PipeWire 0.3.62 | Hacker News Reader

PipeWire 0.3.62

gitlab.freedesktop.org

149 pointsjrepinc94 comments

I use PipeWire on my desktop and laptop, and its doing a better job than pulse ever did.

That said, I've had daily issues with it, from it slowing video playback (weirdly, by about 10 seconds over 20 minutes or so, only noticable when watching something synchronized with someone else), over crackling noises with speakers when no audio is playing, to straight up lags and skips every few seconds with certain devices.

Audio on linux is a complete joke, and I'm glad pipewire is there to tackle it.

I've experienced that same lack of synchronization with other people.

I can recommend Syncplay. For a variety of reasons, but it’ll also deal with desync.

I don't think it works on Twitch...

If you haven't, post an issue on the Pipewire repo. I've found that the main author, Wim Taymans, is very proactive in fixing any issue if they're reproducible.

I've opened two bugs, and two times he fixed them personally in less than a week, which is incredible for an open source project of that size.

I've had the same experience. Had a couple issues and they ended up fixed in the next version.

Video playback timing is actually pretty difficult, with only trade-offs, no "right" solution.

The easiest solution is to use your computer's accurate clock to time the frames. This means, if your video is at 30 FPS, you will have gone through exactly 3000 frames after 100 seconds. But your monitor might not be refreshing at exactly 60 FPS, it might be targetting the 59.94 standard, or it might be targetting 60 FPS but with a clock that's a bit less accurate than your computer's, so every now and then, you get a stutter as a video frame has to shown for 1 refresh interval longer or shorter than it should.

What we could do instead is to calculate a fixed pattern to display frames. If your monitor is 30Hz and the video is 30 FPS, we show exactly one video frame for every monitor refresh. If your monitor is 60Hz and the video is 30 FPS, show each video frame for 2 monitor refreshes. If the monitor is 30Hz and the video is 60 FPS, skip every other video frame. This will already introduce some deviation from the "ideal" timing if your display clock isn't super accurate. And if the monitor is 59.94Hz, that's probably close enough to 60 that we want to show each frame of 30 FPS video exactly twice, but that means the video is playing at 99.9% of its intended speed.

10 seconds over 20 minutes is a surprisingly big gap though. Playing every frame of 30 FPS video twice on a 59.94Hz monitor would net a 1.2 second deviation over 20 minutes.

EDIT: Why the downvotes? Everything here is correct according to my understanding of the situation after implementing video playback software and working with other people's. If anything is wrong, please correct me, don't just silently downvote.

Excuse my ignorance, but don't monitors take their clock signal from the host? I would think that synchronizing a stream of frames to the display refresh rate is a mismatch to handle on the display end, not the host end where frames might get doubled or skipped (and eventually introduce long-term timing errors). This is assuming an 60Hz rate advertised by the monitor.

I don't know whether the clock is technically in the host machine or in the monitor, I imagine it depends on the type of connection. But the idea of a different "display clock" and "playback clock" still applies, since a program's own timekeeping based on usleep + gettimeofday will be different from the monitor clock, wherever that happens to reside.

And unless you're using some kind of variable refresh rate, the monitor always, always refreshes on a fixed interval. It doesn't do any "synchronizing a stream of frames to the display refresh rate". But with VSync, you can synchronize your software with the display refreshes, which is how you can decide to draw every frame twice and that sort of stuff.

Pixel clock is generated on the host. All the VSYNC/HSYNC pulse timings are also generated by the host. Monitor has little choice but to respect it, or refuse to operate.

On some embedded platforms, if you can't get a precise clock you need from clock subystem, or massage the display mode timing to fit whatever pixel clock you can generate, you may be running your display at 34 FPS instead of 60 FPS if you're not careful eneough. And you'll barely notice unless you measure it, or try to play video or do anything else sensitive to timing.

Don't forget that while you might not notice a 20 minute video taking 1.2 seconds more than 20 minutes - but you will notice if the audio plays at the correct speed and ends after exactly 20 minutes - and you will probably notice before the video ends.

Yeah, you obviously have to sync audio and video with each other. It's just usually not that important to sync the audio and video with some "true" time as measured by an atomic clock.

