No just joking, YouTube music mostly. It's convenient, available everywhere, has a large catalogue, and good enough quality for me.
With all respect you’re not the definition of an audiophile at all. If anything you’re kind of the opposite
Not everyone can discern the difference between a 96KHz FLAC and 256kbps AAC.
I can't. But I still can (barely) tell the difference between 256kbps AAC, and 96kbps AAC.
But I can tell if a song was well-engineered or a mess.
I believe those who can't discern the difference between bitrates (especially on high bitrates), but have the appreciation for good music, good mixing, and good mastering, can still be considered audiophile.
That's not the comparison at hand, we're talking YouTube audio compression vs any actual music track.
Especially when your browser or application requests a high quality bitrate, youtube compression is opus 128.
A person could make the argument that it’s not lossless so it’s not worth listening to, but opus is extremely high quality especially at that bitrate.
If you wanna try it for yourself, take a flac or whatever, upload it to yt, then use something like yt-dlp -x that defaults to the highest quality to redownload just the audio stream.
YouTube Music Premium offers AAC 256kbps as the highest quality.
Opus 128 is only for the audio of YouTube videos. Not YouTube Music.
and according to that same link it's 160, not 128 (format id 251!). someone else pointed that out itt.
one of my downloads had an average bitrate of ~140 when queried with mediainfo, so i believe em.
I don't have the premium account, what's aac256 comparable to?
AAC 256 should be at least on par with MP3 320 CBR, might also be on par with ogg vorbis at the same bitrate
As I get older and the abuse I put my ears through starts showing up, I completely agree. After upgrading my music library to FLAC from VBR mp3s, I stopped having the, "Oh! There's a subtle instrument going on in this part of the song!" moments.
It doesn't stop me from trying to listen to the highest quality music formats that I can get my hands on, but I 100% know if I think there's a difference to my mid-40s ears, it's probably a placebo.
Yes. As a lifelong musician (live & recording), you’d think I’d be more fussy about audio quality…
But I’m just not. Just like the 4k vs 2k “debate”… It’s all about CONTENT.
Also a musician here. I cared a lot when I was younger, but I have so many other more important things to care about now. You only have so my capacity to care about stuff in your life, and the quality of my music doesn't even come close to mattering these days.
FLACs from CDs, deemix-gui, qobuz-dl, and Soulseek. 102,000 songs. Play at home with Logitech Media Server. On the road I've transcoded it all to 128kbps Opus so i can fit it on a microsd card and I play it with PowerAmp. I mostly use Blessing2 Dusk earbuds with a Shanling MW200 bluetooth neckband, but sometimes also I use Focal Clear OG open-back over-ear cans with a qdelix 5k for bluetooth.
Audiophiles don't listen to music, they listen to their headphones
„Audiophiles don't use their equipment to listen to your music. Audiophiles use your music to listen to their equipment.“
Alan Parsons
I dunno if that's actually an Alan Parsons quote but up vote for any mention of his name.
Does sound like something he'd say.
FLACs through PlexAmp, either to nice headphones ($500 range) or two channel stereo into some decent speakers with a decent subwoofer. I'd like to upgrade to "full range" speakers one day and save the subwoofer for movies.
PlexAmp does FLAC when connected to Wi-Fi but I have it set to transcode if I'm using mobile data.
At home it gets played through Chromecast Audios (R.I.P) which keeps it all digital until it hits my receiver.
Spotify -> MOTU M2 -> HiFiMan Ananda non-stealth
"High resolution" audio is completely useless for listening (16 bit 44.1 kHz is the best it gets) and there is little value in lossless encodes for listening purposes too, so I don't get the point of all those "Hifi" streaming services.
If you own lossless encodes, I guess it doesn't hurt to use them even for listening as storage is cheap these days.
Speaking of which, I'd like to switch to purchasing my music though because Spotify will certainly continue on its path towards full enshittification. I want to be in a position where I own all my favourite music before Spotify will be infected with ads on premium plans. Oh and artists are somewhat more likely to be paid a little for their work that way (I hope...)
I plan to use the free YT music for discovery at that point.
Completely full of ads already, I routinely get promoted podcasts and gig ticket and merch notifications despite them being turned off.
I started using Spotify lite on my phone. And thankfully, there's plenty of alternative clients on desktop (such as ncspot). No crap UI elements, just playlists.
Spotify through Sonos at home and work. Spotify on Google earbuds when out and about.
I used to really love music discovery on Spotify. I now find it's the same ald songs over and over. It finds what you like and reinforces that rather than gradually expand it.
I used to really love music discovery on Spotify. I now find it's the same ald songs over and over. It finds what you like and reinforces that rather than gradually expand it.
I'm in the same boat. For years now it's felt like every daily mix and discovery playlist is 10 songs I recently just listened to on repeat and then 2 songs that aren't even tangentially related and I'm left questioning why they were being shown to me.
I agree on the discovery being crap on Spotify. I started to listen to the podcast NPR new music Fridays, and get my discovery that way nowadays.
I have converted all my CDs to FLAC and I mostly listen to my music collection in stereo speakers instead of headphones because I find the sound more natural. I have built my sound system around the moOde audio software.
Music collection as flac, navidrome as streaming server, symfonium as android app and B&W P5 or B&W Pi7 S2 for headphones.
I really wanted to like symfonium (even tho its not open source), bc it is a beautiful client, but it is a battery hog. I had to go back to ultrasonic.
I actually found all the subsonic clients to be quite heavy on my battery, so I just stuck with the one I liked the best.
FLAC's on NAS. Bluesound Node to stereo system, controlled with Roon. PlexAmp when remote.
Tidal is actually giving their lossless plan to their lower tier subscription, just got an email about it. Pretty nice.
At home: Spotify through Amazon Fire TV through Klipsch The Fives.
On the move: Spotify through Jabra Elite 4 Active.
In the bathroom: Spotify through UE Boom.
I really want to ditch Spotify, but in the meantime...
Well, TIDAL just got some price cuts, and their library is pretty comparable. Just in case you didn't know.
Just read that today! Thank you.
Same, but I want to export my playlists and liked songs from Spotify. Going through that manually atm seems like too much of a hassle.
If you plan to move to another service, there exists a number of tools to aid in moving playlists between streamers. It is really easy, once you find a good one.
Helped me break the feeling of being locked in due to have 100s of playlists.
Tried one service but didn't work with some Spotify lists, like the yearly ones. Any good recommendation that might include these as well?
Qobuz for me.
Best streaming sound available but I had some skipping issues even on very good connections and options for auto Playlist generation and new music discovery was way behind other services. Great if you always knew exaewhat you wanted to hear, but I went to Tidal and their focus on quality is better than most other services but the music discovery algorithms really are quite good, I find myself more eager than ever to tune in to a streaming service.
Budget audiophile here: I wear Superlux HD681 semi-open back cans paired with a Creative G6 DAC/amp.
The headphones are $25 but have the the most realistic soundstage I've ever heard in a pair of cans, even better than $500+ ones. Pinna activation is almost perfect; feels more like being surrounded by speakers than wearing headphones. Makes them amazing for gaming and movies, but not the best for music due to harsh siblants in the 12kHz range, which I've managed to EQ out a bit using Equalizer APO. Nice neutral sound otherwise, mids are almost perfectly flat and bass is tight—yet full—extending well below 20hz. Honestly you can't do better without spending half a grand or more.
Ehhh, I'm ballin on a budget, so take that into account.
Generally, if I really want to sink into the music, I'm going with either my lgg7 and my beyerdynamic 770 80 ohm; or whatever device can connect with my usb DAC, a fiio q3.
I do have other options, but that's my main listening because I simply don't have the budget to do a proper system with how little I get a chance to listen to music away from headphones. My computer has a decent sound card, and some klipsch speakers that aren't bad. There's a home theater unit with cd/bluray hooked up, as well as the shieldTV, and the ability to connect via Bluetooth or cable to whatever device I prefer.
My car is decent, but not audiophile level. More bass focused than anything else.
I do have other headphones. Some tin t2s, some sonys, an old set of koss, that kind of thing.
File wise, its flac and opus.
I use poweramp and/or usb audio player pro. I prefer poweramp, but the other does bit perfect, which I do like on occasion, and it's more DAC friendly.
I'm happy with the options I have, all considered. Most of it was picked up either on sale or used. I would save up while shopping, then get the best I could get when I was ready. But the key to me is that when I want to, I can listen to anything I have and hear the nuance of it. The sound is as clean as I can get it on my budget, and in all reality, my old ears can't make use of anything fancier.
You spend almost fifty years living and listening to it loud, you aren't going to get much bang for your buck out of the really high dollar, precise gear. Hell, I can barely tell a difference between lossless files and mp3 om any given listening method. It's there, I can still hear a difference, but it's barely there for me. The better gear helps, but not enough to keep upgrading for tiny changes.
I use deemix to get songs and jellyfin/finamp to listen on my phone. I do miss the discovery of new music from things like Spotify or YouTube music. If anyone has suggestions for music discovery I'd love to hear about them.
Open the Nicotine program that connects to the Soulseek network, then chat with the heads on there. Name a few artists you like and they can hook you up.
The most knowledgeable music listeners around.
Pretty sure you can search for ppl who have files of an artist you like, and then view their entire library.
(NB. Been 10 years since I've used it, so YMMV)
Seriously though, the real answer is to resurrect whatever Audiogalaxy was doing in their recommendations-algo, shit was dope.
CDs ripped to FLAC and streamed using Emby. I also use Amazon Music. At work I have a pair of ATH-M30x headphones I really like. At home ibhave some Sennheiser HD350, which are ok, but I don't like them that much as they're not that comfy.
I prefer going through the hifi - Audiolab 6000A amp, Wharfedale Pacific Evo 40 floor standers and a Wiim mini. I also have a NAD C541i CD player. On my PC I go through a NAD C320 amp and Wharfedale Diamond 9.1 bookshelves.
I listen to music mostly on my computer and in the car. The car system is nothing special. I listen through either some ATH-M40fs cans, or Presonus Erie 3.5 monitors, which are honestly glorified bookshelf speakers, but decent for the price, IMHO. All running from my (older gen2) Focusrite 2i4 interface.
I used to listen in the train/metro/bus a lot more, but I now work remotely. That’s where I used Bluetooth stuff. No need to worry about the cable getting stiff in the cold or stuck in my winter jacket. I had a pair of Beats Studio 3 I paid less than $100 for that were pretty decent for the price I paid. The sound was as bass heavy as you’d imagine from the brand, but not terribly overpowering for casual listening, and the ANC in particular was pretty impressive. I also had some Anker wireless earbuds I got with a coupon on Drop (formerly Massdrop) that were good enough for listening to podcasts and having background music.
In terms of platforms, YouTube Music mostly, and a hand picked selection on Plex for stuff that’s not on there or that I want to have always available. The music discovery algorithms are completely useless for me though. It’s the one thing Spotify did better than YTM for me. The “My Mix” playlists and artist radios have been pushing me the same artists for months on end now. Want to know the ironic part? I discover most of my music on YouTube (not Music) nowadays…
Honestly as far as cheap small monitors go, I really don't mind the Eries. They're not perfect for sure but they give a generally balanced sound and I paired them with a nice mackie sub to get pretty decent frequency coverage. Certainly perfectly decent for producing a variety of music and generally for listening to things.
