symmetry,
z is pinna interaction,
and S is jaw position.
if any one of those values change slightly, the transfer function is different.
but what you are saying is a small change could cause a very large difference in the output?
disproportionately
I have never tested a headphone where a small change in a plausible wearing position creates a huge change in frequency response outside of breaking the seal
if such a headphone existed, why would we think a 5128 measurement would find this deviation for someone else's ear at the same frequency
or the same amount
a fixed amount of movement of the earcup corresponds to different % changes in distance of a given point on the headphone baffle to nearby parts of the pinna, depending on which point on the baffle is used
or what pinna is used
so it stands to reason that a large seating change causing a 1db change at 8k for one person's ears, would take a smaller seating change to recreate that boost with a different pinna
the earcups are moving across something
It's not about the positioning. Headphone measurements on a GRAS or 5128 or any other rig already should be an average of many seatings if the person knows what they're doing.
It's about showing how the headphone behaves on different listeners even after compensating their HRTFs away as many headphones behave quite differently. You can design a headphone specifically for one particular rig and it won't have the same behaviour on another rig or listener.
Data from gras cannot be directly compared to 5128, which is one of the big issues with squiglink
Squiglink also just shows raw measurements, not compensated, which also is a step back
Squiglink is handy for collating stuff together but for more proper evaluation there's a reason why headphones.com and LTT are both using a more modern approach
oh I thought those bounds were related to positioning
what's the modern approach ltt and headphones.com are using
so it's the lower/upper bounds from several different rigs, all df compensated or similar?
Love the smell of new electronics.
this depends on the specific transfer function.
consider a system with high sensitivity.
suppose the change in the transfer function with respect to x is quite large.
would this make sense?
would the transfer function change relatively greatly if the headphone's seal ranges from perfect to poor?
suppose that the value of x ranges between 0 and 1, where 0 is a poor seal and 1 a near-perfect seal.
but now lets suppose we're testing.
well, i'll let that sit
if someone wants to lean into the probability side of things I can do that.
why are you acting like this is math
this formula(tm) is as mathematical as f***around = find out
just say what you mean
what i was attempting to do
i don't really get what all of that is supposed to be for when it basically boils down to "maybe, maybe not"
that's what i was trying to get at
none of us have any idea what H(f) actually contain inside
you can only guess about it's qualities
it seemed like what you were trying to imply was that the transfer function might be very sensitive
--
I don't know, I am so confused because I reread that tirade like 5 times and it doesn't seem like it says anything more than me trying to get the essay from 300 words to 500
What if the creator (i think Super* Review did) added that kind of function to squig link
a button "HpTF Variance" and woosh. you can see the "variance"
we may need a open source B&K Measurement website
because for me its sometimes annoying to find 5128 measurements because so many people do different measurements on their own website. then you have to go there, trace and so on...
there's no official DF curves for probably most of the rigs used
alright, i do not plan to compare data from one rig to another
but to find information on the peaks or dips, since it's an objective measurement with the highest accuracy as of now with the 5128 . so I'd know where i may find the sound to be sibilant or dull.
iirc you can compare IEC 711 and Gras if the same Pinna was used
but im not sure
keyword may
your own ears are going to create plenty of peaks and dips that aren't on any measurement
the rig with the highest accuracy is probably the one that's closest to your earshape
of course the canal simulator quality/tech matters too, but wow is earshape a game changer
if its an average of many seating position its a decent representation.
im not interested in comparison of different rigs, i know each rig have its own representation of measurement.
ft1 vs jackson ear
wont find that on a rig
but you may find things that resemble certain parts
like the small 800hz notch
we eq to our preference but nothing wrong in seeing its frequency response on a HATS
I didn't say anything about preference
i do say it
what I mean is I didn't eq this to preference, I eq'd it to hrtf/df tilt
cool.
š¤
what I'm getting at is that yes, you can see the frequency response of a headphone on a HATS, but it's literally only for the hats
it's not for you or me
it may be close enough that you can base some things off of it, but it's far from a reliable metric of how something will sound to you tonally
well its fine if the peaks or dips aren't there on our own head, theres no problem, so whats wrong with seeing the measurements on a HATS
i know that it deviates from our head, but its not bad to base it somewhere
the power you give it in your purchasing decisions
it must be understood and regulated yes
i give it a demo -> see measurement -> purchase or not came after.
the actual and more off-topic answer I'd give you though is that delivered frequency response has almost no bearing on how a headphone ultimately sounds
well, some objectively good on a measurement can sound bad to me and vice versa.
with +7db bass?
it already HAS a ton of bass (with proper seal)
Bass and mids are oftenly very VERY close
the most varaince starts at ~3kHz
true
inverted eq
ah ok inverted. makes more sense now 
even tho i don't understand the EQ anyways
i just Trace it an substract that with the FR
yes
OR
measure speakers, make that "neutral"
and try to match the tuning of the callibrated speaker to your headphone
at the end. add preference (mostly easy ones like bass, warmth, air bla bla)
well thats assuming i have decent room acoustics
i dont
i just measured my shitty placed speakers in my room
EQd it to ~ -0.5dB Tilt
EQd my HD800s. Done
sounds fcking awesome
your speaker calibrated out of the box which speaker is that
HS7
yamaha?
