Linux audio

Discuss linux audio
fucking poettering pulse

Other urls found in this thread:

overclock.net/a/how-to-get-the-best-sound-with-and-properly-configure-pulseaudio
freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Audiophile/
people.xiph.org/~xiphmont/demo/neil-young.html
4chan.org/g/
youtube.com/watch?v=8Jj3SKVvuoY
thehum.info/
xiph.org/video/
hydrogenaud.io/index.php/topic,107354.0.html
twitter.com/AnonBabble

in case any of you fags wanna try checking it yourself get package paman

here's a short guide:
overclock.net/a/how-to-get-the-best-sound-with-and-properly-configure-pulseaudio
freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Audiophile/

cat /etc/pulse/daemon.conf#show the default config ;/#=commentspulseaudio --dump-resample-methods#show sampling modes

If you have better configs that reduce latency or CPU load you're welcome to post them in here.

freebsd.org
openbsd.org

just uninstall it. alsa works fine without it

alsa works just fine, you should give up those distros that make hard deps on pulse, fuckin shit

For playback this is fine. Unless you have bat ears and the ultrasonar equipment to reproduce frequencies above 22kHz.

HAHAHAHAHAHAHAHAHHAHAHAHAHAHAHAHA!

whats wrong with linux audio faggot?

that's not the same.
what you're saying is the frequency range of human hearing

Audio sample rate is different. 20kHz exist but sounds bad (amateur sound recorder). 40-44.1k is the standard for normies that lack storage (compressed mp3/m4a). 96k 192k is the standard for studio master quality audio (lossless CD's and recording studio).

Hertz means 'per second' so it means samples of data per second. What you're saying is samples of oscillations per second measured in digital form (like pitch A=440 Hertz).


it's just the systemd programmer's defaults.


this.
I wanna see linux userbase crash right now while trueos/bsd systems overtake this fucking tainted linux botnet shit.

why the hell are you using pulseaudio? your gnome/kde/cinnamon depends on it? then switch distro or uninstall pulse. Install gentoo.

Isn't their apulse too for those few badly designed applications that require PulseAudio?

I switched it to 48000Hz because most of my music is in Opus which uses 48kHz. Is this correct? I have a USB DAC that supports 192kHz and I think 24bits but I think those are overkill

Everything from terrible driver support to a serious lack of good resource managing software to use your audio with. Linux isn't alone in this though. The only worthwhile audio is on Windows sadly. Nothing compares to Foobar2000.

I've always had the issue of it constantly fucking up and switching audio drivers, resetting the kHZ etc. It's just piss poor configuring when it comes to Linux. The constant battles between pulseaudio and alsa. Alsa is better but still shit. Let's not even talk about pulseaudio. That abomination should die along with systemd. The biggest issues I've had on Linux has been with audio.

however it can run on wine, at least on macOS

also, in 10 years there will be a new player, even better than foobar2000, for all unix-like systems. it'll be written in Rust and under GPLv3 license

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Let's discuss music players.
The best one I found so far is sayonara.

Fuck off retard. The point I made stands. No need to explain me thinks I already know.

If mediamonkey wasn't proprietary it would be great media player / databasing software

And why is it the best? What's the resource management like? Does it allow for DAC syncing? Does it allow control of the DAC in the software? Which frequencies does it support? Can it be synced with spotify or iTunes? You can't just say something is the best and then leave it at that.

I just use mpv on mac os x. Everything for mac and linux is shit when it comes to audio.

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fast, lightweight, GPL3

resource who? the human resources mgmt?
that sounds too proprietary
are we still talking about software here?
dropped. you should be on facebook.

So it's shit. You don't seem to understand, if you don't give people the ability to do things, people won't use it and it will fade into obscurity like most music players. You'll have more luck just using mplayer or mpv. I don't have spotify or itunes.

Calm down Stallman. It's just having the ability to control the DAC through the music software instead of having to use other software. You want it to link up with the DAC, as in select the device through the program and communicate with it directly without having to go through two programs. That is also resource management.

Frequencies... as in 44khz up to 192khz. Does it allow you to change the default of 44khz to others? So you recommend a music player without knowing what the fuck you're talking about. Nice.

Also let me guess. You don't even have a DAC and enjoy electric interference from playing directly from your motherboards soundcard? :^)

I never understood why so many people like Foobar2000. I've always preferred Winamp/Audacious.

Fucking retards like you not knowing basic shit like Nyquist–Shannon and still talking out of their asses are quite the problem around here.

people.xiph.org/~xiphmont/demo/neil-young.html
Educate yourself, moron.