I’d put them in that gap between general purpose computer/multimedia speakers, and “proper” monitors. That product range used to be a pretty terrible place to be in, but these surprised me for sure. They’re flat-ish enough that I don’t feel like I’m shooting myself in the foot using them for light production work. The bass is indeed not quite it, but what can we really expect from drivers that size. I don’t have great experience using subs for production, but that’s probably me. They’re surprisingly good for the price point and form factor, at the very least.
Yeah I think flat enough is the right phrase. Their bass is definitely lacking but with a well configured sub (I set the crossover at about 80Hz I think) you can compensate. My only feeling about producing with a sub is unless you're in a very well acoustically treated room, it's worth checking your mix on good headphones and a few sets of speakers to make sure your interesting sub bass parts are actually coming through nicely. They are good though to really work out what's going on in the sub frequencies of your mix. Also makes it really obvious when those areas are getting muddy.
At home:
FLACs ripped from CDs (prefer to buy albums I enjoy instead of Spotifying them) -> KORG DS-DAC 100 -> TEAC AX-501 -> Elac Carina BS243.4
On the go:
The same FLACs on Pixel 6 Pro -> B&O Beoplay HX
I’ve got a special speaker assembly that I shove up my ass*. The bass response is particularly pleasing.
this isn’t true.
Tidal HiFi/medium tier ->Equalizer APO with just a tiny bit of tuning -> a xDuoo stack of USB DAC + hybrid tube amp -> Sennheiser HD560S
Definitely a little bit of overkill. But still overall fantastic budget, and do it all setup. Competitive shooters, movies, and music all sound fantastic.
My next goal is a multibit DAC + tube only amp -> something like a HD 6XX. Or maybe a good solid state -> planar magnetic headphones.
At Home:
FLACs via mpd with a topping headphone amp and Audeze LCD2C headphones
Vinyl using an Audio Technica LP120, a Denon AV receiver and cheap wharfedale bookshelf speakers and a Klipsch subwoofer. That Setup isn't really audiophile tbh, especially because the room sounds terrible.
Streaming via Qobuz on both systems
On the go:
Everything encoded as Opus 128 kbit/s to fit on my phone. Played over Lypertek Tevy true wireless IEMs. Not really audiophile but tbh when I'm not at home I care much more about convenience as long as the audio quality is good enough.
also Qobuz, but at MP3 320 quality to save bandwidth
I wrote my own scripts to tag the music and encode it to FLAC and Opus and use syncthing to copy the files to my phone. So whenever I add an album to the library it will be available every where I want in the specified format without any manual copying involved. It's a little janky but has worked surprisingly well for years.
FLACs/Qobuz via Roon. I spend the most time in my office so that’s where my favorite setup is. LS50 Metas + SVS SB-1000 Pro + Peachtree GaN stack.
I also love my HD660s with the Bottlehead Crack tube amp I built.
Pixel 8 Pro Spotify -> "TempoTec Sonata HD PRO" USB DAC -> Meze 99 Classic headphones.
Does anyone think it's worth moving to Tidal for my music?
Also, I'm running out of space on my desk. I can put the stack of Schiit on top of a speaker with minimal effects, right?
I did recently and will not be going back to Spotify. There are so many small things with Tidal - actual patch notes each update, updates which clearly address user reported concerns/issues, straightforward playlist management and queue controls, an actual shuffle that isn't some weird interaction based algorithm, and of course the quality. There's been so many times I'll be listening to a song, which I've listened to many times on Spotify, and notice something in the backing track which I wasn't aware of or some aspect of a singer's voice or instrument which really pops and adds texture. They also have great recommendations and a Daily Discovery playlist. And finally - it's just music; no scrolling through podcasts or non-music this... Just high quality, easy to manage, music.
HD 560S for the cans. For my source, I use spotify, using my local library of FLACS for the stuff I like a lot, and just normal spotifly for everything else.
For earphones I have a set of KZ ZSN Pro X IEMs for when I'm on the go, when I'm at home I have my Audio Technica ATH-M50X.
On the player side I love InnerTune as a YouTube Music Frontend, while for analog I refurbished my father's BSR turntable and Phillips amplifier, both straight from the '80s
Mostly? I have uncompressed FLAC encoded music on my Plex server, and I listen to that streaming through over ear (Bose NC-700) headphones on a computer, or on our home theater system (Monitor UK, 2 stand speakers, 2 rear wall speakers, 1 subwoofer) with an Onkyo receiver.
I also listen to Tidal hifi a bunch and electronica on youtube because some of the Boiler Room and other club mixes are pretty dope :)
A technics changer or linear tracker. I think the changer has a shure cartridge still but the linear tracker has an at. Sometimes through a pair of numark ttxs with m447s and a rane.
Flacs on a server direct streamed to my source. Jellyfin is nice.
for on the move I buy sony phones just cause they still have a headphone jack. I prefer to download what i want before i leave but also not a big deal.
at home i use moodeaudio attached to my setup or kodi
I use the schitt magnius and modius as my DAC amp and the meze 99 classics as my headphones (though im looking on upgrading because my dacamp is overkill)
Spotube is my music player but by necessity im looking for something better if somone wants to recommend 👀
Love the Meze 99 Classics, worth every penny!
24bit 96kHz FLAC (purchased from Bandcamp & HDTracks) ->
JRiver Media Center software player ->
Merging Anubis Pro DAC ->
Coleman Audio M3PHmk2 passive monitor controller ->
Pass Labs X250 class A solid state power amp ->
B&W Nautilus 802 3-way floor standing speakers
Or if from vinyl
KAB modded Technics SL1200mk2 ->
Shure V-15MR cartridge ->
Simaudio Moon LP5.3 balanced preamp ->
(in 20' x 14' x 9' room with bass traps, absorbers and diffusors by GIK, ATS, and Auralex)
My current chain is Tidal + Schiit Asgard DAC/amp + Audeze LCD-X. Moved from Spotify to Tidal last month and will never go back. I definitely prefer headphones over speakers, but have really been enjoying IK Multimedia iLoud Micro Monitors.
Moved from Spotify to Tidal last month and will never go back.
Will you consider moving back if Spotify bring HiFi as it announced? I mean no once can beat it's catalog.
I definitely can't argue about the size of their library! While the continued dragging of their feet on HiFi was frustrating (years of telling us it was coming), the thing which finally drove me away is their constant tweaking of playlist and queue management.
I mainly use their desktop client and controls would disappear with each update- no way to block songs, inability to remove a song from auto generated queues, playlists not syncing between devices, songs being weighted in a shuffle. I made a post on their forums about the missing options for their autoplay queues- their response was that while there was no button or context menu option to remove a song, I could select it and use the delete key. I just gave up on whatever type of user experience they want me to have.
Ah, makes sense.
Did you look at Qobuz too? Seems pretty decent
I did! I do think it's a great alternative, but when moving some of my playlists over, I saw too many missing songs. They were my more niche playlists/genres so I was kind of expecting it. Tidal didn't have all of them either, but did have more so I decided to go with them.
With a drink.
Sennheiser 6XX
Plex, though I do occasionally listen to online radios using my podcast player
MusicBee on PC
Vinyl Music Player on my phone
Local mp3s and flacs work the best
I dabble with YouTube Music and music-map.com for music discovery
Haven't found a nice self hosted music streaming setup that I'm happy with (unsatisfied with the apps and features). I want a nice looking app (super subject of course) that supports offline play and ReplayGain. I'm super happy with Navidrome but not with the Windows/Android apps
Amazon music streaming has flac with their HD quality, I really like my Vanatoo speakers with optical in
If I want the highest quality streaming, then Amazon Music.
Otherwise, things I've purchased in 96khz or 192khz from ProStudioMasters.com
I work in the audio post industry, so I'm generally listening on my work rig either through Genelec speakers or Beyer DT880 Pro headphones, fed by a UA Apollo audio interface.
On the go: Truthear Nova IEMs + DAC via Sony Xperia 5 III LineageOS for microG phone or Shanling Q1 DAP (rarely Sony WF-1000XM3 if wireless is a requirement)
At home: Moondrop Variations IMEs + DAC via Moto M2 audio interface (all machines running Linux)
Music from: Bandcamp or Soulseek via Nicotine+, occasionally YouTube for discovery
Easier question to answer: how don’t I listen to music:
Out of my phones speaker.
I’ve got a few pairs of earbuds, headphones, headphone dacs, and 2.0 system attached to my TV,
Oh and the “premium” audio system my Prius came with. Spotify, Apple Music, Plex… wired, wireless
Were you looking for something specific?
Is there an active community outside of Reddit and headfi where one can talk about this? I haven't seen anything on Lemmy.
The people there tend to discuss things which can go slightly over my head, but that's something I appreciate since it gives me things to look into and learn.
Thanks a bunch
Sources:
• FLAC on Plex or Jellyfin
• Apple Music set to highest quality
Output:
• Bluetooth to Car speakers when driving
• AirPods when walking
• AppleTV to Denon receiver to Polk speakers when playing music for whole house (occasionally I use a turntable here instead)
• iPad to a 2.1 Edifier setup playing VSQ when falling asleep
Not often enough:
• Technics SL-1200MK5G or SL-1500C to my AKG K240 Studio headphones
• high
Edit:
Now I’m gonna have to go back through all my old Lemmy posts because there’s so much info here it feels like I doxxed myself.
I also transfer all my music to my Hauwei watch GT2.
Edit: not sure if I count as an audiophile.
I use Neutron Audio Player which has a profile for my headphones but at the same time I don't really think Bluetooth could realistically be called audiophile.
So yeah I do the best with what I've got but don't really go crazy with it.
At the houses of my audiophile friends.
I’ve got a shitty little apartment, no home system. But I drive Uber, and I take great pride in always having excellent music playing when I’ve got a passenger.
I play spotify through usb to the car’s system. It doesn’t sound so great.
But most of my friends are more well off than me, and have great home sound systems. One’s got an underground theater, with a super heavy door. You close that door, the silence is like being in a tomb.
I still have my iPod. It works great.
Sadly I have to rely on Windows to continue filling it up with songs. But it sounds better than my phone, even with AAC files (I have quite a lot of ALACs on there but they don't make a difference sound wise).
Really wish that Apple revives the iPod to target it specifically to audiophiles.
I match the music to the speaker. I don't buy gear to match the music.
At home I have a set DML panel speakers set up in a 2.1 channel system with a subwoofer. The panels themselves are made of EPS polystyrene that has been sanded down and coated in wood glue, are about 1 meter tall, 30 centimetres wide and 2 centimetres thick (3 foot 3 inches tall, 1 foot wide and 4/5 inches thick) and have rounded edges and corners. Each panel has a Dayton Audio 10 watt exciter mounted to it on the location recommend on their website. The subwoofer is a ported down firing unit, which I have placed in the corner of the room for corner loading.
Not sure if I count as an audiophile but here’s my list o’ stuff I’ve acquired over the years and like the best:
My house had speakers built into the ceiling when I moved in so I have a Denon AVR-S760H amp and play audio through the surround sound with an AppleTV, record player, or whatever other source. (I forget the record player model but it’s just one of the mid-tier Sony ones.) I also have some Sennheiser HD 598 headphones that I love the sound on. They’re open back so not appropriate for travel but if I’m alone and not in the living room, that’s my go-to.
I really like Sennheisers and I eventually splurged on a pair of 4.50 SE over the ear ones for travel. They have noise cancellation and a closed back so they work great on flights or trains. I like them a lot.