ye
I mostly listen to the mc450 without eq these days
my brain is accepting it's massive tilt
Blue: Target 5128 DF Tilt
Red: FT1 (measured on 5128)
Yellow: Jackson EQ

RAW
after EQ
(bass looks different now cuz i changes some stuff but didnt measured again)
i dont have reliable mic for measuring all i have is my phone mic.
yeah that aint gonna work
fr. get UMIK-1
its easy plug and play
you can download the CAL file from their website (for your specific unit ofc)
and in REW it just works easly
thanks for the help
ye
UMIK-1 is not the best
but for little stuff like home setups it should be more than enough
and for me it made a HUGE upgrade
i dont have a monitoring speaker yet, but probably be useful for EQ ing my car speaker

I think you still have it upside down
on your EQ + = - right?
my eq is basically a df-esque compensated frequency response measurement
in terms of how it's shown
seems confusing enough.
bcos i dont understand it
an eq is the opposite of a frequency response measurement
Blue: 5128 DF -1dB Tilt/oct
Red: FiiO FT1 measured on 5128
Orange: "positive" Jackson EQ
Green "negative" Jackson EQ
rephrase please
bcs idk what u mean by that
you perceive a frequency response
you add eq to negate the frequency response so things sound normal
this is subtraction
and the result is 0
sometimes you need to add stuff
For that you would need to know exactly what your HRTF is and how the FT1 measures on your head
I basically know this with a small margin of error
looking at your EQ. i doubt that heavily
to put it simply what do you achieve with the EQ on FT1? do you add bass shelf or reduce the bass?
I don't know what you did to get the orange line, but it's now leaning in the right direction
I get that that's the theory, but it doesn't work in practice
I get a lot of nulls in the 70hz area on a lot of headphones
seal issue?
not all headphones, not always 70hz, not always the same amount either
nah seal's good, I've put rubber bands on before and used more sealed earpads
also that'd kill the bass extension with closed backs
things can also happen near 1khz
I've confirmed weird things near 400hz on the mc450
the 3khz variation thing just doesn't apply to actual human beings
fr.
i just don't trust in what you hear
the measurements i showed where human beings
also i doubt that your Anatomy is that different that it's really like that
well iguess thats is his own HRTF
then there are things they can hear that aren't being shown by the measurement process here
i doubt it personally that it's THAT different
i think he mixes up the sound he is used to with what he hears as neutral
that wouldn't make sense
ofc it does
my brain can't consciously correct for that 70hz dip for example in real world listening
I had to encounter it with frequency sweep testing and many reference sources
you'll also notice I have lots of eq's where I have minimal or no bump in that area, and it aligns more with measurements
the same process is used
still i doubt it in general
unless there is actually something different about you
maybe really weird Pinna. but from your vids it looks "average"
bcs no matter what i do with my EQs.
The headphones sound all really close to each other in tuning
always have to lower 3kHz region cuz. i dont like it
and some annoying ass peaks in treble
like here. my HD800s EQ
and this sounds REALLY close to my Speakers
(in terms of tuning)
added a bit preference (the 3dB Bass shelf) and done
on other headphones it doesn't look that good sadly š
funny thing about that EQ is
well i guess many products rarely sound the way we want it to be, such as for me in IEMs i find those "meta" tuned to have too much sub bass or some other they add too much treble energy between 4-10k (which is very perceivable to me)
i have Ear canal diff
I'm a little surprised you'd think a bass notch not shown on rigs is that weird. or that current research suggests it's impossible for that dip to naturally exist under a good seal
if research is that limited though, it will reveal itself once people stop being so certain in theories laid out by the existing research
but as you know I'm not fond of frequency response shenanigans in general
like the one with the DT 770?
I don't think the world will gain much of anything if we conquer frequency response
i guess its just that quite many iems add lower treble energy based on seeing 5128 graph.
maybe someone can save their k92
Frequency Response is just awesome
very cool so see sound i agree
we need a way to measure a individual one
in ear mics
has flaws
just tipping the electrical signal from the ear into an amp and ADC
but its a way to compensate for calibration person to person, like how CIEMs works
time will tell more truth š
when it comes to our brain and how it works, humans are

in a way it's come full circle
imagine your brain knows where all your organs and muscles are.
But you may had sore muscles of some you never knew there was onexD
we wish to solve the brain or maybe ears, but unfortunately our brains are in the way of progressing research on this
elon musk brain chip plays music directly to our brain ( with unskipabble ads )
my question is
If you play a insanely loud signal directly in your brain
you should be able to hear it as loud
but will it damage you?
bcs. earts where not used to process that signal
Finally i can listen to 200dB music without getting hearing damage 
drugs or something
also. are ours ears really lossless?
in our ears there are moving parts.