Usually playback will be locked to a hardware clock, typically the audio clock. This does mean that the audio clock and frame clock will drift over time, as they're not locked to each other; it doesn't matter how accurate the clocks are, they will drift. This is one source of dropped/doubled frames. It's possible to sync to the video framerate instead, of course, though audio desync artifacts tend to be more noticeable and objectionable than video ones. You can handle this by dynamically measuring the clock drift and variably resampling the audio, but this is relatively complicated and computationally expensive. Generally this type of drift isn't too noticeable, these clocks are usually well within 100ppm, which would result in 1 frame slip every few minutes.

The issue with framerate mismatch is a fundamentally different one, in that it's not a synchronization problem. It's the result of a non-integral ratio between display frames and playback frames, and exists regardless of clock source. So it's the harder of the two to deal with, and is worse in magnitude too at several slipped frames per minute with e.g. 29.976fps vs. 30fps. In the classical approach, this simply results in frames getting doubled or skipped, since it's not tracking the video rate at all, it just asks for the display to be updated at its frame-time, and the display system will either get a chance to show it or not. The slightly more refined approach is to play back at the closest integral ratio (e.g. play 29.976fps video at 30fps on a 60fps display), and resample the audio to match this new rate, but again this is computationally expensive and complicated.

mpv team has a good writeup: https://github.com/mpv-player/mpv/wiki/Display-synchronizati...

I don't even know why this is even necessary. Back in the day I just set up the ALSA daemon to save/restore volume levels and that was it. In all programs I select ALSA as output and boom done. So I'm confused to why pulse audio and pipewire even exist.

Firefox can't do sound with raw ALSA. No, I know that doesn't explain why PulseAudio exists. But I think Pipewire can do more - isn't it supposed to be able to move video around as well as audio? Or something.

I also just use raw ALSA. It's there anyway, it works, and I don't need more daemons (I don't need NetworkManager either).

Because the complexity of audio is hard. If you just have a desktop system with stereo speakers it is easy. However if you ever plug a set of headphones in things get hard. Should all sound now go through the headphones or not? Different people will want different answers (I have headphones plugged in all the time, but I'm not wearing them all the time, so if someone sends me an IM I need the beep to come out the regular speakers). There are monitors that have built in speakers making this harder, does the user have better speakers plugged into the computer output or not?

That is the regular desktop user. If you do any for of audio processing (pod casting, music recording) you may have a large number of different inputs and output connected, and want to do mixing of some sort. Likewise if you do videos you may have more than one "webcam" that you want to mix. These are fairly common use cases (many people have a dream of being a musician or an actor, and so have this type of setup that they are trying to use in the efforts to make it big)

> Should all sound now go through the headphones or not?

The correct answer to that specific question: as a sane default, all sound should now go through the headphones. That's a sane default on the Iphone, probably on Macbooks, and it also happens to be the one that at least a stock Gnome or Ubuntu desktop chooses. In fact I have trouble thinking of a saner default than that one!

Forgetting about sane defaults seems to be a common problem in FOSS communication. It's unfortunate, because describing something as unknowable can convince a new developer to avoid inspecting the code around that particular problem.

E.g., it took a long time for Gnome to get the sane default I describe above. I'd bet that lurking in the appropriate mailing list history is at least one didactic post from an audio developer explaining how we can never truly know what a user's preferred setup could be.

Raw ALSA can only have one program playing sound unless your hardware has support for hardware mixing (this would generally mean it's very old and also very expensive). More likely you are/were using ALSA's dmix plugin which is a small barebones version of the core functionality of pulseaudio or pipewire. They replace dmix but also give you things like per-program volume control, better latency, better power usage, better mixing, bluetooth support, the ability to do fancy things with audio routing without restarting every program playing audio, etc.

I am not sure about "better latency". The latency with Pulseaudio cannot be controlled via GUI and is often quite large (on order of hundred ms or more). In my opinion, to get minimum latency one needs to access audio card directly, without intermediate daemons.

PA isn't great when it comes to latency, but JACK and PW are.