Using cmus at the moment, not perfect but the "least worse" I found.

pig disgusting defaults
eat shit portereing
t. grorious alsa hardcoded 44.1 with linear resampling dmix
other sample rates DO NOT EXIST

I use it for the tools. It allows you to trans/encode between various formats.

Clementine

It's awful. Dmix up/down samples everything and produces crappy audio artifacts. The only alternative is to use the crippled system where only one thing can access the audio, making it a pain in the ass to watch youtube.
Linux has awful audio.

Why is pulseaudio such a resource hog?

You should file a bug, if you think there's a problem.

Music is mostly 44.1, what should you use?

You have absolutely no idea what you are talking about.

kys

I like Deadbeef. No library functions but that's fine. I use playlists.


Yes but can you get good resampling without high CPU usage?

Deadbeef is the only one Ive gotten to look like my foobar did on windows so its my favorite.
no pic because tor users cant upload pictures, but imagine an album art pane taking up half the screen with a tabbed playlists pane with controls/seek/etc above it on the other half.

Pretty sure you can. Just test using ffplay and sox.

lmfao yeah user there's nothing wrong with linux audio. It's not like every (now dead) audio community wasn't just pages of hardware issues and configs.

I have 88kHz -> 192kHz hi res audio with hi bitrates. Unless you only listen in headphones with RIPped music or not working in the music production world 44kHz is fine together with low bitrate.


there's noticeable latency when I try to playback my audio in higher speeds.

resampling is actually just as worse with windows and can even take up to 20% CPU @ 192kHz
if the user before is right, microsoft is plotting to tivoize linux and make their very own distro with their project: systemd .

m8, just convert it with sox. Here, target 44.1KHz and 16 bit depth.

resample(){ sox -q --buffer 8096 -G "$1" -c 2 -b 16 "$2" rate -v 44100 dither -a}

Whoops, use dither -s, actually.

This probably means I'm missing a codec or some shit, right?

Tivoization can only happen with hardware. Systemd is not a physical machine.

it can be something with your browser
try mpv + youtube-dl

Tried that and the audio works just fine, so it's probably an issue with Firefox. I'll install a different Void .iso with a dedicated DE like Mate or LXDE and see if that works any better.

Newest firefox REQUIRES pulseaudio. FF sucks.

It's doesn't, though.
t. Gentoo

not him but Void Linux doesn't ship Firefox 52 yet they do however also package icecat which is nice

this is why nobody takes Holla Forums seriously

Who the fuck cares you retarded applefag? It can reproduce audio above what any human can perceive. Fuck faggots like you, you're the reason audio didn't really make any progress in recent history.

You're retarded, everything you said is wrong and you should fucking kill yourself.

Did anyone here ever try to set up OSS4 and can give me some pointers? I have no idea how I'd go about this and heard there are compatibility problems.

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44.1khz is the standard due to the fact

Hi-Fi (96-192khz) is good because:

I would agree that 192khz/24bit sound cards sound WAY WAY better than 44.1khz/16bit only sound cards - both playing the same music rated at 44.1khz.

One way to try this out is android phones.
Almost all crapdroid phones out there come with 44.1khz and 20khz /16bit -only sound cards and some even use fucking 8-bit which sound terrible as fuck. EQ doesn't even work on these crapdroids.

You would notice a big difference in audio quality if you own samsung phones with 192khz/24bit sound card (or if you have some audiophile music player with same sound card but 10x the price).

Conclusion: 44.1khz playback is better if you have better sound card and audio processing stack.


This.
Linux MIDI is fucked up. Don't even know if proper VSTi support exists on linux (at least there is some ~2005 unmaintained crapware that won't work because of outdated deps or some obscure audio distro that succ and nobody uses)


That doesn't remove the fact that systemd obviously provides a safe landing cushion for Windows task scheduler and services so they can finally make their very own distro (like AOSP, maybe WOSP) and sell apps spyware and tivoized or locked hardware based on their current business model :^)


Are you a mad winfa/g/?
pls go back >>>4chan.org/g/

Linux audio is cancer. Alsa, pulse, jack, midi etc. what a joke. There is a reason every serious audio studio uses mac and protools.

No it doesn't. Ubuntu studio makes a nice effort with ardour, but vst plugins basically don't work.