But when something moves it creates also energy as heat
probably need that bass vest
can't drugs stop neurons from talking to eachother
so the signal we perceived is not the actual original one. its different
im just messing around 
people go on rave and be
its easier to get the singer to sing privately, than to understand audiophile science
but its not lossless
bcs the signal loses stuff of its original amount of energy 
its lossless you shrink the singer to enter your ear canal
i dont want a chipmunk voice in my ears
and it also would tickle me
its okay, you can always+ 50db of bass shelf and slowed down using tiktok edit so it wouldn't sound like chipmunk
as soon as data has so much as the thought of being represented by a single electron, it is lossy
dont give audio company idea. they'll made another $1m DAC with no electrons lost.
neutron bombardment audio codec
audio cables aint pure enough unless we made audio cable from fiber optic with lossless light transmission
but even the rate of time is slightly different in every position in the universe
there will always be rate distortion among other things
no not rate distortion in a typical sense
we just build a whole Audio Chain with that idea
but like vinyl
wait guys
waves stretch over a distance
thats why light gets red
so we already listen to "slowed" music
from serious audio talk to audio tomfoolery
thats why my music sounds 0.00000000001 semitones off
but you cannot confirm the existence or non-existence of this magic dac unless the box is opened
and I won't let you open this
schrodinger's dac
Schrƶdingers DAC?
this is the secret to lossless technology
its existence cannot be disproven if not observed
that's encompassed in all the various human related transfer functions (if that were a term)
STRF
psychological and physiological transfer function?
Yo brothers, I recently bought Sennheiser HDB630 and was trying out abx testing, however i noticed when i change sampling rate to 96k in audio MIDI macos i belive (not sure) is upsampling/resampling it. Is there a way it can autosample as per the source. IPhone handles it perfectly
they way i confirm sampling rate is through the sennheiser app. ( tried both USB and analog connections)
macos definitely changes output sampling rate automatically in my experience
realistically, it doesn't matter as long as the output sampling rate is greater or equal to the source
hmm, strange
I have spent years in DSP and signal processing.
it is just math.
I understand the frustration, but you can't ignore the way we describe systems, which is mathematically.
you even considering looking at a frequency response measurement is mathematics.
your measurement rig likely has DSP.
this is even before considering the headphones themselves, where the driver acts as a mass-spring-damper system on top of its electrical characteristics.
on top of all that you have the probability models you use to approximate many characteristics, which is where measurements are derived.
A measurement is rarely ever a direct reading of a physical thing. It's a statistical estimator. On any measurement rig, you are merely estimating the headphone's frequency response with some statistical certainty.
this isn't math
it's a list of things that affect hptf and it's written like math by someone who wants to feel smart
you can't plug in numbers for the variables here. you can't even get numbers for many, and there's no standardized ways to know what formulas are to be plugged into this one in order for this final formula to work
just quit while you're ahead. we get it
ok buddy.
guess my 10 years of describing any signal processing technique with math is irrelevant.
Not sure what that coupling/test rig is doing without DSP but jackson knows all.
nor what this is about: https://www.mathworks.com/help/audio/ref/interpolatehrtf.html
hey, don't forget it's not just htrf, seal issues, head size, etc but also how your brain processes sound
in practise
brain is important
we need to use it when listening, as well as when quantifying how headphones may work
avoiding overextending the scope of research is as important as avoiding placebo
it doesn't really seem like the formal math definitions were relevant for what the conversation was about
our brain still expects our an atomies processed unique sound
you may be able to set your audio perfectly for your physical head but then comes how your brain perceives which frequency and there we go⦠you now need to know the neural pathways and chemical activity in your brain xd
then people get earpiercings, stretchers of different materials and sizes and locations, changing hair , a clogged up nose closing of that tube (here its called eustachius) , open and close their mouth ,... earwax , so...
oh yea. forgot about the earpiercing resonance.
š used to go to parties where the volume was so loud with so much sound pressure that when walking 15meters in front of the stage , you had trouble keeping your balance and walking straight because your middle ear had no clue what was happening anymore š , it does clean them out pretty well š
convince me to start using IEMs again. I bought expensive ones like over 50 >100 sometimes and all of it breaks within a month or two, literally used it on a laptop only, didnt wear while sleeping, didnt wear while going outside as well. asked 12 people about their experience, 10/12 answered theirs broke within 2-5 months of usage same price range. Is CHU 2 even gonna last more than 2 months?
I opted to leave the loop open since I realized I was getting too into the weeds.
thanks for the reasonable reply. That's a more reasonable comment to make. If anyone's ever curious about the math, then I'm happy to talk about it and can provide MATLAB examples.