I also have some Beats Fit Pros that I use a lot. Most earbuds don’t stay in my ears very well so the little nubbin hook on the Beats Fits is really what prompted that decision but the audio quality ended up being perfectly fine for the form factor. Sometimes, you’re exercising or just listening to a podcast or a work call. They ended up being a good purchase.
NAS -> ALAC, high-res files -> Nvidia Shield (via Plex) -> Yamaha RX-A8A receiver -> Polk Monitor 70 tower speakers
At home mainly records. Rega P6 as a player, marantz amp and totem speakers or koss esp/95x headphones.
On the go Qobuz on my phone to cayin ru7 dac and campfire Andromeda iems.
I buy it if I can find it on a platform where the money is actually going to the musician. Then, I upload it in CD quality FLAC format to FunkWhale, and also add it to the SD card in my DAC (a Shanling Q1). Where it's convenient I listen on the DAC, where it's not I stream through FunkWhale.
Not sure if I merit being called an audiophile, but...
Huge collection of mp3s ripped from CDs. Stored locally, currently using a Unihertz Jelly Star as a glorified digital audio player, running BlackPlayer EX, which I like for it being a good mix of minimalist and giving me freedom to customize as much as I feel I need. When I'm using headphones, I want my ears uncovered so I use Shokz bone conduction headphones.
Very loud.
Car mostly now. 2.5” Pioneer dash speakers, 6.5” Polks and 6.5” Kenwoods, 10” Pioneer sub and monoblock amp. About a million times better than any upgraded audio system in a new car. Crystal clear audio, very tight controlled bass. It’s sublime.
Otherwise in the house from Apple Music Lossless through the Sonos Arc+sub gen 3+ surrounds and HomePod minis, very rarely through the home theater Atmos syste (Yamaha TSR-700 and Onkyo fronts and sub, and Niles in ceiling surrounds).
I’m a firm believer in not wasting money on expensive amps and gear for marginal gains (pardon the pun). I went to school for audio engineering and have mixed on $100K speakers. They sounded phenomenal but I have more fun in my car with its ~$600 system than anywhere else. Audio is very psychoacoustic. When you’re groovin’ the system almost doesn’t matter.
I've got speakers for every occassion. Several in-ears, over ears, monitor phones, Bluetooth speakers, and main amp and stack. Because of this it all sits in that top of middle range to bottom of high range, else I'd be broke.
Mainly use Spotify and vinyl.
Mainly use Spotify and vinyl.
Talk about chalk and cheese…
Well, I don't bother with lossless anymore outside of my own production. Not everything in the house is hooked up to EQs and I'm not hauling them out everytime I play music. So it's almost pointless these days.
TIDAL, K3/K7 (the K7 isn't portable), Sennheiser HD600s, and a pair of Hifiman HE1000s that I just bought. Both DACs work on all of my devices.
Openback neutral headphones. Listen to music the way it was mixed. Obviously higher bitrate is better, but I cave in to the convenience of streaming and am content with minimum 320kbps for casual listening. Definitely lossless for critical listening.
I lose some information because of the Android resampler however most of my library is 16/44.1 flac. Although my collection of 24/96.2 is growing.
Buy albums on Bandcamp, Stream from Tidal, get a USB DAC + either vintage amp/speakers (almost anything pre 80 is good) or modern amplified speakers.
Bluetooth Xiaomi headphones because convenience is king (and I can't afford to pay more than $200 for audio equipment lol)
in silence.
Dynaco ST-70 (stereo tube amp, mine is maybe 1960s?), 8Ω tap to either Klipsch Heresy II or Vandersteen 1c speakers.
I've had the Klipschs for 20+ years, so to me they're sort of reference/completely neutral speakers. (I know Klipschs aren't everyone's cup of tea though.)
PC (MPD with Ario frontend) -> SMSL DO100 -> Rotel A11 Tribute -> KEF Q150. I'm upgrading to KEF LS50 Metas next week, can't wait.
The best quality that is convenient.
On the go? Bluetooth headphones from Spotify.
At my desk? Open back sennheisers from the FLAC from the NAS, or Spotify.
Any sufficiently high quality audio stream from my Plex or Tidal, always set to max volume in app/OS settings -> Topping D30 -> JDS Atom -> Sennheiser HD6XX.
Good enough for me.
Not an audiophile, so bexcuse the ignorance, but what is the logic of max volume in app?
The goal is to send the exact, unmolested digital samples from the file out to the DAC, which then sends its analog signal to the amp where you worry about how much to amplify that signal for listening.
When you set everything to 100% volume in software, you can assume that there is no software doing anything to alter the digital signal before sending it to the DAC (scales each sample by 1.0). But when you're under 100% volume in software, it assumes you don't have any analog control over the volume, so it needs to step in and alter the digital signal so that it shows up quieter to the DAC (ex. scaling each sample by 0.25). Depending on how that's implemented, it can result in losing resolution and thus quality of the signal.
I think this mattered more on older software that's more likely to use a smaller bit depth, but bugs happen, so why risk it and spend those extra cycles on a process that can only result in a worse signal, right?
There’s some confusing stuff in this response so before I get into the weeds, for all the people reading out there: you don’t lose quality by using your operating systems volume control.
Okay, with that out of the way, let’s say you wanted to adjust the volume of a digital stream that’s composed of samples. Each sample represents the original analog signals voltage at that slice of time when it was encoded. The number of slices per second is the sample rate, expressed in kilohertz and the voltage of the original signal is converted to a number, which is stored as a binary value whose length is expressed in bits, each of which can be either a one or zero and is referred to as the streams bit depth.
So you could have a stream whose sample rate is 44.1khz for example and that would mean that it was sampled 44,100 times per second. That same stream might have a bit depth of 16, and that would mean that the original signals voltage level was divided into 65,536 possible values. Depending on some other factors, that stream might just be cdda (a compact discs digitally encoded song information).
Now let’s say you had a computer that was handling that stream and was asked to reduce the volume of the stream by half by a user who can only stand to listen to it at that volume.
One way to do that job would be to decode the stream back into an analog voltage, attenuate it, recode it and then send it on its merry way. That would incur a decoding operation, require routing of that signal to either dedicated hardware to perform the attenuation and send the signal back and an encoding operation to make that now half as loud signal back into a digital stream that can then be sent wherever it’s destined.
Another way of handling that operation is simply dividing every slice of the streams 16 bit component by two, something that computers are very good at doing quickly.
It should come as no surprise then that the latter process is generally how it’s done.
But does that reduce quality or result in worse signal? That’s the question, right?
Well, any variation of one bit or less could be essentially deleted. A person could say “ah hah! The signal has been degraded!” And they’d be technically correct, but it wouldn’t matter.
In our example, the computer whose hands are all over our precious data stream is sending that adulterated information to a dac, which true to its moniker will convert the signal from a digital stream into an analog signal. That analog signal will then be sent to an amplifier with an analog volume control and from there to a set of speakers.
The amplifiers analog volume control is a resistor in the shape of a 3/4 arc with a wiper that can move back and forth across it, allowing anything put in one side to be resisted (or in the case of our ac signal, impeded) a varying amount depending on the users selected position of the knob attached to the wiper.
Okay but what is resisting a signal though? Well, a resistor will reduce the voltage between its two ends proportional to its resistance, measured in ohms. More ohms means more resistance.
For the purposes of our example, let’s assume the user has chosen an amplifier and dac combination such that the amplifiers volume control at minimum setting applies the minimum resistance necessary to completely attenuate the dacs maximum output and is not applying any resistance at its maximum setting. In other words, that it all works as expected and is perfect.
In this case, what’s the difference between sending a stream with data corresponding to a .5V signal that gets amplified as opposed to a stream with data corresponding to a 1V signal that goes through a resistor to bring it down to .5V before being amplified?
nothing
In fact, the digitally attenuated stream will probably sound better (closer to the original) because it’s not subject to the bourns/alpha ppm lottery!
Now.
Don’t let this stop you from listening to music however you like. My ass is itt admitting to using 40 year old record players to make sounds to cook to. But don’t worry about the computers volume control.
Yes, if everything aligns perfectly, there is no impact. The bit shift would be when you set the volume to exactly half, but that's probably not going to be the case. The app volume control alters the signal slightly, multiplied by the OS altering it slightly, which has a virtual certainty of introducing a floating point rounding error on every single sample, so now the ratios between your samples is ever so slightly different. And for what reason? What did that operation gain you?
And no you're not going to hear a difference, but the point of being an audiophile is less about hearing a difference, and more about good quality preservation practices.
okay, lets consider the worst possible rounding error in a 16 bit division operation:
i'll divide the level of one sample by some number that will not divide evenly, lets say three, and consider the impact of the rounding error. for the purposes of making it so I know there will be a rounding error i'll choose a number that three has a really bad time dividing into, say 65536.
65536/3=21845.3 with the .3 component repeating. perfect, that's exactly horrible!
so if we were to just do the simplest rounding possible to fit that into a 16 bit integer, the decimal component is dropped, rounding down to 21845.
but what is the significance of that error? a samples volume level that's one integer value off introduces a .003% error in level, but this isn't supposed to be 21846, it's supposed to be 21845.3. so the error that's introduced is .001%!
that's a pretty tiny error, but what if it was periodic and consistent enough to produce a harmonic component? that'd create harmonic distortion!
lets say there's a periodic and consistent rounding error that has a frequency of 1000Hz. so every thousandth of a second, the rounding caused by the volume being set at 1/3 causes a rounding error and makes a sample off by .001%. such a repeating error would introduce a harmonic component into the signal that the dac produces and be measurable as harmonic distortion at 1000Hz!
but how measurable? well if, for example, the harmonic component of the signal introduced was at it's absolute worst, and oscillated between a positive going error and a negative going error, it could introduce a peak at 1000hz of...
.002% of your dacs DBFS. so far below the noise floor it's immeasurable.
even if you had two software volume controls set at 1/3 daisy chained together doubling that error it would be immeasurable. although if we picked a computer software package to use instead of a bunch of hypothetical worst cases the total volume of a signal would be summed and then applied once in order to minimize just this problem. the people writing that software are pretty smart and doing that saves their program a step!
but measurability or audibility isn't the point, as you said. the point is to reproduce the sound as accurately as possible! so it really doesn't matter how tiny the effect of rounding errors due to prime denominator volume settings is if it's larger than the effect of the analog volume control that whatever signal the dac manages to reconstruct from our mangled stream is put through. we're trying to adjust the volume down to a comfortable listening level, after all.
so how bad is the volume control? well, if i were to go to mouser and look at the potentiometers section, i could choose one with a tolerance as low as... .5%! and that's a three-gang model that costs $50!
but what if i used a precision potentiometer? why, there's a .15% tolerance part that's available for the very reasonable price of $825!
okay the precisions are out of my price range, but those $50 ones could work. tolerance is just a number anyway, right? we want linearity! what does the datasheet say about linearity... 2%!
that's not even considering the amplifier design's distortion. lets assume it's perfect.
so just passing the signal from the dac through the amps volume control causes possibly 200 times more error and distortion than adjusting the volume control in the computer.
i get the pursuit of the best possible reproduction, but the computer volumes got those cermet pots beat hands down.
