As we are running towards enshittification with speed. I’m here today wondering. Where can I get highest quality music/soundtracks? Is flac highest quality? What is Dolby, and is it just an EQ? Is 192kbit enough or where is 384?
How do I go from digital to my ears without loosing so much on static? And what’s the quality on Vinyl?
Other than physical, what shops do you recommend for digital music. I know beatport.
What streaming services do you recommend to the others? Is Tidal better?
And how can I do all this on Linux?
And yes you can hear the difference, stop neglecting.
Usually you can get it off of CDs, that’s the best media to get some high quality music. It’s pretty hard to find music that’s higher quality because it has a niche audience. Not even streaming apps that have contracts with record lables have access to masters in higher quality than 44.1KHz 192kbit/s. It’s up to the label to release anything higher quality or not.
Don’t have a clue. Music to me is stereo with spatial perception given by the mastering process and acusting characteristic of the transductor that reproducing the sounds.
FLAC is just an efficient and open format to pack music in very high quality. Can be compressed though. Uncompressed music can come in all sorts of “containers”, just like video does.
What you can do to reduce it to a minimum on your side is checking if your audio chain is as clean as possible and relying on bit perfect playback. Which I achieve using Foobar2000 and the FiiO ASIO drivers for my DAC + some more configurations I have on a document somewhere. The point of it is bypassing Windows and playing straight through the DAC making sure it matches the bitrate and sampling frequency of the source, which is something Windows doesn’t do by default. In doing so the control is handed over completely to the audio player and you can’t listen to anything else but music until you close it.
The absolute highest quality is anything that’s digitally identical to the original that came from the mastering engineer. This includes WAV or FLAC files because the L in FLAC is for lossless.
Practically though, high bitrate lossy files are indistinguishable from the originals. ABX testing has proven all the big codecs all the way back to MP3 to be capable of producing transparent encodes provided sufficient bitrate and that the encoder wasn’t crap.
Dolby is a company. They’ve created a series of audio standards and both analog and digital codecs. Their early stuff was analog noise reduction for tape but these days they’re most well known for their multi-channel digital codecs (AC3, EAC, Dolby Digital, Atmos, and variants).
A bitrate of 192k is sufficient for transparently encoding two channel audio with all the major codecs. Any improvement offered by higher bitrates outside particularly difficult to encode source material is negligible.
For the best digital playback try to avoid sample rate conversion and use a quality DAC. After that we’re back in the analog domain so this where all the traditional audiophile considerations come into play.
All you really need is to match your playback device (which is hopefully a decent DAC) to your source material. DACs are oblivious to bitrate. If you listen to two channel music this is typically 44.1kHz and videos or pretty much everything else is almost always 48kHz.
The benefit of a player that can take exclusive control over the playback device is that it can change the sample rate of the playback device to match your source material on the fly. So if you have a mixed library with some 44.1kHz files and some 48hHz or 96, 192, or whatever it’ll always try to avoid sample rate conversion by matching your source. Personally I find the mode switching far more annoying than even the crappy sample rate converter windows uses.
The Windows converter, I would guess, is always the issue when the mic sound starts popping/crackling.
I don’t understand how audio is so problematic. I should be always getting max output quality, no matter how bad the input is. Higher bitrate includes the smaller ones. Like it’s just that there is more bandwidth for the audio. And with the audio chipsets should be able to determine what the input format is, then it should decide what that piece of sound should be like?
I don’t want to go into too much detail, but the reality is that working around the constraints of how audio hardware works makes this impossible. Software has to worry about samplerate and bitdepth conversion to ensure that the actual PCM audio format matches whatever mode the soundcard is currently processing audio in. There’s a lot of magic that goes in in the background in Windows and with PulseAudio (or whatever soundsystem you’re using on Linux), which is essential to making audio usable.
It’s difficult to tell if you’re having a specific issue (mic popping?) or whether you’re just looking for general advice.
As others said, FLAC will give you the highest quality because it is “lossless” (it decodes to the same as uncompressed CD audio / PCM). People often forget that there’s both 16-bit and 24-bit FLAC, with the latter being higher quality (at least in theory). Samplerates higher than 48,000 Hz don’t offer any benefit to quality in a consumer setting, when you’re listening to finished music.
Another “pro tip” is to avoid using Bluetooth - don’t use Bluetooth headphones because they compress the audio. If you’re listening to FLAC over Bluetooth, it defeats the purpose because you’re just listening to lossy compressed audio again, although the quality will still be higher than listening to MP3/AAC (since it would avoid double compression).
Compressed audio has basically reached peak fidelity, with AAC and Opus being excellent codecs that offer phenomenal quality at relatively low bitrates (256k+). FLAC is the best, but it comes at higher bitrates (though it is variable).