Have you tried not smashing them with a hammer?
tha math, yeey š its all fun and games untill you actually have to figure it out lol š
ow for those intrested aswell here is the whole collection : https://web.ece.ucsb.edu/Faculty/Rabiner/ece259/Reprints/ š
Someone's in DSP
more like understanding it for then doing it in c++ for VST
long term project in my spare time atm as doing it myself quickly ,... didnt work out š
so its more specificly that (specific linear phase low/hi pass filter , then it is DSP mathematics intrest in general ,... ( i have the basics principles down form Analog Electronics class in school (how they translate to a digital analog i never dug myself into, i understand the digital audio part well enough and how it should work when analog :D)
(and yet i prefer digital audio as format over analog :D)
Feel free to reach out with questions.
https://www.amazon.com/COOMAX-Earpiece-Invisible-Earphone-Covert/dp/B00XT3X0I4
isnt this dangerous
trying to wear earbuds
without anyone noticing
https://www.amazon.com/gp/product/B0FXSC665C?psc=1
everyhting thats listed as discreet is way too obvious
and these the only stuff i can find
how discrete does it have to be , because things might be more discrete as you might think , this guy only got found out because he lost signal and started tapping his ear , and only security cameras after the fact revealed it when they were watching for it ... https://www.youtube.com/watch?v=Dm--1fZbSIg
How He Cheated the Casino Out of $32 Million Using Its Own Cameras
š° How He Cheated the Casino Out of $32 Million Using Its Own Cameras
The true story of the man who outsmarted Australia's most secure casino using nothing but surveillance feeds, a hidden earpiece, and perfect timing. No masks. No guns. Just strategy, human weakness, and a sy...
this might also be a verry discrete option : https://en.wikipedia.org/wiki/Bone_conduction
Bone conduction is the conduction of sound to the inner ear primarily through the bones of the skull, allowing the hearer to perceive audio content even if the ear canal is blocked. Bone conduction transmission occurs constantly as sound waves vibrate bone, specifically the bones in the skull, although it is hard for the average individual to di...
well
discrete enoug
to where you wouldnt see if you were walking by me
the 2nd one is a good option
this is NOT reaching the united states in 24 hr š
i used tho have a good shop for these things but i cant find it anymore the sold like crazy stuff, , a electronic board the size of a flat cell battery that once turned on beeped loud but random intervals between 1minute and 4 hours , and it beeped between 0.2sec and 3 seconds or so never long enough to pinpoint and to sporadic and possibly long silent times to wait for it but to loud not to care, hide it in someones office and watch them go mad over a month or 2, start tearing everything open and appart ... š
was made so battery lasted insane amounts since int barely used any power ever š
very funny. reply again once you have a logical answer. here's some more context. I only use it on my laptop, never on my phone, never while the laptop is charged, and never while I go to sleep. just a light use on laptop. based on that context, your brain should be able to comprehend that the IEM is not smashed with a hammer.
Very hard to believe 10/12 ppl you talked to have broken iems
if someone says discrete instead of discreet one more time I'm gonna crash out
any recommendations for 2.1 speaker set up budget $150 also its for my pc
i actually have a dumb question, on your computers (wich os?) , i think most of you have the sampling rate of your dac's set to something like 48khz or 96khz , but if you play music sampled at 44100hz , do you just play it at 48khz or does your audioplayer(wich?) resample automaticly? or does your os or soundcard driver(asio4all?) resample everything to the correct outputsamplerate?
i have my sytem(linux / pipewire) set to 48khz btw at the moment, and my audioplayer foobar outputs directly to easyeffects(pipewire) so also 48khz , but i do have a resampler in my dsp chain(pphs) that resamples to 48khz,...
(why , if you ask me if i have broken headpones, then yes prolly 3 ones still laying around somewhere, same for inear plugs 20 broken wired or so, , and wireless budds also 1 set broken and two sets that miss one of the two earbuds,.... dont really have real iems but i would guess that if i had a couple prolly there would be a broken one amongst them
Ok but thats not how stats work
I thought macos, and possibly pipewire when configured correctly, switch the output to 44.1 kHz automatically. But otherwise (I think always on windows), it gets upsampled
probably the buffer is so enormous that it has almost immeasurably 0 loss
on windows i really have the feeling that that is not the case ,... playing a 44100hz track with everything set to 48khz makes the track play faster to my feel but its been a while so
that is certainly placebo, the only difference would be a slight background hiss at an extremely low volume
and even with a hube buffer , if you can buffer 2880K samples ,if you would fill it with an audio track sampled at 44100 hz of 1 minute long , the last 234K samples of your buffer will be empty so when the buffer plays at 48k your left with less than a minute of audio
why would that be ?
you don't just take the 44.1 kHz samples and send them directly to the 48 kHz output and zero the rest of the samples
You essentially reconstruct the original signal, then resample it at new positions (distance of 1/48000 of a second each), and send that to the output
and then of course there's a bit of error in that process when done in a world where you have a finite length of time and a finite precision of the samples, so you use randomness to spread out the error into some sort of noise
Also iirc the high frequency gets attenuated a bit, but not to an audible degree
no you dont zerro you have a file with 234k samples of audio , 1 minute long at 44100hz exactly,...