Interesting point. What about the case where you have your digital volume set to 1%? Would this not squeeze the samples into 1/100 the dynamic range? If I set my volume to 1% it seems to me like those samples now have to all exist within the bottom 1% of the 16b range. Do you not lose at least 5-6 bits of precision on your signal doing this?
You don’t lose precision when you lower the volume (in either an analog or digital realm). You lose actual information!
Let’s say you have a recording you can only listen to with your volume at the 1% setting. Analog or digital, it doesn’t matter.
Your whole system has an acoustic noise floor at something like idk, 10 acoustic decibels. That is a really charitable number because I’ve never measured one that low and it directly corresponds to the loudness of another healthy persons breathing at rest. To give you an idea of how quiet that is, the acoustic decibel scale generally puts a ticking mechanical watch at twice as loud (20 decibels).
I don’t want to talk about decibels because I don’t want to explain the math in the detail I’ve been giving these posts, but we gotta at least cover a little:
Decibels are the measure of sound energy, their scale is logarithmic, so the base of the log function determines how many of decibels make for twice as much.
There are different decibels for measuring in different mediums with different references and they even use different logarithm bases.
Acoustic decibels are log10, so that 20 decibel ticking wristwatch is twice as loud as a person breathing and half as loud as whatever the workplace safety scale says 30 decibels is equivalent to.
Okay so now that we have a floor, we need to establish a ceiling. Let’s say that you did everything right and hooked all your stuff up, turned the volume on the amplifier all the way down, put your headphones, played a maximum volume test tone, maxed out the volume on the software, then turned the amplifiers volume control up until it caused you immediate physical pain. If you have really good hearing, that’s 115 acoustic decibels. Let’s say you got to 120 with the amplifiers volume control up all the way.
Okay, so the noise floor of your headphones on your head is 2^11 as quiet as the loudest sound you can tolerate hearing.
Now you set the volume control to 1%. Doesn’t matter which one. Everything gets 99% quieter. The parts of the signal that were 120 decibels before are now 1.2 decibels. They have been divided by 100, and it’s possible that rounding errors have added .006% error to their harmonic content. .006% of 1.2 is .0072 decibels. Not only is the loudest sound you can stand to hear now quieter than a person breathing, it’s below the noise floor of your system. Far, far below the noise floor. And any rounding error from dividing by 100 is as well!
Okay but what happens when you’re listening to music though? Let’s put aside all that hypothetical stuff and get rockin! Instead of talking about test signals and boring crap, let’s talk about a song!
So same established setup from before, but now you’re listening to a recording of someone playing the banjo while rocking in a chair. There’s a lot of different harmonic content in this signal, the birds chirping, the persons breathing, the wind, the chair creaking the boards of the porch and of course, the instrument itself!
All these different things are at different volumes and they represent components of the harmonic content of the signal you’re listening to. When you turn the recording down, you’re attenuating the signal. All of the signal. If you apply enough attenuation through your chosen volume control to lower the level of the banjo by 40 acoustic decibels then all the other components of the signal are lowered by 40 decibels too. If they were previously 50 acoustic decibels through your headset, they’re part of the noise floor.
The quietest information is simply lost.
Edit: there are massive amounts of information that have been simplified so much as to make this post incredibly inaccurate. Please do not use this as a reference for understanding how we measure or interact with sound. I’m sorry for not going into greater detail but it’s too early to explain the relationship and history of acoustic decibels and decibels per volt.
I'm sorry you have to type so much, I am familiar with most of it, but I appreciate your effort to make sure we're on the same page without being a douche about it lol. It sounds like we're saying similar things, but I don't understand why lower precision is different from losing information. To me, that's the same thing, it's a lossy operation.
So the thing is, I have a pair of desktop speakers without any physical volume control that I primarily use for convenience. And for whatever reason, a comfortable listening volume with them is between 1-8% in the OS volume control. I guess the internal amp is just hardwired to be way too loud?
Anyway, I assume that this setup is resulting in objectively lower quality output than if I were to have a 100% signal going to a decent quality DAC/amp with analog volume outputting to the same speakers. And not in a "technically" kind of way, but in a very real "we just crushed the signal into 1/25th of its original scale" way. Would you agree? Am I mistaken?
no worries. i've been enjoying going back through this. you're basically me 25 years ago emailing the winamp ppl to find out what volume control i should use to turn down (for what!).
so there's a misconception here between compressing a signal and attenuating it. imagine you are looking at a frequency spectrum chart of some song instead of listening to it. it's got some loud sounds, which show up as big peaks on the chart, and some quiet sounds which show up as small peaks on the chart and there's a noise floor which is the stuff in the chart that's not a peak at all.
if you plug a potentiometer in between your signal and the spectrum analyzer and turn down the volume, youll see all the peaks, loud and small, be reduced in amplitude by the same amount. this is called attenuation. the quieter sounds could be reduced until they are part of the noise floor and become imperceptible while the louder sounds would still show up.
it doesnt matter if you achieve attenuation by dividing the 16 bit level component of a stream of samples or by using a resistor as a voltage divider. the quiet and loud sounds are affected equally. those two ways of achieving attenuation function the same because they are performing the same operation.
now lets say you plug a rack mount compressor effects module in between your signal and your spectrum analyzer instead. you could apply more compression to the signal and achieve exactly what you describe, a smaller distance between the quiet and loud sounds, reduction of the original scale, removal of dynamic range, effective bit depth reduction! it would be actual factual "we just crushed the signal into 1/25th of its original scale".
and if you used a module (or software package) with the capacity for it, you could tie the compression ratio to a gain control so that the compressors output got quieter when you turn the compression ratio up, resulting in more heavily compressed sounds at a quieter volume. that's a neat little mastering trick to make recordings sound "lively" and "intimate". makes all those pick scrapes and finger swishes stand out alongside the plucked strings.
you could also do the inverse, make all the quiet sounds louder, so that the guitar is as loud as the kick drums' transient and it would make your whole song sound much louder and stand out better against background noise in a difficult listening environment like a car radio or cell phone inside a solo cup.
there are even modules that do the opposite, called... expanders! they do what you might expect, increase the dynamic range between loud and quiet sounds. a company called DBX made models for use in home stereos in between tape decks and the amplifier in order to reduce the noise floor of tapes.
but it's none of that is attenuation, the operation that your volume control provides.
and you're correct, both compression and attenuation are lossy operations no matter if they're done with analog electronics or by a microprocessor operating on a buffer somewhere in memory. the difference is that attenuation is literally required to prevent permanent hearing loss and possible equipment damage, while compression is not.
it doesnt matter if you achieve attenuation by dividing the 16 bit level component of a stream of samples or by using a resistor as a voltage divider.
This is the part where I'm not following. In my head, if you're using analog hardware of sufficient quality, you can attenuate the signal to be very quiet, but still preserve it's dynamic range. In fact, the DAC is already outputting a very weak, but faithful analog reproduction of the signal, and an amp with a decent S/N ratio is able to bring that very weak signal up to a listening volume without introducing enough noise to matter.
Hypothetically, if for some reason, you took the signal post-amp, used a pot to attenuate it again down to the energy of the post-DAC level, and again ran it through another amp you would theoretically have the same signal still (I understand that in the real world we would start amplifying noise and the signal would degrade, but stick with me). Nothing about the process necessarily introduces noise and thus destroys the signal, you're only limited to the quality of the components at that point. If you had an infinite chain of theoretically perfect amps and pots, you could repeatedly attenuate and amplify the signal forever without ever losing any quality. It's an analog process that theoretically preserves the signal, +/- some amount of error due to physics.
Meanwhile, 16b is 16b. If you start shrinking all samples relative to each other (ex. down to 1/64 the original volume, or 10b of resolution), different values inevitably have to clamp to the same values (fitting 64k values into 1024 values), losing information and resulting in poorer quality. If you then try to send that 10b signal through a DAC/amp to achieve the same listening volume that you would have had before digital attenuation, it's just a 10b signal bit shifted up. All your LSBs are 0s. You can't possibly attenuate digitally, and then amplify it in any way and hope to get the same signal back. It's a discrete math process which destroys the signal by design.
Would you agree?
the effect of attenuation is the loss of intensity of signal.
loss. it goes away.
attenuation is a lossy process. information in the signal is literally absorbed and radiated away as heat. it cannot be reconstructed because it's gone.
it isn't an analog process that theoretically preserves the signal, it's an analog process that explicitly destroys a component of the signal.
but what if it wasn't...
okay, lets assume for a second that you have a signal with the same harmonic content as one of my previous examples, a high peak when viewed on a frequency spectrum chart, a low peak when viewed on that chart and everything else. these three parts of the signal represent the loud, quiet and "silent" parts of the signal respectively. unlike the previous example we'll let our noise floor for the silent parts be infinitely low. for now. so you start hooking up your perfect amps and pots in line and setting them all to 1% or so and listening. it's sounding pretty good at first, but once you get a few deep, you start getting white noise and clicks and pops and all kinds of craziness.
what the hell! all this equipment is theoretically perfect, why is there noise? it can't be coming from the perfect equipment!
it's not. it's coming from the medium. in our theoretical example all these amplifiers and pots are hooked up with conductive wire. the signal has to propagate through that wire from component to component. atoms of copper are being excited and losing their excitation in proportion to the signal. their state of excitation is being amplified over and over again. the noise is in the wire. by amplifying it over and over again you made it audible. you can't ever escape it. signed, listening to noise gang. come to my modular synth show.
okay, so now that the possibility of ever attenuating a signal without losing information is hopefully put to rest, lets turn to the digital attenuation of the signal in comparison.
level attenuation over the digital domain is also a lossy process. what's being misunderstood here is that the levels aren't being shrunk relative to each other, they're each being divided and the signal that's reconstructed by the DAC no longer contains the quiet parts.
just like those quiet parts were absorbed and radiated as heat by the resistor, the digital version of attenuation does away with the need for all that physics crap and simply deletes them from the stream.
if the levels were being shrunk relative to each other, you'd be compressing the signal like when you use the bitcrusher pedal for your guitar and there would be lots of harmonic distortion. but attenuation and compression are different processes and have significantly different results.
consider a quiet sound, your 1/64th volume signal. a sine wave. its encoded to represent 1/64th of the maximum level of the adc's input because when it was recorded, it represented 1/64th the maximum level of the preamp/microphone/whatever that was plugged into the adc.
is the quiet sine wave of lower quality than one that's using the full bit depth of the adcs output because it's intended to represent the maximum level that the adcs input saw from the preamp/microphone/whatever?
of course it isn't. it just wasn't loud.
and if your loud sine wave was electrically generated by a theoretical perfect function generator which contains no distortion or other sonic content before being sent to the adc, would it be more damaged if it's amplitude were divided by 64 before being decoded or if it were decoded and sent through a resistor whose value was chosen specifically to dissipate 63/64ths of it as heat in order to make it as quiet as the quiet sine wave?
of course it wouldn't.
to your last question, let me rephrase it into something I can agree with: you cannot possibly attenuate and then amplify in any way and hope to get the same signal back. It’s a lossy process which destroys the signal by design.
My ears.
No just joking, YouTube music mostly. It's convenient, available everywhere, has a large catalogue, and good enough quality for me.
With all respect you’re not the definition of an audiophile at all. If anything you’re kind of the opposite
Not everyone can discern the difference between a 96KHz FLAC and 256kbps AAC. I can't. But I still can (barely) tell the difference between 256kbps AAC, and 96kbps AAC.
But I can tell if a song was well-engineered or a mess.