For similar reasons, every mac media production app reinvents its own GUI toolkit :^)

You might want to cut back on your red pills there matey. You're reaching Terry Davis levels here :^)

44.1 Khz is the Nyquist rate for the human hearing range, so it absolutely should be the default.

even 36khz would be enough, unless you are a literal baby

adults are lucky to even hear up to 20khz, and even then music doesn't use those freqs

this has to be bait

Unfortunately not.
Somebody just doesn't understand the difference between sound frequency and sampling rate.

Glad I am.

my nigger

As someone who's never worked with audio before why is it such a thorn in the side for Linux distros? I don't think it's the licensing because it's the software itself that always seems to be a pain

I love reading posts from retards who think the same units are only used to measure one thing. I assume you think wattage for CPUs and GPUs only means power consumption too, lmao.

So you telling me that Watt is not Joules/s?

Not sure what you mean by that. You have just written same thing W = J s^-1 = kg m s^-3. I think he had in mind that Watts are not just used for measuring electrical power. You can use W to measure heat, sound, light, RF output...

->

youtube.com/watch?v=8Jj3SKVvuoY

Yeah like 10Hz? Infrasounds are beyond human hearing yet you can feel them if you've ever been in a rock concert (chest vibration) or go get your subwoofer and playback brown note and shit yourself (literally).
Even deaf people and musicians who'd gone deaf can still feel low frequencies. Ultrasound also modulates the whole spectrum and there's a even a recent science journal about this (forgot link).

nice sampling faggot. ever wonder why speakers sounds crap than any live performance? (live perf without microphone or some sound system crap like real orchestra concerts)

24bit>16bitch

Not to mention that people report sounds beneath 20Hz as unpleasant at best and nauseating and sickening at worst.
Apart from retards who think that feeling explosions in movies with their subwoofer capable of producing those sounds is supercool, the rest of the sane population agrees that any soundsystem that produces frequencies beyond the hearing capacity of humans is unwanted and unneeded.

few animals can hear infrasound clearly like dogs and birds which are the first ones to go nuts before an outbreak of an earthquake (all dogs barking and birds flying).

Having golden ears isn't really that great.
>be the only person to hear "the hum thehum.info/

jesus this whole thread is fucking cancer.

Anything wrong with jack, apart the somewhat difficult setup?

I'll give you the benefit of doubt and say nice b8.

Even for my age I have good ears. When trains have to brake in a station, I am the only one covering my ears because the high pitch noise from metal on metal is hurting my ears. You only see young people do this.
But "golden ears" is what the placebo effect makes you hope you have. It's a common vice amongst audiophiles to believe that they can hear things others can't. Of course, there is a difference between people. But audiophiles claim they can hear the difference between a 24bit/96kHz recording and a 16/44.1 because they have superior senses. This is bullshit. ABX tests have proven this over and over again and scientists are still waiting to closely examine anyone who claims to have golden ears to let themselves get examined. The correct term here is 'good functioning ears', where old age and abuse (listening to loud music for extended periods of time) haven't lowered the hearing range down to 16kHz (which is the average for about anyone around 30). You would be surprised of all those audiophiles on the internet if they would take a test to examine their hearing range that most wouldn't come anywhere near 20kHz, while they claim they can hear all those high frequencies of their remastered Beatles albums bought on HDtracks.

And hear is good advice: never ever stand near the speakers on concerts or a night out. One night only can damage your ears beyond repair and even give you tinnitus. Use earplugs whenever the sound gets so loud you can't hear people speak when they talk on a normal volume.

Yet again someone who doesn't understand sampling rate != hearing range.

lol

You don't understand that frequencies is used both to describe the sampling frequency and the frequency of sound waves.

Stop trying to derail the thread because I post things audiophiles don't like to hear.

He uses the theory correctly you fucking retard.

He states that a sample rate of 44.1kHz, which would mean the maximum produced sound waves are at a frequency of 22.05kHz is far enough to cope with the average hearing range which is about maximum 16kHz for most people and 20kHz for young people. If you don't understand that, you should reread the wiki page for the Nyquist–Shannon sampling theorem.

It seems that there is still some confusion between sampling rate and hearing range. When someone states that his audio recording is 44.1kHz it means that ADC that was used to record audio sampled input signal 44100 times per second (Hz). In other terms: each 1/44100 of a second ADC reads input voltage, converts it to digital value and sends the conversion result to the next device. It's exactly same when you play back your audio. DAC in your sound card sets voltage on the output 44100 times per second based on digital input data.

44.1kHz is sufficient to cover human hearing range 20Hz - 20kHz. The reason why we have higher sample rates is because of syncing with video, 48kHz for example plays well with 24fps video 48000/24=2000 but there is a remainder with 44100/24=1837.5.