next if you play that file, you copy the values for each sample , to the buffer , since you only have 243samples , you fill the first 234k samples oif the buffer (theoretically if the buffer is nto emptying)
you dont make more samples unless you resample the file
your not reconstructing anything , reconstructiong the original signal is jte job of the DAC not the buffer
de buffer feeds the dac
at a certain samplerate , 48k in my example
I'm confused what you think is happening; the audio system on the OS is resampling the audio on the fly
im just saying its not resampling i think ,
maybe the driver (asio4all has an option for it yes)
I mentioned reconstructing as a way to describe how resampling works conceptually, I believe it's accurate
what would it do if not resampling? if the DAC is running at 48 kHz and expects 48,000 samples in a one second period, but you only give it 44,100 samples, what are you doing to the remaining 3,900 samples the DAC is accepting
that's why I referred to "setting to zero", i.e. if the buffer was zero initialized and you only set the first samples, leaving the rest as zero
so looked it up windows only resamples in shared mode , not in exclusive mode and asio bypasses the whole oreal and also by default does not resample, and i think most audio people would be using asio (whats the point of a dac if not so)
meaning it's just reconfiguring the DAC to use 44.1khz when you play music directly out at 44.1 khz
na you give it audio, at 48000 samples a second, if the audio file is recorded at only 44100 samples a second , doesnt matter you feed the samples at 48000 a second so the audio file wil reache its end faster than expected
It'd be pretty obvious if, due to some bug or something, it wasn't resampling since you'd hear a pop or something every (48000/buffer size) seconds
The music player is gonna be clocked against an absolute clock (real time), not somehow clocking against the sample rate of the output dac
its esier to look at it like a bucket , of audio , if you fill it with water , fi you ar draining the bucket at 1cl a second and you poor a bottle with 100samples in you have water for 100seconds , h9ow fsast the bottlel with teh 100 samples was filled doesnt matter, only if its different form the bucket it will eitehr sound faster or slower
From another perspective , look at it this way:
48000/44100 ~= 1.088
It would be extremely obvious something is wrong, i.e. try speeding up a video by 9% and listen
if you switch asio from 44100hz to 192ljz while audio is playing you can actually hear it š the buffer suddely gets drained faster and audo is faster for a moment (untill the buffer empties ofc,(asio also switches the host so depending on that it depends after that)]
Oh are you talking about it just sounding weird for a split second after changing settings? I thought you meant it perpetually sounds different
no i mean you can hear the screech when you do that k after that since the hoist also siwtches and whatever host you use might just either stop playback or it migh do resampling itself (or with a DAW , the instuments just startmaking more samples ...
It's not unusual to import an audio file into Cubase 14 and discover that it's the wrong speed. 99% of the time it's a Sample Rate issue, which is easy to fix - just watch this video to find out how!
This video gives an example of how you might encounter this issue, explains what Sample Rates are and shows you how to fix the problem with Cubase ...
It's hard to read that message; but if you're not resampling, the source and output rates differ, and the buffer is a finite size, then something has to break down. Yeah actually you would hear it speed up, but you'd also hear pops as you hit regions where there were no samples remaining
Remember that you're breaking up the audio into finite length chunks, you're not taking the entire audio file as a whole and sending it to the DAC
There's some point the difference has to be reconciled, that time should probably be at the end of every buffer
This part responds to that video link, btw
regionss where no samples are remaining do not exist unless your cpu cannot copy samples fast enough
a track is just a series of samples , the metadata has the samplerate but your cpu just copies samples at the rate it can untill the buffer is filled or the source ran out of samples (in case of live mic orso)
its why rt kernels are nice for djing as it guarantees that samples will be copied at a given interval, hence if the system can handle it now it wont ever underrun no matter the load on the system
Imagine you roll a wheel with a constant circumference and rotational speed, and the linear distance you're actually traveling is larger: something has to make up for that difference (unless you like compress space and time using gravity or something); the wheel has to slip at some point
I am almost certain this isn't how audio players work
I will take a look at one
im kind of verry certain , your not on a wheel but on a conveyerbelt that moves at a speed , if you empty a box on it how fast the box(the track) recorded doesnt really matter ,...
Idk how to explain it but it should be obvious from this file in cmus https://github.com/cmus/cmus/blob/56446f70e86445caa96f6f4f0188f175fc1fd9a0/op/pulse.c that the clock determining the rate at which audio is played is happening elsewhere (from a system time clock), not the sample rate of the output device
a track 44100hz of one minute is just a blob of 2646000 samples ,
there is no seconds diveder in it or whatever ,
But no DAC takes a 1 minute 2 MB buffer for your audio and holds it to play back
the samples have no lenght only a value of volume
it is being divided into small buffer sizes; you can see your buffer size in pipewire I believe defaults to like 1/4096 of a second or something?
whats a good headphone amp for the dt 770 pro
It might be in terms of samples
But you're chopping up the audio file into small chunks and playing those for a certain amount of time
my dac is set to 32 samples of buffer , so my cpu cant copy much at a time and has to switch to that task allot to keep the buffer filledwhen blaying, but it doesnt care if the track waa 441000 hz it just copies the samplesand they dont have lengt , the dac just takes at a given rate
The audio player takes a group of 4410 samples out of the audio file, and knows that's 10 ms so sends it to the (hypothetical) device that's expecting a 4410 sample buffer that it will play for 10 ms
It does though, there's a sample rate the DAC is set to. Otherwise how would it know how much delay to put between each sample from its buffer during reconstruction into your ears?