I believe those who can't discern the difference between bitrates (especially on high bitrates), but have the appreciation for good music, good mixing, and good mastering, can still be considered audiophile.
That's not the comparison at hand, we're talking YouTube audio compression vs any actual music track.
Especially when your browser or application requests a high quality bitrate, youtube compression is opus 128.
A person could make the argument that it’s not lossless so it’s not worth listening to, but opus is extremely high quality especially at that bitrate.
If you wanna try it for yourself, take a flac or whatever, upload it to yt, then use something like yt-dlp -x that defaults to the highest quality to redownload just the audio stream.
YouTube Music Premium offers AAC 256kbps as the highest quality.
Format ID 141: https://gist.github.com/AgentOak/34d47c65b1d28829bb17c24c04a0096f
Opus 128 is only for the audio of YouTube videos. Not YouTube Music.
and according to that same link it's 160, not 128 (format id 251!). someone else pointed that out itt.
one of my downloads had an average bitrate of ~140 when queried with mediainfo, so i believe em.
I don't have the premium account, what's aac256 comparable to?
AAC 256 should be at least on par with MP3 320 CBR, might also be on par with ogg vorbis at the same bitrate
As I get older and the abuse I put my ears through starts showing up, I completely agree. After upgrading my music library to FLAC from VBR mp3s, I stopped having the, "Oh! There's a subtle instrument going on in this part of the song!" moments.
It doesn't stop me from trying to listen to the highest quality music formats that I can get my hands on, but I 100% know if I think there's a difference to my mid-40s ears, it's probably a placebo.
Yes. As a lifelong musician (live & recording), you’d think I’d be more fussy about audio quality…
But I’m just not. Just like the 4k vs 2k “debate”… It’s all about CONTENT.
Also a musician here. I cared a lot when I was younger, but I have so many other more important things to care about now. You only have so my capacity to care about stuff in your life, and the quality of my music doesn't even come close to mattering these days.
FLACs from CDs, deemix-gui, qobuz-dl, and Soulseek. 102,000 songs. Play at home with Logitech Media Server. On the road I've transcoded it all to 128kbps Opus so i can fit it on a microsd card and I play it with PowerAmp. I mostly use Blessing2 Dusk earbuds with a Shanling MW200 bluetooth neckband, but sometimes also I use Focal Clear OG open-back over-ear cans with a qdelix 5k for bluetooth.
Audiophiles don't listen to music, they listen to their headphones
„Audiophiles don't use their equipment to listen to your music. Audiophiles use your music to listen to their equipment.“
Alan Parsons
I dunno if that's actually an Alan Parsons quote but up vote for any mention of his name. Does sound like something he'd say.
FLACs through PlexAmp, either to nice headphones ($500 range) or two channel stereo into some decent speakers with a decent subwoofer. I'd like to upgrade to "full range" speakers one day and save the subwoofer for movies.
PlexAmp does FLAC when connected to Wi-Fi but I have it set to transcode if I'm using mobile data.
At home it gets played through Chromecast Audios (R.I.P) which keeps it all digital until it hits my receiver.
Spotify -> MOTU M2 -> HiFiMan Ananda non-stealth
"High resolution" audio is completely useless for listening (16 bit 44.1 kHz is the best it gets) and there is little value in lossless encodes for listening purposes too, so I don't get the point of all those "Hifi" streaming services.
If you own lossless encodes, I guess it doesn't hurt to use them even for listening as storage is cheap these days.
Speaking of which, I'd like to switch to purchasing my music though because Spotify will certainly continue on its path towards full enshittification. I want to be in a position where I own all my favourite music before Spotify will be infected with ads on premium plans. Oh and artists are somewhat more likely to be paid a little for their work that way (I hope...)
I plan to use the free YT music for discovery at that point.
Completely full of ads already, I routinely get promoted podcasts and gig ticket and merch notifications despite them being turned off.
I started using Spotify lite on my phone. And thankfully, there's plenty of alternative clients on desktop (such as ncspot). No crap UI elements, just playlists.
Spotify through Sonos at home and work. Spotify on Google earbuds when out and about.
I used to really love music discovery on Spotify. I now find it's the same ald songs over and over. It finds what you like and reinforces that rather than gradually expand it.
I'm in the same boat. For years now it's felt like every daily mix and discovery playlist is 10 songs I recently just listened to on repeat and then 2 songs that aren't even tangentially related and I'm left questioning why they were being shown to me.
I agree on the discovery being crap on Spotify. I started to listen to the podcast NPR new music Fridays, and get my discovery that way nowadays.
I have converted all my CDs to FLAC and I mostly listen to my music collection in stereo speakers instead of headphones because I find the sound more natural. I have built my sound system around the moOde audio software.
Music collection as flac, navidrome as streaming server, symfonium as android app and B&W P5 or B&W Pi7 S2 for headphones.
I really wanted to like symfonium (even tho its not open source), bc it is a beautiful client, but it is a battery hog. I had to go back to ultrasonic.
I actually found all the subsonic clients to be quite heavy on my battery, so I just stuck with the one I liked the best.
FLAC's on NAS. Bluesound Node to stereo system, controlled with Roon. PlexAmp when remote.
Tidal is actually giving their lossless plan to their lower tier subscription, just got an email about it. Pretty nice.
At home: Spotify through Amazon Fire TV through Klipsch The Fives.
On the move: Spotify through Jabra Elite 4 Active.
In the bathroom: Spotify through UE Boom.
I really want to ditch Spotify, but in the meantime...
Well, TIDAL just got some price cuts, and their library is pretty comparable. Just in case you didn't know.
Just read that today! Thank you.
Same, but I want to export my playlists and liked songs from Spotify. Going through that manually atm seems like too much of a hassle.
If you plan to move to another service, there exists a number of tools to aid in moving playlists between streamers. It is really easy, once you find a good one.
Helped me break the feeling of being locked in due to have 100s of playlists.
Tried one service but didn't work with some Spotify lists, like the yearly ones. Any good recommendation that might include these as well?
Qobuz for me.
Best streaming sound available but I had some skipping issues even on very good connections and options for auto Playlist generation and new music discovery was way behind other services. Great if you always knew exaewhat you wanted to hear, but I went to Tidal and their focus on quality is better than most other services but the music discovery algorithms really are quite good, I find myself more eager than ever to tune in to a streaming service.
Budget audiophile here: I wear Superlux HD681 semi-open back cans paired with a Creative G6 DAC/amp.
The headphones are $25 but have the the most realistic soundstage I've ever heard in a pair of cans, even better than $500+ ones. Pinna activation is almost perfect; feels more like being surrounded by speakers than wearing headphones. Makes them amazing for gaming and movies, but not the best for music due to harsh siblants in the 12kHz range, which I've managed to EQ out a bit using Equalizer APO. Nice neutral sound otherwise, mids are almost perfectly flat and bass is tight—yet full—extending well below 20hz. Honestly you can't do better without spending half a grand or more.
Ehhh, I'm ballin on a budget, so take that into account.
Generally, if I really want to sink into the music, I'm going with either my lgg7 and my beyerdynamic 770 80 ohm; or whatever device can connect with my usb DAC, a fiio q3.
I do have other options, but that's my main listening because I simply don't have the budget to do a proper system with how little I get a chance to listen to music away from headphones. My computer has a decent sound card, and some klipsch speakers that aren't bad. There's a home theater unit with cd/bluray hooked up, as well as the shieldTV, and the ability to connect via Bluetooth or cable to whatever device I prefer.
My car is decent, but not audiophile level. More bass focused than anything else.
I do have other headphones. Some tin t2s, some sonys, an old set of koss, that kind of thing.
File wise, its flac and opus.
I use poweramp and/or usb audio player pro. I prefer poweramp, but the other does bit perfect, which I do like on occasion, and it's more DAC friendly.
I'm happy with the options I have, all considered. Most of it was picked up either on sale or used. I would save up while shopping, then get the best I could get when I was ready. But the key to me is that when I want to, I can listen to anything I have and hear the nuance of it. The sound is as clean as I can get it on my budget, and in all reality, my old ears can't make use of anything fancier.
You spend almost fifty years living and listening to it loud, you aren't going to get much bang for your buck out of the really high dollar, precise gear. Hell, I can barely tell a difference between lossless files and mp3 om any given listening method. It's there, I can still hear a difference, but it's barely there for me. The better gear helps, but not enough to keep upgrading for tiny changes.
I use deemix to get songs and jellyfin/finamp to listen on my phone. I do miss the discovery of new music from things like Spotify or YouTube music. If anyone has suggestions for music discovery I'd love to hear about them.
Open the Nicotine program that connects to the Soulseek network, then chat with the heads on there. Name a few artists you like and they can hook you up. The most knowledgeable music listeners around. Pretty sure you can search for ppl who have files of an artist you like, and then view their entire library. (NB. Been 10 years since I've used it, so YMMV)
https://nicotine-plus.org/
Seriously though, the real answer is to resurrect whatever Audiogalaxy was doing in their recommendations-algo, shit was dope.
CDs ripped to FLAC and streamed using Emby. I also use Amazon Music. At work I have a pair of ATH-M30x headphones I really like. At home ibhave some Sennheiser HD350, which are ok, but I don't like them that much as they're not that comfy. I prefer going through the hifi - Audiolab 6000A amp, Wharfedale Pacific Evo 40 floor standers and a Wiim mini. I also have a NAD C541i CD player. On my PC I go through a NAD C320 amp and Wharfedale Diamond 9.1 bookshelves.
I listen to music mostly on my computer and in the car. The car system is nothing special. I listen through either some ATH-M40fs cans, or Presonus Erie 3.5 monitors, which are honestly glorified bookshelf speakers, but decent for the price, IMHO. All running from my (older gen2) Focusrite 2i4 interface.
I used to listen in the train/metro/bus a lot more, but I now work remotely. That’s where I used Bluetooth stuff. No need to worry about the cable getting stiff in the cold or stuck in my winter jacket. I had a pair of Beats Studio 3 I paid less than $100 for that were pretty decent for the price I paid. The sound was as bass heavy as you’d imagine from the brand, but not terribly overpowering for casual listening, and the ANC in particular was pretty impressive. I also had some Anker wireless earbuds I got with a coupon on Drop (formerly Massdrop) that were good enough for listening to podcasts and having background music.
In terms of platforms, YouTube Music mostly, and a hand picked selection on Plex for stuff that’s not on there or that I want to have always available. The music discovery algorithms are completely useless for me though. It’s the one thing Spotify did better than YTM for me. The “My Mix” playlists and artist radios have been pushing me the same artists for months on end now. Want to know the ironic part? I discover most of my music on YouTube (not Music) nowadays…
Honestly as far as cheap small monitors go, I really don't mind the Eries. They're not perfect for sure but they give a generally balanced sound and I paired them with a nice mackie sub to get pretty decent frequency coverage. Certainly perfectly decent for producing a variety of music and generally for listening to things.
I’d put them in that gap between general purpose computer/multimedia speakers, and “proper” monitors. That product range used to be a pretty terrible place to be in, but these surprised me for sure. They’re flat-ish enough that I don’t feel like I’m shooting myself in the foot using them for light production work. The bass is indeed not quite it, but what can we really expect from drivers that size. I don’t have great experience using subs for production, but that’s probably me. They’re surprisingly good for the price point and form factor, at the very least.
