Why are there no good music players for Linux?
Look at Winamp. You have all the tools you need to manage even the largest music library in those panels. It takes only a few seconds to build a playlist out of exactly what you want.
On Linux, there's nothing like it. Either it just plays sound files without library management, or the library management is embedded in subpages or is otherwise tough to find.
Have you even tried any players in Linux? Even the default player in Ubuntu has pretty much all that. Clementine has MORE. Quod Libet is what I use.
That is the ugliest GUI I have ever witnessed.
>Storing music offline in the era of 2020
LMAO autist
go use VLC, autist LMAO
>Look at Winamp.
yeah, and if you weren't underage you would know about XMMS.
Clementine is great OP, give it a shot
Audio in general is a trainwreck on Linux.
>Look at Winamp.
Take your meds, grandpa. Use Spotify or youtube red
Got you covered brah
>yeah, and if you weren't underage you would know about XMMS.
XMMS looks like Winamp but lacks any library management features.
No it isn't with oss4 (or wasn't with the oss3).
>XMMS
no working library function
Says who? Works great for me. I have alsa, jackd, and pulseaudio all running fine simultaneously, and
>mpd+ncmpcpp
>audacious
Yes, because 'latency' is the only thing that matters.
Judging from the use of audio server like jackd, you're some kind of audio 'producer', yet you're using sound system that absolutely stinks and can't produce decent output quality. If you're using alsa to output and listen to your work, you're producing garbage.
Deadbeef is fine.
Something more compilcated use beatbox or clementine
>Linux/Unix audio is so marginal business that there is room only for one idea at time. This idea has remained the same for past 15 years. This magic subject is “low latencies”. In particular this is the favourite subject of many
>ALSA believers because ALSA is said to provide superior latencies. IMHO the whole low latency hoopla is pure bullshit.
>It looks like very few Linux/Unix audio hackers understand what do low latencies really mean. Having “zero” latency might look much better (or sexier) than latencies of ~1 ms. ~1 ms in turn looks better than ~10 ms. However they forget that speed of sound is limited to ~340 m/s (at sea level). This means that:
>-During 10 ms sound travels 3.4 m.
>-During 1 ms it travels 34 cm.
>-During a 48 kHz sample period it travels mighty 7 mm.
>In the other way this means that:
>-3.4 m distance between the sound source and the microphone (or speaker and the listener) causes a delay of 10 ms.
>-34 cm distance causes a delay of 1 ms.
>-7 mm distance causes a delay of 1 sample at 48 kHz.
>And also:
>-Most sound sources (musical instruments) are much larger than 34 and it is inpractical to place the microphone closer to them than 32 cm. This means that in the real (analog) world there are always sound propabation delays that are in ~1 ms range.
>-Distance between front and back rows of a big symphony orchestra or choir is several meters so the differences in the delays are in the ~10 ms range.
>Do this kind of delay matter in the real world? Do recording engineers pay any attention in getting the microphones located within few centtimeters from the instrument. Do you have heard anybody complaining that a symphony recording has annoying delays between instruments? No, you don’t. This kind of latencies simply don’t matter in the real world.
>So why all this low latecy bullshit in audio programming? Why an application with “zero latencies” is considered to be superior to an application that does “low enough” latencies”? Getting reasonable latencies (down to ~10 ms) is trivial. You don’t need to use any special techniques to get there. Maybe the problem is that this is not sexy enough.
>There are many situations where the latencies need to be kept in precise control. However in these situations there is no need to push the application or the device to work in some special hyper low latency mode. Instead very simple math can be used to correlate different streams with each other. This is not something that requires clever programming but just some elementary school math. When two or more audio streams (locked to the same sample clock) are started a the same moment (sample) then they will get counting at the same rate forever. This means that sample N will always belong to the same moment of time on all the streams.
>There is also one special situation when the driver/device level latencies matter. This is when the latencies can get acumulated in way or other. However (IMHO) this can only happen if the overall application design is busted.
>So next time you hear/see somebody talking about lowest possible latencies then you know this guy is potentionally a clueless idiot. In particular if he is blaming lack of super duper real time extensions in the kernel for the hiccup caused by the application.