The buffer itself doesn't have any temporal meaning so that has to be metadata on the side
In the form of a sample rate on the DAC
The sample rate on the DAC is what's creating that relationship between buffer length and the period of time it's presented for
If the DAC is changed to 48 kHz, receiving a 4410 sample buffer has a different meaning in time
That statement is so wrong.
I buy all my music on Compact Disc now.
Actually cheaper than buying lossless a lot of the time
Also, that's irrelevant.
guys i have like a very bad earbuds
is it possible to change its CODEC somehow
like make it a better codec
or is it hardwired to the earbuds, like is it fixes
fixed*
what earbuds are you talking about specifically
Umm
Like its kinda a local brand, popular in india but not anywhere else
Boat 701 prime anc
@thin void a while ago I remember you did a video of some japanese piezoelectric headphones.
You mentioned that they "sounded like hearing damage."
Do you remember what headphones they were and if the video is still up? I can't for the life of me find it or the headphones.
after digging in to it , (havent fully reached the bottom of it) it seems we were both wrong to a certain degree, for direct channels (like asio, or exclusive mode on windows) no resampling is happening on the audio channel side of things , and if that was all what is to it i would have been the closest to the real workd , but any media application that allows for eq ing or any other audio effect , will do a thing internalli called suppersampling do do the audioprocessing with... so at this point the original samplerate of the audio has been lost totally. and what happens next isthat whatever asio is set to as samplerate, can be output arbitrarely ,... this only works mostly fawless if their is no extreme difference like trying to have a media player play gsm audio (8k/s samplerate) to play at 192k/s . thats the player side , so the cpu can now copy a stream of samples to the soundcard, when on wasapi and not exclusive mode , windows uses a 512samples large buffer in memory for every audio stream, before they get mixed together to another 512k buffer , that then gets copied to the adress space of the soundcard. if the driver of the soundcard hasnent prepared an adress space for the buffer on the card , hardware "mode" the card unless designed to do this wont even use its buffer internally(think AC97 or onboard sound eg, that do not really have a hardwarebuffer) , when set to hardware buffer and something else doing the mixing (reason why you can only assign one stream per soundcard using asio in an asio driver) most DAC's completely ignore whatever an audio sample is supposed to be anyway(has been lost anyway) the only thing they are aware of is how fast the whole of the buffer should be shifted out ,... at wich point the dac suppersamples it again before presenting it to the actual DAC in the dac , wich is usually set to a verry specific samplerate that cannot change, since suppersampling is a lossy opperation depending on the alorithm used ,...
it seems this wast always the case btw,... (duh) , and back when 44100khz (wich was picked for a verry specific reason, only way to be consistent on both ntsc and pal that is also above the nyquist frequency of the human ear for most people) it mostly would have worked more or less the way i tought it worked.
kind of sad for all the audio files , that basicly all their pure audio still gets supersampled and introducing artifacting to then being downsampled again introducing more artifacting and filtered to remove most of it,... making what is going to the speakers about as impure as , playing vinyl on a planet with gravity like earth lol
also ignoring what i said above for a second and assuming we are still when things were simpler ...