Yeah I think flat enough is the right phrase. Their bass is definitely lacking but with a well configured sub (I set the crossover at about 80Hz I think) you can compensate. My only feeling about producing with a sub is unless you're in a very well acoustically treated room, it's worth checking your mix on good headphones and a few sets of speakers to make sure your interesting sub bass parts are actually coming through nicely. They are good though to really work out what's going on in the sub frequencies of your mix. Also makes it really obvious when those areas are getting muddy.
At home: FLACs ripped from CDs (prefer to buy albums I enjoy instead of Spotifying them) -> KORG DS-DAC 100 -> TEAC AX-501 -> Elac Carina BS243.4
On the go: The same FLACs on Pixel 6 Pro -> B&O Beoplay HX
I’ve got a special speaker assembly that I shove up my ass*. The bass response is particularly pleasing.
Tidal HiFi/medium tier ->Equalizer APO with just a tiny bit of tuning -> a xDuoo stack of USB DAC + hybrid tube amp -> Sennheiser HD560S
Definitely a little bit of overkill. But still overall fantastic budget, and do it all setup. Competitive shooters, movies, and music all sound fantastic.
My next goal is a multibit DAC + tube only amp -> something like a HD 6XX. Or maybe a good solid state -> planar magnetic headphones.
At Home:
On the go:
Everything encoded as Opus 128 kbit/s to fit on my phone. Played over Lypertek Tevy true wireless IEMs. Not really audiophile but tbh when I'm not at home I care much more about convenience as long as the audio quality is good enough.
also Qobuz, but at MP3 320 quality to save bandwidth
I wrote my own scripts to tag the music and encode it to FLAC and Opus and use syncthing to copy the files to my phone. So whenever I add an album to the library it will be available every where I want in the specified format without any manual copying involved. It's a little janky but has worked surprisingly well for years.
FLACs/Qobuz via Roon. I spend the most time in my office so that’s where my favorite setup is. LS50 Metas + SVS SB-1000 Pro + Peachtree GaN stack.
I also love my HD660s with the Bottlehead Crack tube amp I built.
PC Spotify -> Schiit Modi -> Schiit Vali 2 -> PreSonus Eris E4.5 speakers.
Or
Pixel 8 Pro Spotify -> "TempoTec Sonata HD PRO" USB DAC -> Meze 99 Classic headphones.
Does anyone think it's worth moving to Tidal for my music?
Also, I'm running out of space on my desk. I can put the stack of Schiit on top of a speaker with minimal effects, right?
I did recently and will not be going back to Spotify. There are so many small things with Tidal - actual patch notes each update, updates which clearly address user reported concerns/issues, straightforward playlist management and queue controls, an actual shuffle that isn't some weird interaction based algorithm, and of course the quality. There's been so many times I'll be listening to a song, which I've listened to many times on Spotify, and notice something in the backing track which I wasn't aware of or some aspect of a singer's voice or instrument which really pops and adds texture. They also have great recommendations and a Daily Discovery playlist. And finally - it's just music; no scrolling through podcasts or non-music this... Just high quality, easy to manage, music.
HD 560S for the cans. For my source, I use spotify, using my local library of FLACS for the stuff I like a lot, and just normal spotifly for everything else.
For earphones I have a set of KZ ZSN Pro X IEMs for when I'm on the go, when I'm at home I have my Audio Technica ATH-M50X.
On the player side I love InnerTune as a YouTube Music Frontend, while for analog I refurbished my father's BSR turntable and Phillips amplifier, both straight from the '80s
Mostly? I have uncompressed FLAC encoded music on my Plex server, and I listen to that streaming through over ear (Bose NC-700) headphones on a computer, or on our home theater system (Monitor UK, 2 stand speakers, 2 rear wall speakers, 1 subwoofer) with an Onkyo receiver.
I also listen to Tidal hifi a bunch and electronica on youtube because some of the Boiler Room and other club mixes are pretty dope :)
A technics changer or linear tracker. I think the changer has a shure cartridge still but the linear tracker has an at. Sometimes through a pair of numark ttxs with m447s and a rane.
Flacs on a server direct streamed to my source. Jellyfin is nice. for on the move I buy sony phones just cause they still have a headphone jack. I prefer to download what i want before i leave but also not a big deal. at home i use moodeaudio attached to my setup or kodi
I use the schitt magnius and modius as my DAC amp and the meze 99 classics as my headphones (though im looking on upgrading because my dacamp is overkill)
Spotube is my music player but by necessity im looking for something better if somone wants to recommend 👀
Love the Meze 99 Classics, worth every penny!
24bit 96kHz FLAC (purchased from Bandcamp & HDTracks) -> JRiver Media Center software player -> Merging Anubis Pro DAC -> Coleman Audio M3PHmk2 passive monitor controller -> Pass Labs X250 class A solid state power amp -> B&W Nautilus 802 3-way floor standing speakers
Or if from vinyl KAB modded Technics SL1200mk2 -> Shure V-15MR cartridge -> Simaudio Moon LP5.3 balanced preamp ->
(in 20' x 14' x 9' room with bass traps, absorbers and diffusors by GIK, ATS, and Auralex)
My current chain is Tidal + Schiit Asgard DAC/amp + Audeze LCD-X. Moved from Spotify to Tidal last month and will never go back. I definitely prefer headphones over speakers, but have really been enjoying IK Multimedia iLoud Micro Monitors.
Will you consider moving back if Spotify bring HiFi as it announced? I mean no once can beat it's catalog.
I definitely can't argue about the size of their library! While the continued dragging of their feet on HiFi was frustrating (years of telling us it was coming), the thing which finally drove me away is their constant tweaking of playlist and queue management.
I mainly use their desktop client and controls would disappear with each update- no way to block songs, inability to remove a song from auto generated queues, playlists not syncing between devices, songs being weighted in a shuffle. I made a post on their forums about the missing options for their autoplay queues- their response was that while there was no button or context menu option to remove a song, I could select it and use the delete key. I just gave up on whatever type of user experience they want me to have.
Ah, makes sense.
Did you look at Qobuz too? Seems pretty decent
I did! I do think it's a great alternative, but when moving some of my playlists over, I saw too many missing songs. They were my more niche playlists/genres so I was kind of expecting it. Tidal didn't have all of them either, but did have more so I decided to go with them.
With a drink.
Sennheiser 6XX
Plex, though I do occasionally listen to online radios using my podcast player
MusicBee on PC
Vinyl Music Player on my phone
Local mp3s and flacs work the best
I dabble with YouTube Music and music-map.com for music discovery
Haven't found a nice self hosted music streaming setup that I'm happy with (unsatisfied with the apps and features). I want a nice looking app (super subject of course) that supports offline play and ReplayGain. I'm super happy with Navidrome but not with the Windows/Android apps
Amazon music streaming has flac with their HD quality, I really like my Vanatoo speakers with optical in
If I want the highest quality streaming, then Amazon Music.
Otherwise, things I've purchased in 96khz or 192khz from ProStudioMasters.com
I work in the audio post industry, so I'm generally listening on my work rig either through Genelec speakers or Beyer DT880 Pro headphones, fed by a UA Apollo audio interface.
On the go: Truthear Nova IEMs + DAC via Sony Xperia 5 III LineageOS for microG phone or Shanling Q1 DAP (rarely Sony WF-1000XM3 if wireless is a requirement)
At home: Moondrop Variations IMEs + DAC via Moto M2 audio interface (all machines running Linux)
Music from: Bandcamp or Soulseek via Nicotine+, occasionally YouTube for discovery
Easier question to answer: how don’t I listen to music:
Out of my phones speaker.
I’ve got a few pairs of earbuds, headphones, headphone dacs, and 2.0 system attached to my TV, Oh and the “premium” audio system my Prius came with. Spotify, Apple Music, Plex… wired, wireless
Were you looking for something specific?
Is there an active community outside of Reddit and headfi where one can talk about this? I haven't seen anything on Lemmy.
One place I frequent anytime I'm looking for an upgrade or just general information is https://www.superbestaudiofriends.org/index.php
The people there tend to discuss things which can go slightly over my head, but that's something I appreciate since it gives me things to look into and learn.
Thanks a bunch
Sources:
• FLAC on Plex or Jellyfin
• Apple Music set to highest quality
Output:
• Bluetooth to Car speakers when driving
• AirPods when walking
• AppleTV to Denon receiver to Polk speakers when playing music for whole house (occasionally I use a turntable here instead)
• iPad to a 2.1 Edifier setup playing VSQ when falling asleep
Not often enough:
• Technics SL-1200MK5G or SL-1500C to my AKG K240 Studio headphones
• high
Edit:
Now I’m gonna have to go back through all my old Lemmy posts because there’s so much info here it feels like I doxxed myself.
General listening: Spotify + car speakers w/ EQ
Immersive listening: FLACs + HD 560S w/ EQ + Scarlett 2i2 + foobar2k
Modded Rockbox iPod with Wolfson DAC and 500gb storage. Around 150gb of CC music.
Audio Technica ATH-M50x headphones.
A bunch of old CDs, played on 2 shelf stereos or in car.
On phone, .opus files with Aeropex Aftershockz(average sound) or 2x UEBoom2 paired in stereo(slightly above average)
I’ve got 3 options currently:
I’m not crazy picky about my digital source as 95% of the audio quality comes from the hardware to my ears. It’s rare I notice a poor quality encode.
Buy on Bandcamp, listen with Strawberry.
Download from Deezloader, listen with Lollypop on desktop and Auxio on mobile.
I also transfer all my music to my Hauwei watch GT2.
Edit: not sure if I count as an audiophile.
I use Neutron Audio Player which has a profile for my headphones but at the same time I don't really think Bluetooth could realistically be called audiophile.
So yeah I do the best with what I've got but don't really go crazy with it.
At the houses of my audiophile friends.
I’ve got a shitty little apartment, no home system. But I drive Uber, and I take great pride in always having excellent music playing when I’ve got a passenger.
I play spotify through usb to the car’s system. It doesn’t sound so great.
But most of my friends are more well off than me, and have great home sound systems. One’s got an underground theater, with a super heavy door. You close that door, the silence is like being in a tomb.
I still have my iPod. It works great.
Sadly I have to rely on Windows to continue filling it up with songs. But it sounds better than my phone, even with AAC files (I have quite a lot of ALACs on there but they don't make a difference sound wise).
Really wish that Apple revives the iPod to target it specifically to audiophiles.
I match the music to the speaker. I don't buy gear to match the music.
At home I have a set DML panel speakers set up in a 2.1 channel system with a subwoofer. The panels themselves are made of EPS polystyrene that has been sanded down and coated in wood glue, are about 1 meter tall, 30 centimetres wide and 2 centimetres thick (3 foot 3 inches tall, 1 foot wide and 4/5 inches thick) and have rounded edges and corners. Each panel has a Dayton Audio 10 watt exciter mounted to it on the location recommend on their website. The subwoofer is a ported down firing unit, which I have placed in the corner of the room for corner loading.
Not sure if I count as an audiophile but here’s my list o’ stuff I’ve acquired over the years and like the best:
My house had speakers built into the ceiling when I moved in so I have a Denon AVR-S760H amp and play audio through the surround sound with an AppleTV, record player, or whatever other source. (I forget the record player model but it’s just one of the mid-tier Sony ones.) I also have some Sennheiser HD 598 headphones that I love the sound on. They’re open back so not appropriate for travel but if I’m alone and not in the living room, that’s my go-to.