PulseAudio optimizes for power consumption but tries to be adaptive to handle lower latency as well. PipeWire seems to optimize for latency but try to be adaptive to reduce power consumption as well. These are conflicting desires, lower latency requires higher power usage because you have to use shorter buffers instead of just giving the sound card a big chunk of audio to work through and putting the CPU to sleep. JACK could potentially still do better and ALSA could potentially still do better than even that but PipeWire's goal is to be close enough or even match JACK while still having all the other features people use PulseAudio for.

I'm not really sure what the difference between "ALSA just works" and "ALSA as configured by my distribution and including the DMIX plugin just works" is.

Like at the end of the day option 2 is still "ALSA just works".

pulseaudio exists because it's user friendly.

JACK exists becase some people need low latency and dynamic control over non-trivial audio routing.

pipewire attempts to bring the best of both worlds and make it container friendly for security purposes.

> JACK exists becase some people need low latency

Wouldn't latency be lower if the application would access audio card directly without any daemons in between?

Then you would only be able to do fraction of functionality JACK provides you. May be enough for the most simple use-cases, but people who reach for JACK aren't usually after those.

The answer is, surpisingly maybe, no. JACK (and PipeWire) manage to add features without adding any latency to what ALSA can provide. Will it consume more CPU? yes, it does...

Because ALSA doesn't allow two programs to use the same audio card simultaneously. ALSA is good for exclusive access, when you want to remove everything between an application and audio card.

Also ALSA cannot do other nice things, like switching audio output when you plug an USB audio card.

It can and does as it was configured by most distributions.

The dmix plugin does, not ALSA.

Because they do things many users find useful:

-Per-application volume control

-Seamless hardware switching mid-stream (plug in headphones)

-Different channel configurations into the same device simultaneously

-Cross-application volume ducking

Most of the things people below say ALSA didn't have, OSSv4 actually had. But basically, they made it proprietary, Linux switched to ALSA, and when OSSv4 became open source again it lost traction. But from my experience at the time, it had none of the problems ALSA had, it had none of the problems pulseaudio had, and it had none of the problems ESD had.

This is a clear case of when ideology won over an objectively better product to the detriment of all users for more than a decade.

Ehhhh OSS is really not great, even v4. Re-using Unix file I/O functions for audio sounds clever, but falls apart pretty quickly in the real world. Even in the best case where you manage to shove every audio feature into open, read, write, and ioctl, it's never going to feel good to use (type safety...).

I don't mourn the loss of OSS.

OSS relies on mixing in the kernel, which is bad for reasons explained here: https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/FAQ...

Also with OSS you still can't seamlessly switch between devices, bluetooth or arbitrary signal routing between apps. So, no... good riddance.

By the time pulse finally worked right!

2060: PipeWire11 is being deprecated and replaced with a compositing audio server named Weyband

Not to defend Pulse too hard but I use Linux every day and am specifically interested in audio and outside low latency work (where I use jack), Pulse has rarely been a problem since many many years. What exactly are the issues people experience?

I regularly hit a couple.

This one breaks audio and video in virtual machines:

https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/46...

The effects of this one vary:

https://gitlab.freedesktop.org/pipewire/wireplumber/-/merge_...

For me it breaks audio in Chrome and VLC:

https://github.com/wwmm/easyeffects/issues/1789

I guess I wasn't very clear.

It was issues with Pulse I wanted to know more about. Low latency is obviously not it's forté, otherwise it just works, for me that is.

You were pretty clear. I just completely misread. :)

> That said, I've had daily issues with it, from it slowing video playback (weirdly, by about 10 seconds over 20 minutes or so, only noticable when watching something synchronized with someone else), over crackling noises with speakers when no audio is playing, to straight up lags and skips every few seconds with certain devices.

How is that "better than pulse ever did"? The only issue I had with PulseAudio on several machines in last decade or so was Bluetooth A2DP connection being very capricious.

I have recently switched to PipeWire and it actually improved Bluetooth, but I had to disable Wireplumber's libcamera backend because it made it regularly eat 100% of RAM. No complaints after doing that. Crackling noises or desynchronized video playback would instantly take it into "unacceptable" territory.

Sounds like you have severe issues with both PA and PW. Maybe your hardware is the problem?

I estimate about 75% chance it's a configuration problem on his end.

The "I hate Pulse Audio and especially Lennart" crowd tends to want to either go back to "Alsa" or tried at some point to install the old OSS drivers and then gave up.