Stop making uninformed claims, it makes you look like an idiot. I'm kind enough to assume you aren't one.
Watch these videos: xiph.org/video/
Read this: people.xiph.org/~xiphmont/demo/neil-young.html

This also goes for all the special snowflakes here who are convinced that they have such special ears and such high end audio equipment that they require sampling beyond 44khz (for playback purposes).

Or you know, just continue taking your placebo. Remain happy and dumb.

I had never given A/V sync any thought when I encode videos, thanks for mentioning that.

Not that I have had any problems, but it sounds reasonable that the demuxing and decoding would be simplified.

I'd just like to interject for moment. What you're refering to as Linux, is in fact, GNU/Linux, or as I've recently taken to calling it, GNU plus Linux. Linux is not an operating system unto itself, but rather another free component of a fully functioning GNU system made useful by the GNU corelibs, shell utilities and vital system components comprising a full OS as defined by POSIX.

Many computer users run a modified version of the GNU system every day, without realizing it. Through a peculiar turn of events, the version of GNU which is widely used today is often called Linux, and many of its users are not aware that it is basically the GNU system, developed by the GNU Project.

There really is a Linux, and these people are using it, but it is just a part of the system they use. Linux is the kernel: the program in the system that allocates the machine's resources to the other programs that you run. The kernel is an essential part of an operating system, but useless by itself; it can only function in the context of a complete operating system. Linux is normally used in combination with the GNU operating system: the whole system is basically GNU with Linux added, or GNU/Linux. All the so-called Linux distributions are really distributions of GNU/Linux!

I wish there was 16/24 test.

44.1 kHz syncs perfectly with both PAL and NTSC due to the two being 50Hz and 60Hz AC respectively (with 25/30fps respectively). They had to make standards in order for audio to work on both ancient hardware so they went for 44.1 which is mathematically and magically works on both.

When high-end equipments started to show up (like Java-powered DVD players) they went for 192kHz and newer video formats showed up (plus dvds can have both NTSC and PAL mode). Back then if you use a different bitrate or sample rate for audio the video won't play or worse they would but the sound is out of sync. Computers these days are advanced and can decode a mix of unstandard formats or even terrible encodes.

There only thing relevant here is 24/16 bits since noise differences appear.
24bitvs16bit
Staircase isn't actually accurate way of depicting those signals but 24bit is more accurate than 16bit due to smoother and much more detailed sine wave.

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You could use a plugin for foobar2000.
hydrogenaud.io/index.php/topic,107354.0.html

No application "requires PulseAudio". At most, they may require libpulse which is a thin wrapper library that sends an audio stream to the PA server if it exists, and falls back to bare ALSA if it doesn't. You don't need to run the actual PA server for libpulse-using applications.


That really makes no difference. PA doesn't synchronize source and output sample clocks, so it will always resample your audio - even if the sample rates match. This is easily the most retarded and cancerous design decision in PA.

Suggest a good poettering-free distro to me. Not Gentoo or Void please.

This bait is just too strong. Here's your (you).
You're full of shit though.

disgusts

Even if it does/doesn't, does it really matter? Isn't the average person supposed to only hear around 20khz sound? 44.1/48khz's nyquist ranges are both above 20khz. so if one can do 22050hz and the other 24khz shouldn't really matter. Is there some fuckery i'm missing?

Yes. It's called aliasing.

I never understood audio aliasing can you give me a good example?
also I just fucked around on audacity and made a beep that goes from 30hz to 20khz and apparently, i've tested this on multiple soundboxes but i can't hear above 18khz or so

You can hear it yourself just take the signal you made and lower the sample rate and you will be able to hear it all.
Also audacity didn't generate a 30khz tone unless you set the sample rate to 60khz+.

neil-young should shut up his mouth.
Neynquist theorem states only for
44khz but infinite bits of resolution in every
sample.
There is a trade between resolution
in samples and number of samples (frequency)
For example you can use only 1bit but
you need a high frequency 2.MHz as sas-cd.
If you need more resolution you need to
increase sample resolution (bits) or number of samples (frequency).

WTF!!!
Dubs confirmed.
Are there people picking this bait?

I think you are confusing things up, mate. 44.1 kHz refers to the sampling rate of the data, not the sound frequency. The resolution of the digitalized sound wave, if you will.
I'm kind of sad you didn't insult OP while being so ignorant because I would've enjoyed ridiculing you more.

Lol, retards.

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I want freech to leave.

Shrug ¯\_(ツ)_/¯ Emoji

Imagine if getting monitor displays to work was as bad as getting sound to work... oh wait.