I really like Sennheisers and I eventually splurged on a pair of 4.50 SE over the ear ones for travel. They have noise cancellation and a closed back so they work great on flights or trains. I like them a lot.
I also have some Beats Fit Pros that I use a lot. Most earbuds don’t stay in my ears very well so the little nubbin hook on the Beats Fits is really what prompted that decision but the audio quality ended up being perfectly fine for the form factor. Sometimes, you’re exercising or just listening to a podcast or a work call. They ended up being a good purchase.
NAS -> ALAC, high-res files -> Plexamp -> Sound Blaster's recently top-end sound card (name?) -> Schiit Heretic amplifier -> Sony MDR-1ADAC headphones
Or
NAS -> ALAC, high-res files -> Nvidia Shield (via Plex) -> Yamaha RX-A8A receiver -> Polk Monitor 70 tower speakers
At home mainly records. Rega P6 as a player, marantz amp and totem speakers or koss esp/95x headphones.
On the go Qobuz on my phone to cayin ru7 dac and campfire Andromeda iems.
I buy it if I can find it on a platform where the money is actually going to the musician. Then, I upload it in CD quality FLAC format to FunkWhale, and also add it to the SD card in my DAC (a Shanling Q1). Where it's convenient I listen on the DAC, where it's not I stream through FunkWhale.
Not sure if I merit being called an audiophile, but...
Huge collection of mp3s ripped from CDs. Stored locally, currently using a Unihertz Jelly Star as a glorified digital audio player, running BlackPlayer EX, which I like for it being a good mix of minimalist and giving me freedom to customize as much as I feel I need. When I'm using headphones, I want my ears uncovered so I use Shokz bone conduction headphones.
Very loud.
Car mostly now. 2.5” Pioneer dash speakers, 6.5” Polks and 6.5” Kenwoods, 10” Pioneer sub and monoblock amp. About a million times better than any upgraded audio system in a new car. Crystal clear audio, very tight controlled bass. It’s sublime.
Otherwise in the house from Apple Music Lossless through the Sonos Arc+sub gen 3+ surrounds and HomePod minis, very rarely through the home theater Atmos syste (Yamaha TSR-700 and Onkyo fronts and sub, and Niles in ceiling surrounds).
I’m a firm believer in not wasting money on expensive amps and gear for marginal gains (pardon the pun). I went to school for audio engineering and have mixed on $100K speakers. They sounded phenomenal but I have more fun in my car with its ~$600 system than anywhere else. Audio is very psychoacoustic. When you’re groovin’ the system almost doesn’t matter.
I've got speakers for every occassion. Several in-ears, over ears, monitor phones, Bluetooth speakers, and main amp and stack. Because of this it all sits in that top of middle range to bottom of high range, else I'd be broke.
Mainly use Spotify and vinyl.
Talk about chalk and cheese…
Well, I don't bother with lossless anymore outside of my own production. Not everything in the house is hooked up to EQs and I'm not hauling them out everytime I play music. So it's almost pointless these days.
TIDAL, K3/K7 (the K7 isn't portable), Sennheiser HD600s, and a pair of Hifiman HE1000s that I just bought. Both DACs work on all of my devices.
Openback neutral headphones. Listen to music the way it was mixed. Obviously higher bitrate is better, but I cave in to the convenience of streaming and am content with minimum 320kbps for casual listening. Definitely lossless for critical listening.
Plex -> Android -> Synfonium (use internal decoder) -> Meizu Hii+ DAC -> IEMs
I lose some information because of the Android resampler however most of my library is 16/44.1 flac. Although my collection of 24/96.2 is growing.
Buy albums on Bandcamp, Stream from Tidal, get a USB DAC + either vintage amp/speakers (almost anything pre 80 is good) or modern amplified speakers.
Bluetooth Xiaomi headphones because convenience is king (and I can't afford to pay more than $200 for audio equipment lol)
in silence.
Dynaco ST-70 (stereo tube amp, mine is maybe 1960s?), 8Ω tap to either Klipsch Heresy II or Vandersteen 1c speakers.
I've had the Klipschs for 20+ years, so to me they're sort of reference/completely neutral speakers. (I know Klipschs aren't everyone's cup of tea though.)
PC (MPD with Ario frontend) -> SMSL DO100 -> Rotel A11 Tribute -> KEF Q150. I'm upgrading to KEF LS50 Metas next week, can't wait.
The best quality that is convenient.
On the go? Bluetooth headphones from Spotify.
At my desk? Open back sennheisers from the FLAC from the NAS, or Spotify.
Any sufficiently high quality audio stream from my Plex or Tidal, always set to max volume in app/OS settings -> Topping D30 -> JDS Atom -> Sennheiser HD6XX.
Good enough for me.
Not an audiophile, so bexcuse the ignorance, but what is the logic of max volume in app?
The goal is to send the exact, unmolested digital samples from the file out to the DAC, which then sends its analog signal to the amp where you worry about how much to amplify that signal for listening.
When you set everything to 100% volume in software, you can assume that there is no software doing anything to alter the digital signal before sending it to the DAC (scales each sample by 1.0). But when you're under 100% volume in software, it assumes you don't have any analog control over the volume, so it needs to step in and alter the digital signal so that it shows up quieter to the DAC (ex. scaling each sample by 0.25). Depending on how that's implemented, it can result in losing resolution and thus quality of the signal.
I think this mattered more on older software that's more likely to use a smaller bit depth, but bugs happen, so why risk it and spend those extra cycles on a process that can only result in a worse signal, right?
There’s some confusing stuff in this response so before I get into the weeds, for all the people reading out there: you don’t lose quality by using your operating systems volume control.
Okay, with that out of the way, let’s say you wanted to adjust the volume of a digital stream that’s composed of samples. Each sample represents the original analog signals voltage at that slice of time when it was encoded. The number of slices per second is the sample rate, expressed in kilohertz and the voltage of the original signal is converted to a number, which is stored as a binary value whose length is expressed in bits, each of which can be either a one or zero and is referred to as the streams bit depth.
So you could have a stream whose sample rate is 44.1khz for example and that would mean that it was sampled 44,100 times per second. That same stream might have a bit depth of 16, and that would mean that the original signals voltage level was divided into 65,536 possible values. Depending on some other factors, that stream might just be cdda (a compact discs digitally encoded song information).
Now let’s say you had a computer that was handling that stream and was asked to reduce the volume of the stream by half by a user who can only stand to listen to it at that volume.
One way to do that job would be to decode the stream back into an analog voltage, attenuate it, recode it and then send it on its merry way. That would incur a decoding operation, require routing of that signal to either dedicated hardware to perform the attenuation and send the signal back and an encoding operation to make that now half as loud signal back into a digital stream that can then be sent wherever it’s destined.
Another way of handling that operation is simply dividing every slice of the streams 16 bit component by two, something that computers are very good at doing quickly.
It should come as no surprise then that the latter process is generally how it’s done.
But does that reduce quality or result in worse signal? That’s the question, right?
Well, any variation of one bit or less could be essentially deleted. A person could say “ah hah! The signal has been degraded!” And they’d be technically correct, but it wouldn’t matter.
In our example, the computer whose hands are all over our precious data stream is sending that adulterated information to a dac, which true to its moniker will convert the signal from a digital stream into an analog signal. That analog signal will then be sent to an amplifier with an analog volume control and from there to a set of speakers.
The amplifiers analog volume control is a resistor in the shape of a 3/4 arc with a wiper that can move back and forth across it, allowing anything put in one side to be resisted (or in the case of our ac signal, impeded) a varying amount depending on the users selected position of the knob attached to the wiper.
Okay but what is resisting a signal though? Well, a resistor will reduce the voltage between its two ends proportional to its resistance, measured in ohms. More ohms means more resistance.
For the purposes of our example, let’s assume the user has chosen an amplifier and dac combination such that the amplifiers volume control at minimum setting applies the minimum resistance necessary to completely attenuate the dacs maximum output and is not applying any resistance at its maximum setting. In other words, that it all works as expected and is perfect.
In this case, what’s the difference between sending a stream with data corresponding to a .5V signal that gets amplified as opposed to a stream with data corresponding to a 1V signal that goes through a resistor to bring it down to .5V before being amplified?
nothing
In fact, the digitally attenuated stream will probably sound better (closer to the original) because it’s not subject to the bourns/alpha ppm lottery!
Now.
Don’t let this stop you from listening to music however you like. My ass is itt admitting to using 40 year old record players to make sounds to cook to. But don’t worry about the computers volume control.
Yes, if everything aligns perfectly, there is no impact. The bit shift would be when you set the volume to exactly half, but that's probably not going to be the case. The app volume control alters the signal slightly, multiplied by the OS altering it slightly, which has a virtual certainty of introducing a floating point rounding error on every single sample, so now the ratios between your samples is ever so slightly different. And for what reason? What did that operation gain you?
And no you're not going to hear a difference, but the point of being an audiophile is less about hearing a difference, and more about good quality preservation practices.
okay, lets consider the worst possible rounding error in a 16 bit division operation:
i'll divide the level of one sample by some number that will not divide evenly, lets say three, and consider the impact of the rounding error. for the purposes of making it so I know there will be a rounding error i'll choose a number that three has a really bad time dividing into, say 65536.
65536/3=21845.3 with the .3 component repeating. perfect, that's exactly horrible!
so if we were to just do the simplest rounding possible to fit that into a 16 bit integer, the decimal component is dropped, rounding down to 21845.
but what is the significance of that error? a samples volume level that's one integer value off introduces a .003% error in level, but this isn't supposed to be 21846, it's supposed to be 21845.3. so the error that's introduced is .001%!
that's a pretty tiny error, but what if it was periodic and consistent enough to produce a harmonic component? that'd create harmonic distortion!
lets say there's a periodic and consistent rounding error that has a frequency of 1000Hz. so every thousandth of a second, the rounding caused by the volume being set at 1/3 causes a rounding error and makes a sample off by .001%. such a repeating error would introduce a harmonic component into the signal that the dac produces and be measurable as harmonic distortion at 1000Hz!
but how measurable? well if, for example, the harmonic component of the signal introduced was at it's absolute worst, and oscillated between a positive going error and a negative going error, it could introduce a peak at 1000hz of...
.002% of your dacs DBFS. so far below the noise floor it's immeasurable.
even if you had two software volume controls set at 1/3 daisy chained together doubling that error it would be immeasurable. although if we picked a computer software package to use instead of a bunch of hypothetical worst cases the total volume of a signal would be summed and then applied once in order to minimize just this problem. the people writing that software are pretty smart and doing that saves their program a step!
but measurability or audibility isn't the point, as you said. the point is to reproduce the sound as accurately as possible! so it really doesn't matter how tiny the effect of rounding errors due to prime denominator volume settings is if it's larger than the effect of the analog volume control that whatever signal the dac manages to reconstruct from our mangled stream is put through. we're trying to adjust the volume down to a comfortable listening level, after all.
so how bad is the volume control? well, if i were to go to mouser and look at the potentiometers section, i could choose one with a tolerance as low as... .5%! and that's a three-gang model that costs $50!
but what if i used a precision potentiometer? why, there's a .15% tolerance part that's available for the very reasonable price of $825!
okay the precisions are out of my price range, but those $50 ones could work. tolerance is just a number anyway, right? we want linearity! what does the datasheet say about linearity... 2%!
that's not even considering the amplifier design's distortion. lets assume it's perfect.
so just passing the signal from the dac through the amps volume control causes possibly 200 times more error and distortion than adjusting the volume control in the computer.
i get the pursuit of the best possible reproduction, but the computer volumes got those cermet pots beat hands down.