This means trying to configure dmix to work properly, trying to configure your audio outputs with tools like 'alsamixer' and things of that nature.

This is about a 100% guaranteed approach to screwing up audio in Linux. For years it is what unsuspecting newbies were told by "really smart people" on the internet when faced when any sort of audio issue and all it really does is ensure that their OS audio is going to turn into a dumpster fire.

The only way to fix it is to aggressively find your "custom" audio settings for every application, find all the alsa configuration files and delete them. And then find out where your distribution saves your alsa mixer settings between reboots and link them to /dev/null and reboot.

Audio settings of this nature are very persistent and nothing Pulseaudio or Pipewire can do to help you fix it.

> The "I hate Pulse Audio and especially Lennart" crowd tends to want to either go back to "Alsa" or tried at some point to install the old OSS drivers and then gave up.

Well, yeah, since when pulse showed up and started breaking people's audio it effectively replaced alsa (even though they kind of operate at different layers). So if you didn't like that, of course the solution was to rip out Poettering's bugfest and revert to alsa. Thankfully at some point pulse stabilized a lot and in my experience tends to just work out of the box these days, but people tend to have their habits set by the early versions that were pushed out before they were ready.

As someone using strictly alsa/dmix/alsamixer for decades now, I'm surprised to learn my OS audio is a dumpster fire.

To me it's a low-feature deterministic stack that rarely does what I don't want it to.

But I haven't introduced a bunch of fuel for the fire like bluetooth audio devices either... to each their own.

> Crackling noises or desynchronized video playback would instantly take it into "unacceptable" territory.

> Sounds like you have severe issues with both PA and PW. Maybe your hardware is the problem?

Seconding this! My experience with PA and PW has been basically identical to yours. If possible, GP, try some different hardware! You shouldn't have to put up with that crap.

Hm, I might try that. I swapped headphones, mics, etc, the next thing to swap would be the motherboard, or to get a dedicated sound card.

> How is that "better than pulse ever did"? The only issue I had with PulseAudio on several machines in last decade or so

It's "better than Pulse ever did" because they do not have the same hardware as you, and because whatever they had before was worse.

Lesson 2 about discussing technical issues on the internet: other posters almost certainly do not have the same hardware as you.

(this is also relevant when it comes to discussing software performance - the fact that Slack loads in 5 seconds on your desktop does not make it "fast", because the entire internet does not have your desktop)

> Lesson 2 about discussing technical issues on the internet: other posters almost certainly do not have the same hardware as you.

Thanks for reiterating my point, I guess.

> How is that "better than pulse ever did"?

Pulse would cause games to crash or not even start (i know this because switching to pipewire resolved that), for example. Pulse, for me, was beyond broken on every machine I used it, in some capacity. PipeWire is, too, but in a way that doesnt ruin my day ;)

> Maybe your hardware is the problem?

Sure, could be, though my hardware is quite common.

>> Audio on linux is a complete joke

Yes, for example, the zoom app is now one of the most widely relied on apps for connecting people over networks. To get its audio output to fairly represent the input (e.g. for musical content), the input client is supposed to use the 'original audio' setting. That setting is not yet available on the linux version of zoom. On the output end, the audio is a mess, sounding as if somewhere along the way a handshake has turned into a fistfight.

Audio is complicated. Another example: an ordinary amateur, part-time, casual guitarist might nowadays play with thousands of dollars of pedals and processors plugged into his signal path. He gets this to work by plugging and unplugging those devices and their cables in various configurations until everything sounds best. Try providing that level of flexibility in software with reasonable default settings for all of the myriad of applications and environments in which one might try to apply computer audio.

* Audio on Linux "used" to be a complete joke. Pipewire fixed that.

For me, its still a WIP than a full fix, but definitely better than pulse :)

Why post some random point release?

Free karma? People like to talk about stuff that they like to talk about, so post a thread about stuff people like to talk about.

I ask that "why post X" question often, but that leads to "why upvote X", which answers it. Enough people find it interesting enough to discuss that it's found its way to front page!

Good question. More context would be good.

Stumbled into PipeWire few months ago when trying to setup Bluetooth headset on Ubuntu OS. It's superior to PulseAudio, but...

