It's an easy target. It sucks far less all the alternatives, but it's an easy punching bag for people who want something to hate.

I lived through the days of horrible sound support on Linux - I think it was my main desktop until my first Mac around ~2007. Playing video or games was a crapshoot. Hot-plugging audio wasn't even in the realm of possibility.

Pretty much. You can even send the audio from different applications to different audio devices on the fly, which is something that Windows 7 can't do by itself (you have to change the default audio output before starting each application).

Pulseaudio was forced down people's throats too early by making it the default in Ubuntu and Fedora. People are still bitter about this. That's your "reasons". If these people were still using some godawful combination of ALSA, eSound, OSS and jackd to do things sound-related nowadays they would be complaining just as loudly about that audio stack.

If it works on FreeBSD it will probably work on Dfly. They're still pretty close in terms of userland, and I don't know this for certain but they probably take many FreeBSD changes and vice-versa.

> Excusing the horrible mess pulseaudio is with the horrible state of Linux Audio doesn't cut it,

The problem is that most people complaining about Pulseaudio don't offer realistic alternatives. Saying Pulseaudio sucks and that everyone should use ALSA is a joke. Pulseaudio does a plethora of things that ALSA does not handle. Not to mention that ALSA is a low-level system, and PA actually sits on top of ALSA.

In all of the vitriol that people spew about Pulseaudio, this is the first time that I've seen anyone point to sndio. I think that says something about the "pulseaudio complainers" crowd.

That said does sndio provide the following features:

- Support for bluetooth audio devices

- Support for streaming audio over a network.

- Support for user-land mixing of audio sources (i.e. don't need root).

- Mixing of multiple audio streams at the same time (e.g. Can your system play an alert sound without interrupting your music?)

- Per application volume settings

- Per application input/output source settings

Unplugging my headphones and letting other people be able to hear what I'm listening to? I mean, with ALSA I can shut down whatever program I'm using, switch to speakers from headphones, then start the program up again but that's a bit cumbersome. Maybe it is possible with ALSA somehow but not with the programs and distros I'm familiar with. When PulseAudio came out all of a sudden Linux went from less convenient to use headphones and speakers with than Windows to distinctly more convenient.

PulseAudio features I have used in the past that I don't think ALSA has:

* playing audio from/to another Bluetooth device (with bluez)

* playing audio on a PulseAudio server running somewhere on the network

* transparent encoding of audio output as DTS (for surround support over SPDIF)

I also seem to remember plain ALSA having trouble with multiple applications playing sound at the same time, which is just pathetic.

I'm sure some of these could in theory be built on top of ALSA, but fact is that today's Linux distributions use PulseAudio, and for me it always worked well.

I'm not sure about ALSA's capabilities in this area, but one thing PulseAudio does well is moving playing audio streams between sound devices. (For example, moving it from my monitor's speakers to my USB headphones.) Being able to control mixers on a per-application basis is also quite nice.

How do I make my bluetooth headset (which supports both A2DP and Handset profile) work with ALSA so it switches to headset profile when mic is active, reverts to A2DP when it's not and switches back to laptop audio when BT is off?

How's the situation for hotplugging sound devices in ALSA nowadays? The top Google results for "alsa hotplug" all point to editing configuration files or udev rules or shell scripts.

On the flip side, if everything can be done by ALSA, and PA is seemingly objectively worse, for what reason are people building their software against PA? Why is it being included anywhere at all?

Just so you and others know, the L in ALSA stands for Linux. It doesn't run on DragonFlyBSD. I think DragonFlyBSD uses a re-implementation of the OSS API that they inherited from FreeBSD. The other BSDs use /dev/audio, which was originally a Sun API.

Let's be fair: PulseAudio is the way to go if you have to have a partitioned audio system because you are using a multiheaded multiuser system where each seat has a monitor and keyboard and mouse and mic and speakers, and each user needs to be able to control their own volume mixing without root privileges.

For every other case I've seen, ALSA is better and causes fewer problems.

ALSA is not an alternative to Pulseaudio.

If your audio chipset doesn't have a hardware mixer, and about of half of laptop chipsets don't, then you can't play more than one stream at a time. If you have an array microphone, the only way it'll work at all is with Pulseaudio. If you want Bluetooth audio, only Pulseaudio bothers to support it.

Pulseaudio uses ALSA anyway (and on DFBSD it uses OSS instead); it's not a replacement, it is the only unix sound server other than Apple's(and maybe OpenBSD's) which actually works. ALSA is an audio device driver ABI; it can only expose what your chipset supports in hardware.