Interesting point. What about the case where you have your digital volume set to 1%? Would this not squeeze the samples into 1/100 the dynamic range? If I set my volume to 1% it seems to me like those samples now have to all exist within the bottom 1% of the 16b range. Do you not lose at least 5-6 bits of precision on your signal doing this?
You don’t lose precision when you lower the volume (in either an analog or digital realm). You lose actual information!
Let’s say you have a recording you can only listen to with your volume at the 1% setting. Analog or digital, it doesn’t matter.
Your whole system has an acoustic noise floor at something like idk, 10 acoustic decibels. That is a really charitable number because I’ve never measured one that low and it directly corresponds to the loudness of another healthy persons breathing at rest. To give you an idea of how quiet that is, the acoustic decibel scale generally puts a ticking mechanical watch at twice as loud (20 decibels).
I don’t want to talk about decibels because I don’t want to explain the math in the detail I’ve been giving these posts, but we gotta at least cover a little:
Decibels are the measure of sound energy, their scale is logarithmic, so the base of the log function determines how many of decibels make for twice as much.
There are different decibels for measuring in different mediums with different references and they even use different logarithm bases.
Acoustic decibels are log10, so that 20 decibel ticking wristwatch is twice as loud as a person breathing and half as loud as whatever the workplace safety scale says 30 decibels is equivalent to.
Okay so now that we have a floor, we need to establish a ceiling. Let’s say that you did everything right and hooked all your stuff up, turned the volume on the amplifier all the way down, put your headphones, played a maximum volume test tone, maxed out the volume on the software, then turned the amplifiers volume control up until it caused you immediate physical pain. If you have really good hearing, that’s 115 acoustic decibels. Let’s say you got to 120 with the amplifiers volume control up all the way.
Okay, so the noise floor of your headphones on your head is 2^11 as quiet as the loudest sound you can tolerate hearing.
Now you set the volume control to 1%. Doesn’t matter which one. Everything gets 99% quieter. The parts of the signal that were 120 decibels before are now 1.2 decibels. They have been divided by 100, and it’s possible that rounding errors have added .006% error to their harmonic content. .006% of 1.2 is .0072 decibels. Not only is the loudest sound you can stand to hear now quieter than a person breathing, it’s below the noise floor of your system. Far, far below the noise floor. And any rounding error from dividing by 100 is as well!
Okay but what happens when you’re listening to music though? Let’s put aside all that hypothetical stuff and get rockin! Instead of talking about test signals and boring crap, let’s talk about a song!
So same established setup from before, but now you’re listening to a recording of someone playing the banjo while rocking in a chair. There’s a lot of different harmonic content in this signal, the birds chirping, the persons breathing, the wind, the chair creaking the boards of the porch and of course, the instrument itself!
All these different things are at different volumes and they represent components of the harmonic content of the signal you’re listening to. When you turn the recording down, you’re attenuating the signal. All of the signal. If you apply enough attenuation through your chosen volume control to lower the level of the banjo by 40 acoustic decibels then all the other components of the signal are lowered by 40 decibels too. If they were previously 50 acoustic decibels through your headset, they’re part of the noise floor.
The quietest information is simply lost.
Edit: there are massive amounts of information that have been simplified so much as to make this post incredibly inaccurate. Please do not use this as a reference for understanding how we measure or interact with sound. I’m sorry for not going into greater detail but it’s too early to explain the relationship and history of acoustic decibels and decibels per volt.
I'm sorry you have to type so much, I am familiar with most of it, but I appreciate your effort to make sure we're on the same page without being a douche about it lol. It sounds like we're saying similar things, but I don't understand why lower precision is different from losing information. To me, that's the same thing, it's a lossy operation.
So the thing is, I have a pair of desktop speakers without any physical volume control that I primarily use for convenience. And for whatever reason, a comfortable listening volume with them is between 1-8% in the OS volume control. I guess the internal amp is just hardwired to be way too loud?
Anyway, I assume that this setup is resulting in objectively lower quality output than if I were to have a 100% signal going to a decent quality DAC/amp with analog volume outputting to the same speakers. And not in a "technically" kind of way, but in a very real "we just crushed the signal into 1/25th of its original scale" way. Would you agree? Am I mistaken?
no worries. i've been enjoying going back through this. you're basically me 25 years ago emailing the winamp ppl to find out what volume control i should use to turn down (for what!).
so there's a misconception here between compressing a signal and attenuating it. imagine you are looking at a frequency spectrum chart of some song instead of listening to it. it's got some loud sounds, which show up as big peaks on the chart, and some quiet sounds which show up as small peaks on the chart and there's a noise floor which is the stuff in the chart that's not a peak at all.
if you plug a potentiometer in between your signal and the spectrum analyzer and turn down the volume, youll see all the peaks, loud and small, be reduced in amplitude by the same amount. this is called attenuation. the quieter sounds could be reduced until they are part of the noise floor and become imperceptible while the louder sounds would still show up.
it doesnt matter if you achieve attenuation by dividing the 16 bit level component of a stream of samples or by using a resistor as a voltage divider. the quiet and loud sounds are affected equally. those two ways of achieving attenuation function the same because they are performing the same operation.
now lets say you plug a rack mount compressor effects module in between your signal and your spectrum analyzer instead. you could apply more compression to the signal and achieve exactly what you describe, a smaller distance between the quiet and loud sounds, reduction of the original scale, removal of dynamic range, effective bit depth reduction! it would be actual factual "we just crushed the signal into 1/25th of its original scale".
and if you used a module (or software package) with the capacity for it, you could tie the compression ratio to a gain control so that the compressors output got quieter when you turn the compression ratio up, resulting in more heavily compressed sounds at a quieter volume. that's a neat little mastering trick to make recordings sound "lively" and "intimate". makes all those pick scrapes and finger swishes stand out alongside the plucked strings.
you could also do the inverse, make all the quiet sounds louder, so that the guitar is as loud as the kick drums' transient and it would make your whole song sound much louder and stand out better against background noise in a difficult listening environment like a car radio or cell phone inside a solo cup.
there are even modules that do the opposite, called... expanders! they do what you might expect, increase the dynamic range between loud and quiet sounds. a company called DBX made models for use in home stereos in between tape decks and the amplifier in order to reduce the noise floor of tapes.
but it's none of that is attenuation, the operation that your volume control provides.
and you're correct, both compression and attenuation are lossy operations no matter if they're done with analog electronics or by a microprocessor operating on a buffer somewhere in memory. the difference is that attenuation is literally required to prevent permanent hearing loss and possible equipment damage, while compression is not.
This is the part where I'm not following. In my head, if you're using analog hardware of sufficient quality, you can attenuate the signal to be very quiet, but still preserve it's dynamic range. In fact, the DAC is already outputting a very weak, but faithful analog reproduction of the signal, and an amp with a decent S/N ratio is able to bring that very weak signal up to a listening volume without introducing enough noise to matter.
Hypothetically, if for some reason, you took the signal post-amp, used a pot to attenuate it again down to the energy of the post-DAC level, and again ran it through another amp you would theoretically have the same signal still (I understand that in the real world we would start amplifying noise and the signal would degrade, but stick with me). Nothing about the process necessarily introduces noise and thus destroys the signal, you're only limited to the quality of the components at that point. If you had an infinite chain of theoretically perfect amps and pots, you could repeatedly attenuate and amplify the signal forever without ever losing any quality. It's an analog process that theoretically preserves the signal, +/- some amount of error due to physics.
Meanwhile, 16b is 16b. If you start shrinking all samples relative to each other (ex. down to 1/64 the original volume, or 10b of resolution), different values inevitably have to clamp to the same values (fitting 64k values into 1024 values), losing information and resulting in poorer quality. If you then try to send that 10b signal through a DAC/amp to achieve the same listening volume that you would have had before digital attenuation, it's just a 10b signal bit shifted up. All your LSBs are 0s. You can't possibly attenuate digitally, and then amplify it in any way and hope to get the same signal back. It's a discrete math process which destroys the signal by design.
Would you agree?
the effect of attenuation is the loss of intensity of signal.
loss. it goes away.
attenuation is a lossy process. information in the signal is literally absorbed and radiated away as heat. it cannot be reconstructed because it's gone.
it isn't an analog process that theoretically preserves the signal, it's an analog process that explicitly destroys a component of the signal.
but what if it wasn't...
okay, lets assume for a second that you have a signal with the same harmonic content as one of my previous examples, a high peak when viewed on a frequency spectrum chart, a low peak when viewed on that chart and everything else. these three parts of the signal represent the loud, quiet and "silent" parts of the signal respectively. unlike the previous example we'll let our noise floor for the silent parts be infinitely low. for now. so you start hooking up your perfect amps and pots in line and setting them all to 1% or so and listening. it's sounding pretty good at first, but once you get a few deep, you start getting white noise and clicks and pops and all kinds of craziness.
what the hell! all this equipment is theoretically perfect, why is there noise? it can't be coming from the perfect equipment!
it's not. it's coming from the medium. in our theoretical example all these amplifiers and pots are hooked up with conductive wire. the signal has to propagate through that wire from component to component. atoms of copper are being excited and losing their excitation in proportion to the signal. their state of excitation is being amplified over and over again. the noise is in the wire. by amplifying it over and over again you made it audible. you can't ever escape it. signed, listening to noise gang. come to my modular synth show.
okay, so now that the possibility of ever attenuating a signal without losing information is hopefully put to rest, lets turn to the digital attenuation of the signal in comparison.
level attenuation over the digital domain is also a lossy process. what's being misunderstood here is that the levels aren't being shrunk relative to each other, they're each being divided and the signal that's reconstructed by the DAC no longer contains the quiet parts.
just like those quiet parts were absorbed and radiated as heat by the resistor, the digital version of attenuation does away with the need for all that physics crap and simply deletes them from the stream.
if the levels were being shrunk relative to each other, you'd be compressing the signal like when you use the bitcrusher pedal for your guitar and there would be lots of harmonic distortion. but attenuation and compression are different processes and have significantly different results.
consider a quiet sound, your 1/64th volume signal. a sine wave. its encoded to represent 1/64th of the maximum level of the adc's input because when it was recorded, it represented 1/64th the maximum level of the preamp/microphone/whatever that was plugged into the adc.
is the quiet sine wave of lower quality than one that's using the full bit depth of the adcs output because it's intended to represent the maximum level that the adcs input saw from the preamp/microphone/whatever?
of course it isn't. it just wasn't loud.
and if your loud sine wave was electrically generated by a theoretical perfect function generator which contains no distortion or other sonic content before being sent to the adc, would it be more damaged if it's amplitude were divided by 64 before being decoded or if it were decoded and sent through a resistor whose value was chosen specifically to dissipate 63/64ths of it as heat in order to make it as quiet as the quiet sine wave?
of course it wouldn't.
to your last question, let me rephrase it into something I can agree with: you cannot possibly attenuate and then amplify in any way and hope to get the same signal back. It’s a lossy process which destroys the signal by design.
Pixel 6, Apple dongle and Truthear Hexa in the streets, Shiit Magni+Modi and Hifiman Sundara in the sheets.