Really unbelievable to comprehend that Linux distros have such bad support for Bluetooth audio headsets - something very common in today's tech world. Android, Windows and Apple's OS all very efficiently switch from A2DP codec which is excellent for audio playback, but has no possibility for bidirectional communication (so you can't use your microphone) and HSP/HFP coded which provides worse sound, but you can talk. Linux doesn't have this fixed, there were some attempts [0], but nothing great came out of it.

[0] https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/18...

I'm actually confused; The past couple years, PulseAudio has been pretty good at switching between the two for me. Getting stuck in HSP isn't unique to Linux of course; it can happen if anything holds the mic open, there's reports of issues with both macOS and Windows too, for the same reason. Especially common with programs like Discord.

PulseAudio absolutely supports it with the Bluetooth module's auto_switch feature. I guess I'm not sure if Pipewire supports it, but it's likely handled by whatever is doing session management. I can only assume it's possible to configure Wireplumber to handle it if it doesn't have native support for it yet.

Hm, not really sure about `auto_switch`, didn't dig deep enough to find it I guess. PulseAudio only offered HSF/HSP which sounded simply terrible. PipeWire's HSP sounds pretty good (mSBC codec), although it's still obviously worse than A2DP. I'll have to look into auto switching and how to enable on either system.

Are you saying that in pavucontrol, you see no A2DP modes at all? That's very strange, since all Bluetooth certified stuff should at least support A2DP SBC mode.

Out of the box, Pipewire should also be able to support LDAC and maybe AptX. PulseAudio can use Gstreamer to support additional codecs, so on PulseAudio with GStreamer 1.20+ you can get better A2DP coverage, assuming you have the right codec packages installed on your distro.

In general, Linux Bluetooth support has been a lot better as of late, to the point where I didn't need to mess with much, so most of what I know about it is actually older. (I used to use a third-party PulseAudio module to get better A2DP support for example, but when I looked it up, I found that it was deprecated in favor of just using newer GStreamer.)

jelicicm mentioned ubuntu in the original comment. I seem to remember that as of Ubuntu 20.04, LDAC wasn't supported. I needed to jump through some hoops to get it to work. It was still using full-on pulseaudio, not pipewire, at the time. I wouldn't be surprised for it to still not support non-free codecs out of the box.

I've indeed had no issues on my Arch install. Works perfectly with LDAC on my Sony headphones and aptx-hd on my Shure.

The mic of the Sony is a bad joke so I never use it, but I've tried it and switching modes worked fine on Linux.

I hear you on that one. I'm patiently awaiting end-to-end support for LE audio with bidirectional audio streams, and maybe, if we're really lucky, Sony will release a pair of headphones that has a better microphone to go with it.

Sorry, not saying that. I do have an option of either A2DP or HSP. I was not familiar with auto switch. But you kind people have filled me in in this thread. A2DP sounds amazing for listening to music, no problems there. The problem is the hassle of switching between profiles, since I tend to use my headphones in work environment (calls, meetings) - this is the reason I still keep a pair of cabled headphones around.

Interesting. Not sure what the deal is. Someone is suggesting that at least for PulseAudio, it could just be the older version of PulseAudio that shipped with Ubuntu. However, you should be able to get it working on Pipewire just as well. Pipewire stuff is still pretty immature, so your distro may have an older and less stable Wireplumber at the moment. In that case, it might be worth just trying to set the Pipewire Media Session option for this instead. Wireplumber is really powerful, though, so it's a nice tool to have available.

+1 For Pipewire + Wireplumber! With it, automatic codec switching has worked quite well for me!

My personal Bluetooth experience also improved drastically when I finally got around properly importing the pairing keys from Windows into Linux. Now everything works flawlessly.

If anyone is dual-booting Linux and Windows, here's a gotcha to keep in mind: Bluetooth pairing works using keys associated the MAC address of a Bluetooth interface. If you pair a device on Windows (key A) and then try to pair/connect to the same device after booting into Linux (key B), it simply won't work without anything telling you the real reason: To the device it looks like someone is impersonating the host interface, since it was previously set up using key A, but suddenly someone (Linux) pretends to be the same host (MAC address still the same) but with key B. The solution is simply looking up the key in the Windows registry and then using the same key in Linux.