- Why would the audio player take a group of 4410 (for example ) ? think about it for a second , as there would could be a good answer for it
... if you dont think about it form the audio palyer does a and or b , but rather as from the standpoint of the cpu, (on windows when set to optimize for background tasks , wich is as close as you can get to RT(fully preemted kernel sheduler aka realtime kernel) behavior (on linux)with windows , the kernel makes sure that the asio program will get serviced on a verry regular interval , the asio program then has the kernel copy data from the buffers in memoy in a FIFO kind of way to the adress space of the soundcard , as much as required to fill up the specified buffer again ... all this time the cpu has no clue about the time since its did not need it , it just copies data as always , the kernel swithces tasks again , one of wich it switches to may be the mediaplayer, if so it wil copy more samples from the audio file located itn the media players memory space , to fill the buffer of the asio driver in memory again, once that is done the cpu checks how many smamples it copied (kind of as its prolly just increasing a samplecounter in memory for ech copy) . another routine in the media player might be to check the number of samples copied since playback to the statingpoint of playback and adust the timer to reflect the change in ms , thats the only point where something that can be considered "time" enters the frame , since time is a clusterfuck in programming anything you do can do without time you do without,.. the DAC has its internal clock generators and depending on how acurate they are it reads data from its buffes to its dac at that pace... if the kernel is not an RT kernel the next time the asio buffergets read is not set and dependant on the systemload , risking underruns , but also makes the number
samples that have to be added to the buffer variable , and becaue of this variable timing you defenetly want to make sure to fill up the buffer completely , and even for an rt kernel youd want to fill up the buffer to as much as you are able to, think that i have my buffer sizes set to : 32 samples at 44800hz playback thats not allot of time or audio thats in the buffer so my cpu has to have refilled it atleast every 0.0006s or i get buffer under runs , unfortunatly the monitoring software i use does not record how long it has been acitve for, .. however it does keep track of the number of underruns that have occured :
the TakeT H2+. I could never forget something that bad lol
š«” thank you kindly
(if you want a funny little experiment)
write some c-code that does one thing and one thing only , check systemtime as acuratly as provided , compare it tho the previously read value and keep the min and max of the comparissons and the average , printing them to stdout ,.. thats only the diffence in reading from it , now think how with that information you can get to writing software that acurately can mark of 10ms intervals without blowing out your cpu to 100% load just for the time checking , if you know the audio is 10ms thats one but that also means that exactly 10ms later you have to make sure to copy the next 10ms be to early and you have to wait(not happening) until there is place for the next 10ms in the buffer , if you're to late you get an under run ,
ah but what if the buffer is 20ms and you copy 10ms each time? since you have no idea how much time was elapsed since the last time , you still have to tink of a way to exactly find out , because if you copy 10ms to the buffer , if you were late a ms each time so far after 10ms the buffer will be empty , if your eary by a ms each time the buffer will overrun so you or you have to go check system time and be verry acurate with it just in order to know if you have to aske the audio program to give another 10ms or not , lastly some code i wrote for a vst that instantly reduces gain in relation to a sidechanel input (like a ~0ms kind of compressor or so) note how at no point in the code any notion of time is made or taken into account š
ow and forgive the nextline '{' in that code i dont normally do that , but its how the VST3 sdk from steinberg has setup the code so ... i went with it
Why do iPhones do this? The lowest volume is super loud on every non Apple headphones. I would think something is broken if I didnāt have apple earbuds to compare to.
because , apple is against apple blasphemy , you buy apple you buy into the apple ecosystem and thats mandatory , the only reason why you can plug in 3rd party devices is because of lawsuits that they can no longer use their proprietary first party only conectors, but for all intends an purposes ist only supposed to fit , not to work properly , so as long as it does something in the regard of what its supposed to , it will do it in the most onnoxiously unusable way, to the amount the cannot be sued for it but also so nobody makes use of ti
i just got a pair of hifiman edition xs, i was wondering if there's something i should so i hear good imaging, because rn listening to music doesn't sound much different from my airpods pro 2. could it just be that my motherboard dac or codec (alc4080) isn't good enough and a dedicated amp/dac would improve it? they're also quite quiet on max volume
A dedicated amplifier will definitely get you to louder levels.
A DAC might improve some of the things you're having issues with, but it's hard to know if what you're hearing is the headphones or the source.
The airpod pro 2's are pretty good too.
i'm listening to a 100mb flac song file so i don't think it could be the source
yeah but these are over ear so i'm guessing i'm supposed to hear music around me kind of?
which i don't think i am currently
The source is your motherboard audio. Lossless media played through shit can still be shut.
The soundstage you'd expect from nice headphones is greatly exaggerated and is usually relative to other headphones.
You will only get actually good soundstage from speakers, or from binaural recordings meant specifically for headphones.
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I've had a pair of ATH-M50x's for a few years, and the hinge on one side is breaking. My last pair went the same way after a few years of use. I'm looking for my next pair of headphones, any suggestions on something with good fidelity and durability? I run through a Scarlett 2i2 as a DAC. All suggestions appreciated!
I do not want ANC in whatever pair of headphones I get
How much money do you want to spend and are you okay with open backs?
I've never owned open back, but I was looking at HD 600's
I think budget wise I'd like to stay in the $500 or less bracket
don't need to spend $500 on them but I think that'd be my upper limit
They're a pretty good option. The 6XX's are a cheaper alternative and are one of my favorite headphones. Very similar to the hd600, just a touch warmer.
Yeah the hd600's are like $250 on amazon right now
Not a bad price.
6xx should be around $180-$200 right now
how do open back compare? I've not used them extensively
Open backs are a much more natural presentation and are easier to tune.
Minor downsides of letting outside noise in and inside noise out. But as long as you're in your own room it's a non-issue.
yeah this would be my desk pair 100%
I say send it
Slightly different sound signature.
560s is gonna have a bit better bass extension and a bit more V shaped.
sent it, thanks for the advice!
That sounds a lot like what I was trying to explain but can you send links to what you were reading?