The problem of Pulseaudio spinning, at least when it used to happen on Linux, is due to issues in the individual audio device drivers. The reason this is a big problem is that Pulseaudio needs to run at a low `nice` value in the scheduler, or you will get buffer underruns (which sound like pops and clicks). Unfortunately this means that if Pulseaudio goes into a hot loop, it will consume all of the available CPU resources until the loop breaks. This problem can be mitigated by running pulseaudio with a control group to limit CPU time (as is done with systemd on many Linux distros). Unfortunately, DragonFlyBSD doesn't have the same thing configured, assuming their kernel supports it, so you get 100% CPU spinning.

Furthermore, DragonFlyBSD doesn't have ALSA; they have OSS.

I wouldn't really consider ALSA as an alternative to PA, considering that PA generally sits on top of ALSA instead of replacing it. aRts and ESD of the olden days would be more comparable.

PulseAudio is not a replacement for ALSA. At the moment ALSA is the only kernel level API for talking with the hardware drivers that's fully supported by upstream (= the kernel devs).

You _can_ install OSS4 if you want to, at the expense of loosing proper power management support; not a big issue on Desktops, but it can reduce a laptops battery runtime significantly.

Either way PulseAudo does not know how to talk to audio hardware by itself (well, it sort of does as far as Bluetooth attached devices is concerned, but the whole Linux Bluetooth stack "BlueZ" is quite finicky on its own). It needs (just like Jack, esd, aRTs, Xaudio* and all the other sound servers out there (*= dead and burried)) some way to talk to the hardware. And that, at the moment usually is ALSA.

When ALSA was first developed the original plan was to implement all the must haves (mixing, resampling) though the userspace module support. Unfortunately it ended up in an unholy mess and never worked satisfyingly. To anyone who's spent considerable amount of time developing audio software (in hindsight) that's not a big surprise. Audio raises very hard and tight deadlines. The way ALSA dmix implements mixing through IPC mechanisms is extremely prone to buffer underruns because of some participating process not getty enough CPU cycles in time to complete to write out the next bunch of audio.

PA does plaster over the biggest cracks in the foundations of the Linux low level audio infrastructure. And given the circumstances it does an admirable job there; if it doesn't break things even more. But it's only treating the symptoms and not curing the underlying issues.

Here's the laundry list of what's required to fix the major bumps in the road (which also PA has to navigate):

- major simplification and cleanup of the kernel side driver model. Too many things are left optional for the drivers to implement

- tight requirements on how kernel drivers must behave (there's a lot of variation in the runtime behavior of certain drivers; some return immediately from read write functions, others have significant delay, timestamps reported may be increment coarsely based on a per-buffer-length base, others deliver smooth playhead/recordhead position updates)

- mixing / fanning _must_ happen in or close to the kernel (the reason for that is explained in the next point). The biggest issue here is that resampling may be required. As long as it's upsampling it's rather unproblematic actually (the only DSP challenge there is not creating artifact frequencies). Downsampling is a much harder problem, because it involves lowpass filtering, i.e. discarding information and the challenge is only to affect the stuff above the Nyquist frequency.

- the audio infrastructure must be able to reschedule processes based on the deadlines and time left until new samples are required. Audio is probably the most demanding of the soft realtime applications. A single missed sample is easily noticeable even to the most untrained pairs of ears. If your buffers underrun you get pops or crackles. However on an interactively used system acceptable audio timing mismatches are in the order of 5ms before an untrained brain is able to pick them up. Raising audio applications to realtime priority alleviates the issues, but the better solution is to keep statistics on the audio frame lengths read/write by a process, how much time it takes for a process to prepare the next frame and use that to augment the scheduling. However this raises the issue being able to circumvent priority privileges through the audio system; a mitigation would be that frame sizes are quantized to a minimum frame size that correponds to the amout of CPU time the process would be alotted in regular scheduling (that's a tough one and I've been strugling with it for some time).

When it comes to OSs with a monolithic kernel a very good question is, if the last two parts absolutely have to happen in the kernel or if it's possible to have them in userspace. If we want to implement it in userspace, then, because of the rescheduling issues this will require the addition of userspace based rescheduling capabilities the syscalls required for this.

Of course with a proper audio infrastructure present through standard kernel and/or driver interfaces the need for a system like PulseAudio vanishes; at least as far as mixing and resampling are concerned. It's still desireable to have an audio manager process that knows how to decide on which device to play certain audio (telephone calls go to the headset, music goes to the speakers and so on).