Unfortunately, the process is a little complicated, but the Arch wiki page [1] does a really good job of explaining this nowadays. I've previously tried to set this up a few years back, but it didn't really work. Perhaps the wiki page was expanded in the meantime, or I simply did something wrong back then.

Just posting this here hoping I might be able to help some people wondering why the hell some of their devices just don't want to connect on Linux.

By the way, AFAIK some devices just don't care about the keys, that's why you might only have issues with a subset of devices.

[1] https://wiki.archlinux.org/title/Bluetooth#Dual_boot_pairing

Not sure if this is what you were pointing out here, so just in case: Pipewire does allow you to switch BT codecs including the ones which do better bidi quality. Including autoswitching if you want it to happen automatically https://wiki.archlinux.org/title/PipeWire#Automatic_profile_... (you can switch manually as well)

Thanks. Will give it a try!

I came here to say the same thing. This feature has had its fair share of bugs over the last year but lately (using Wireplumber) it's been working quite reliably for me. I no longer use any Bluetooth USB dongles!

I've personally had a smoother experience on KDE with PipeWire because it seems to switch quicker for my headphones automatically, and worst case I can just right click it in the bottom right and select the correct codec.

On Mac OS X however with the same headset, there's no manual switching, and I have to try to "coax" it into switching back by playing media like Spotify and it's not super consistent.

Granted I only switch accidentally when I accidentally have my headphones selected as the audio source.

Are you complaining that Linux can't handle Bluetooth because PulseAudio breaks your setup, on a Pipewire post?

Use Pipewire then. As other have said, this is not a problem anymore with the recommended setup of Pipewire + WirePlumber.

What is really unbelieveable is that Bluetooth still cannot do high quality full duplex streams in 2022, and that anyone that wants to use wireless has to accept sounding like a telephone line / old radio from the early 90ties in meetings.

When I ask sales people "does their bluetooth headset actually do high quality full duplex streams" they give me the strangest look.

I am glad to see that I'm not the only one asking for this, i feel like i'm taking crazy pills.

My personal experience is the opposite-- although this behavior perhaps isn't desirable for end users who don't want to fiddle with their codec selection, being able to switch between A2DP and HSP manually in pavucontrol provides me with a much superior experience compared to Windows which would a) keep my headset in HSP because of some rogue application with a stuck open mic and make it really difficult to track it down/undo it and b) (rarely) keep my headset in A2DP while an application is attempting to grab audio or switch it to A2DP in the middle of meetings and break everything until I restart the application trying to use my mic. In my experience, a manual codec switch is a much smoother experience, ironically.

Wideband (mSBC) support was added into Pulseaudio 15 btw: https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/1...

It also added support for using higher bitpools for the SBC codec (sbc, sbc_xq_453, sbc_xq_512, sbc_xq_552) which pushes its quality above APTx and the other proprietary codecs. Info about bitpools and SBC here: https://habr.com/en/post/456182/

When using recent packages (e.g. NixOS) it's become quite good in the last ~6-ish months. It switches automatically when I open zoom, which is very nice

Had no clue.

https://pipewire.org/

Looks like video is getting a lot of work now.

Only slightly OT; does anyone have a solution for playing audio from ssh? The best I have so far is to pick a program that I know is under X (here xterm) and run something like:

    eval "env $(cat /proc/"$(pgrep -n xterm)"/environ |xargs -0 -n1 printf '%q ') bash"

To get the environment setup right.

(edit: I might have misunderstood your comment, but see my edit at the end)

If you use PulseAudio or pipewire-pulse on your local computer and that both computers are on the same network (or at least the machine you ssh into can access your local machine, possibly through a tunnel - PulseAudio listen to port 4713 by default), you can set the PULSE_SERVER environment variable.

  PULSE_SERVER=X.X.X.X whatever_should_play_a_sound

Where X.X.X.X is the domain or IP of your local machine. Your local machine needs to be configured to allow remote computers to play sound locally (by loading module module-native-protocol-tcp as far as I know - a papref setting is available for PulseAudio).

If the computer you ssh into has Pulseaudio or pipewire-pulse, on your local machine you can run:

  PULSE_SERVER=Y.Y.Y.Y pavucontrol

Where Y.Y.Y.Y is your ssh server, and set the default output sink to your local computer's. Then it will play things through your local computer. You need to setup your local computer so it advertises its PulseAudio service, and your server needs to have the module-zeroconf-discover module loaded.