Newer developments like Dolby Atmos are essentially new channel arrangements (instead of stereo), that allow for coding of audio to fit arbitrary speaker arrangements (eg. 25 speaker setup), encoding height information (see Ambisonics: Ambisonics - Wikipedia), or individual sound sources (See DTS:X, kinda like how games do it). These new developments are really all for movies and I don’t think we’ll see any of these adopted for music, at least not for a long time.
Others in this thread already did a good job explaining everything but I hope this fills in a few gaps!
Sample rate conversion won’t introduce pops or cracks. Those are buffer underruns. Artifacts introduced by sample rate conversion are usually described as smearing, preringing or aliasing but those are extremely subtle.
Audio is problematic because you want to keeps the time between when sounds are generated and heard as small as possible so buffers are kept as small as possible but that’s what leads to buffer underruns and the associated crackling, clicks and pops.
It shouldn’t, but if you have a particularly crappy chipset implementation I can see how ports from the chipset could behave differently than those using lanes directly from the CPU. It probably wouldn’t be the first place I’d look for a problem though.
If you’re looking to stop pops, and crackling on your mic you want to increase the size of the buffer on your mic input.
Even a large buffer is typically only a fraction of a second though. If you’re experiencing “long delays” to “peers” that sounds like a networking issue that is probably being caused by something outside voicemeeter.
Sure there are a lot of shiddy streaming services now, but I’m also finding it easier to get lossless and DRM-free formats from my favorite artists. The smaller labels are more willing to do those things, of course. I’m hoping the larger ones catch on, but don’t hold your breath.
I think DSD is the highest quality, but as others have pointed out most people can’t tell the difference between lossless and lossy codecs when the bitrate and encoder are sufficient.
Vinyl is… it varies. How good is your turntable, your cartridge, your pre-amp, is the tonearm properly balanced? etc. You also have to keep the records super clean, which seems to take more time than anything else. Plus the stylus is physically wearing down the vinyl just a little bit every time you play it, so IMHO digital offers the more consistent experience. For me, vinyl is fun for the ritual and digital is nice for the convenience.
On that last one. I am not a musician, but a search landed me there and that forum and those folks have been incredibly helpful with some of my system tuning and hardware errata.
Format wise, it has a chance to be good. Does not mean the mastering engineer/guy with a mixer was awake and sober while doing their thing
Dolby is a company that makes software/encoding schemes for multi-channel and formerly noise-supression/reduction on tapes.
EQ is changing frequency-response of music, can be done in software or hardware.
Depends on the format.
60dB SNR (Vinyl) is ok, 80dB is good, >100dB is marketing wank
Get yourself a mid-range (80 to 200 $/€) DAC from any company, see if it sounds better to you than whatever you were using before.
Has to be mastered with more care, else the mechanism does not work right and the tracks skip (needle jumping the groove). Aside from that, signal to noise (SNR) is kinda bad. Has the appeal of listening to albums in one sitting, instead of jumping tracks all the time.
Have your stuff local, makes things easier.
Mostly plug and play, find yourself a music-player/organizer that you like (Strawberry player is my preference)
That was an issue I was dealing with among other users 2 years ago. Then I just stopped using voicemeeter and Windows. So the issue is no more.
It was Voicemeeter. I forgot the setup that I had, but it was a noticeable delay for me.
Bought one in 2019 or 2020. It has a noticable difference to what setup I had.
I also got the Meze 99 Noir, which also added so much to the listening experience.
Been working on that. But I’m a person who likes radio and playlists/mixes a lot. DJs usually don’t sell those, only available on Youtube most of the time.
Seems that video is more convoluted in HQ than music.
About a year ago I got a 128GB SD card so I could put my entire library of 12,000 mp3 songs on my phone - to play thru Bluetooth in my car stereo. I have a Pioneer head unit with nice JL Audio speakers. 12,000 songs is too much to manage. Most of it are songs I don’t know why they are in there. It’s like a band had a ‘One Hit Wonder’ and I somehow ended up with 100 files of every song they ever recorded + live bootlegs + all the dance club remixes.
Then I tried Spotify… until it infuriated me with terrible “Suggested for you…” choices. I made a 500 song playlist and Spotify won’t use it. I spent more time hitting the ‘Next Track’ button on the car stereo, than actual time I listened to a song that I liked.
So this weekend I bit the bullet. I spent 6 hours editing 12,000 files of bulk random music, down to 3,000 good songs. The other benefit is . . .
jetAudio+ Hi-Res Music Player for Android.
Paid version - The free app has ads.
The audio quality of JetAudio is way better than the quality of Spotify or the Google audio player. I noticed it Immediately. I had been listening to crappy Spotify audio for a year and got used to it. Pure MP3 going thru some HQ audio special sauce is making me happy.
Google and Spotify players just decode audio files as is like any other player. JetAudio adds “enhancements” which although you may find pleasing are not faithful reproductions of the source material. Software that plays back audio files can’t make them more “Hi-Res”. That’s just marketing.
Genuine efforts towards the highest quality playback seek to avoid even the most subtle alterations of the source material.