>Remember that for any level of latency the application must be able to do processing for every single latency period withing the previous latency period. If the latency is 100 ms this is pretty much guaranteed to work in all cases (even if you do some heavy computing in background). However if the latency is 1 ms then the operating system must be able to reschedule the application to run once during every single 1 ms period. Any time the time window gets missed there will be some kind of hiccup in the audio. This may be possible by using nasty real time kernel hacks at least if you can tolerate occasional clicks in audio.
This is from Hannu Savolainen's blog, the main developer of Open Sound System, the developer of the standard unix sound system, once the in-kernel sound system of linux too.
Clementine is very good and will satisfy most foobar2000 and Winamp autists.
Getting down to 10 ms may be trivial, but Windows 7 doesn't do it unless you replace the stock audio stack with a third-party one.
Whether it matters or not, it's still a technical achievement, and personally I can hear a 20ms delay and it's annoying.
The winamp for linux is called cmus.
No config needed, everything works right away.
Load your library: ":/path/to/music"
Search for music: "/search_term"
Add to queue: "e"
General playback is "zxcvb" (ordered like a tape recorder, just like winamp)
You switch view using the number keys.
If you want to make it slightly better, you can improve the controls through cmus-remote scripts and a drop-down terminal.
Clementine
Agreed
Download MOC
its lightweight, barely any CPU usage and looks simple
winamp 1 is class though
Deadbeef. It's just foobar2000 but on Linux.
Using clementine on windows here
it just werks
The latency describes the reaction time from when you start a broadcast to the time the music starts playing.
You fill up a buffer and play that, so you won't be able to hear a 2s latency if you listen to music, but nobody wants a machine that is that slow.
People praise a good audio interface if it can do one of two things:
Give full control of what is being played and do it with as easy an interface as possible.
Depending on what is most important, people choose different interfaces.
i have a vm for Windows, so i could use iTunes on my Desktop (FreeBSD).
clementine
gmusicbrowser
audacity
quodlibet
moc works for me
Ncmpcpp
...
why not cmus? it's literally perfect:
>every imaginable feature
>vim keybindings
>ncurses
>gnu licensed
I don't have much storage on my laptop so I store most of my music on an external drive. Is there any program that I can create playlists and it'll scan both my laptop and external drive and export it so I can transfer it to my dap?
For something like foobar there's deadbeef, for something like winamp there's audacious (it can even use winamp skins) and for managing big libraries quodlibet is great.
freetards actually think their console apps are equivalent to what's offered on real operating systems
news flash buddy, it's current year and no one wants to learn keybinds just to play some fucking music
And it have a proper unix-style API, so you can properly tweak in it.
>I'm using cmus right now
yea cmus is really great.
Regardless of your opinion on terminal music players.
if you already knows Vim keybindings, you don't want to learn GUI bullshit anymore
GUI is inefficient. Why do I need disgusting GUI like OP's to listening music? Nonsense.
It's an early-2000's GUI. People thought it was cool back in the day.
It always looked like shit, back then people didn't mind because you could load a 100gb playlist in seconds while a song was playing and you could instantly search and queue songs up.
That was the power of that interface.
I hope we will soon get that 90's performance on modern music players, but we all know that all those webdevs are shit at programming and probably haven't heard music before anyway.
>wants to learn
If you can learn everything you need in 10 seconds, I would call it intuitive to use.
As for being a text interface, winamp was a text interface and that was fine.
You have to hack away if you are the kind of person who clicks at album art to find songs, but if you know the artist or song name, cmus is fine.
I get the album art in a notification when I use the media keys, it is enough graphical interface for me.
I want a music player to substitute my own radio channel.
The ideal radio channel plays music I like without any advertisement or interruptions (local music playback).
I can request songs I want to hear (queue functionality).
When I don't request songs I hear something else (go back to shuffle from entire library when queue is empty).
If it should be even better, it should automatically download the news from a radio station and then throw that in the playlist every hour, but I don't currently do that.
But nothing about this requires me to have a graphical interface.
This. I would even dare say that MOST people need nothing but what cmus has to offer.
Winamp is a fucking awful 2007 meme.
Foobar2k is the best Windows music player currently and, oh look, Rhythmbox is exactly the same
>2007
lol
Winamp is a 1999 god. It brought lightweight MP3 playback, a groundbreaking interface, and incredible visualizations to the 486DX.