I was using 44100 as an example buffer size, in reality it'd be a small size like 512 samples for example; the point being it's much smaller than the entire song
-> i use 32 samples (see screenshot) i started with this : https://dspace.tul.cz/server/api/core/bitstreams/39cee442-acf3-4ba4-8a9f-7b9988a8abe4/content then fell down the rabithole
Tell us what you think when you get it
appart from the whole thing being somehow timeaware, , due to the software suppersampling , its always filling its own buffer with the sound sampled at the samplerate set by asio, it still just gets copied untill the buffer is filled , as time unaware as it can be, only the otherend is also set to be that samplerate, the fifo, gets cleared at that speed on the DAC then it gets suppersampled again for the actual DAC wich expect and arbitrary but fixed samplerate you cant change so š
Yes
The sample rate is the time awareness
ah you missed my intend behind that question, it can be any size ofc , was more to think about why not one more or one less in general since it can take any amount , if it takes a certain amount because its some time interval your stuck figuring out time on a pc , wich is a pain , because you have to poll something that keeps track of time,and due to the sheduler your never sure when your get your next poll again so your time might be over shot , or your checking so often your hogging the cpu just to check time so you would be trying to check as little as possible but still be acurate? , so its avoided to work without depending on time intervals especially when you can do without,
by that i mean your cpu or your program has no ffing idea how much time is elapsed at any point ,.. and your cpu is just copieng samples to fill the buffer untill full no mater the buffer size or the samplerate,... if its its turn to do that, it reads samples at location a , writes them to b untill b is full
the audio player only gets updated and calculates how much time is elapsed after the fact , and it gets calculated from the number of samples that got cleared (how much could be copied to the fixed size buffer) tells you how far the dac was at clearing and thus playing samples ...
when you say supersampling, do you mean oversampling?
pick any language you want and pick a time interval you choose , now wite some code that prints out the unixtim in ns, everythime that time interval has passed , or as close to it as you can,...
I don't think the implementation details affect the core concept I was trying to say; if you want to transmit directly to device without an audio mixer then both the audio player and the device need to be on the same page and using the same sample rate
However, if you have an audio mixer then it doesn't matter as long as the output is >= the source sample rate; oversampling is lossless with the exception of quantization noise
i mean something like this https://www.analog.com/media/en/training-seminars/tutorials/MT-085.pdf
And the other idea I was talking about is you can't sustainably accidentally speed up the audio only due to a disagreement in sample rates, without having gaps in the buffers, since the audio player is clocked against its own, wall time, not the buffer's emptying
This would never happen and if it did, would be extremely obvious
It's not an audio related statement at all, really, I am going at it from the sort of systems engineering perspective (I believe it would be called)
the way i understood it is that suppersamling is oversampling by allot
so thats actually not true (the gaps thing) they would never occur however your right that you cant have a dissagreement between what the player outputs and what the system presents to the buffer of the audio card, but thats more due to ASIO enforcing it and the player havving a suppersampled signal ready to be downampled to whatever asio demands ,... then that ithas any relevance to the track being played and what its samplerate is ,...
damn safe coding and hardware acces limitations , wast trying to demostrate it with some code but it seem you dont get access to your soundcard directly anymore without going trough the apropriate kernel driver, (being alsa most likely) wich then doenst allow you to mess about with writing incorrect stuff to the memory , since its the driver and is supposed to do thing correctly,... š i was more used to if i want a bit at a certain place in the adress space to be one , i shoud be able to do that š its my pc afterall
rip /dev/dsp
time to write your own kernel module
well , its been on my list but havent gotten around to that yet ... need another paralell life really
have had that one bookmarked ever since i saw this youtube clip https://www.youtube.com/watch?v=juGNPLdjLH4
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ive written a EFI driver before , but thats easier less stuff in your way to wrestle with
not even libc to stand in your way :p
hehe i started doing pc audio like this (but in QBASIC on dos not in C ) using your PC-Speaker (the thing that beeps when you booted old pc's :p)
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Question, does DAC quality matter, or is the amp what carries? Still trying to get my Switch 2 audio to play through my PC without buzzing noises.
Currently deciding between a 2nd DAC to plug into my switch and an HDMI audio extractor
Its both. Kinda like asking if its the CPU that matters or the GPU. and in your specific case, your computer's audio input is also factoring into the equation
so what you need is a way to get high quality audio out of your switch, which is easy. as long as it supports standard USB audio, you can get a decent dongle for ~$20 that will have good clean audio
My input interface is a Scarlet Solo so kinda doubt it's that
eh. you'd be surprised. sometimes you can have weird stuff like ground loop noise
but yeah if you have something serious probably not the cause
Are those standard or just defects
Like is that something easily testable?
It's a buzz where the pitch and intensity changes depending on what I'm doing on my Switch 2
oh well if its dependent on the switch
Can't rn, I'll send a clip tomorrow tho
then its probably electrical noise from the switch
That would make sense but it doesn't have a buzz when undocked, only docked
well that's because the power source is changing from a very clean DC battery to a very noisy power supply plugged into the wall
this is making more and more sense
Dang didn't know DC could be noisy
yeah. batteries are really clean for audio. almost no noise.