You can load a module with pactl (for both PulseAudio or pipewire-pulse):

  pactl load-module module-zeroconf-discover
  pactl load-module module-native-protocol-tcp listen=0.0.0.0

edit: and if you just want to play something on your remote machine, then `PULSE_SERVER=localhost` can work. export $(dbus-launch) instead of your command possibly works too. With Pipewire on Debian I noticed it works out of the box though.

You are misunderstanding; I want to play audio on the remote machine, not on the local machine. The remote machine is running pipewire-pulse.

Yes sorry, I figured it out before seeing your comment and edited my reply.

PULSE_SERVER=localhost or export $(dbus-launch) could work.

Aha, I found more details, and also think I've solved it:

1. I misremembered it was an alsa-client that couldn't use the pipewire virtual device.

2. your mention of dbus did help though; setting DBUS_SESSION_BUS_ADDRESS=unix:path=/run/user/1000/bus seems to do the trick.

I usually just mount the remote drive with sshfs and play the audio locally.

This pulseaudio approach might work with pipewire too -- I haven't tested it but I love this text-mode audio player... https://mathr.co.uk/harry/#sound

Man, how much are distros + bad user config hurting user "Linux" reports.

Thank god for NixOS where (1) we do a decent job of base config and (2) user config errors are basically trivially identifiable.

1. Open `pulsemixer` to monitor pulse clients

2. `ssh localhost "pacat < /dev/urandom"`

Observe that pacat is showing as a pulse client. This stuff is really easy and should absolutely just work.

Did you reply to the wrong comment? This doesn't seem relevant to remote audio

Um, it's just an example? Does this somehow help?

1. (term1) `ssh remote pulsemixer`

2. (term2) `ssh remote "pacat </dev/urandom"`

Observe that (1) you can see a new client appear on the remote, and (2) audio will play from the default playback device on the remote (and you can trivially change that device with `pulsemixer`)

EDIT: for your speakers/ears-sake, I recommend turning down the default playback device before doing this.

I'm running nixos (22.05), and I can't run alsa clients to the pipewire alsa device on the remote machine over ssh without setting the "correct" environment variables (I have yet to identify them, so I use what I posted to clone the environment of an X program).

https://news.ycombinator.com/item?id=30621077

(And yes, I use pipewire. The socket is pipewire-pulse's.)

I was expecting pipewire to provide a pulseaudio ABI compatibility without the cost of the pulseaudio SDK(glib), that for 1 steam game. Well, it pulls glib into the SDK too. I forgot about that game, and I am back on my good friend alsa. The software mixer (dmix/pulseaudio/pipewire/jack/whatever) is hidden behind the alsa API.

Knowledgeable people, please explain. Wouldn't it be better if applications used ALSA for audio output? In this case they could work without Pulseaudio and with (because Pulseaudio adds a sink to ALSA). This way the application doesn't need to know which audio daemon it uses. What are the downsides of this approach?

I use pure Alsa on virtual machines. Image is stripped down so much it barely runs XServer.

Another reason is future proofing. PipeWire is yet another audio protocol, and may not be around in 5 or 10 years.

Not everything has PipeWire. Some flavours of linux has support for 10 years. Even some of my desktop boxes still use Pulseaudio.

And finally what PipeWire actually offers? If you need to dump audio into output, ALSA seems sufficient.

From what I've read PW's official stance is to keep using the existing protocols such as PA and JACK. PW comes with a PA-compatible server, replacement JACK client libraries and an ALSA plugin. So it supports clients from all three at the same time. Precisely so software does not need to be rewritten (I imagine).

People like to use the pulseaudio API because it has features people want, compares to ALSA or JACK: See here:

https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/FAQ...

Oh this thing, and of course it is amazing and popular. But removing it caused so much pain, it's like pulseaudio and asahi all over again.

My gripe is desktop developers and maintainers have this mindset that "users need this stuff, even if they don't want it we will make sure they have it and things will break otherwise".

My wish/hope is for stable/well supported distros like debian to have good support for "minimal" versions of popular DEs.

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