PulseAudio 10 Coming Soon, Using Memfd Shared Memory By Default

[duby229]

Meanwhile, on pretty much everyone's systems, no such issue is detected. Totally PA's fault, and not hardware specific.

Play a movie and ask anyone around you to look at a/v sync. I bet you money almost all of them will notice it.

EDIT: If you don't believe me, just grab a mic.... It's stupid high latency.

EDIT: Listen to Barrack Obama's Farewell speech.... I bet you that system used PA.... Yeah it was stupid high latency.

[RealNC]

EDIT: If you don't believe me, just grab a mic.... It's stupid high latency.

Latency is up to the recording app to configure. High latency means the application you use didn't request low latency from PA.

[Davidovitch]

So, how easy was the setup for the Scarlett? I've been contemplating to buy those after listening to music at a friend's, 'cause I really like how they sound. Plus, that'd let me remove a PCI card in my (sort of) cramped case.

Setup only involves plugging in the USB plug. On my Arch Linux boxes with Gnome3 that was/is the only thing required. Works really well.

[duby229]

Latency is up to the recording app to configure. High latency means the application you use didn't request low latency from PA.

Well, of course I should have guessed that.... Now it's application developers that cause the problem. It was ALSA drivers..... Now if we could just get every single app developer on linux to rewrite their software to work around PA design flaws. After all rewrites were already forced on ALSA drivers, and PA still doesn't work so the obvious choice is to pass the buck somewhere else. It worked once on the ALSA devs maybe it'll work again on every other dev. How great it is. You can always pass the blame. Eh, why not.

EDIT: The default behaviour absolutely needs to be, don't touch the audio stream -ever-, unless the user specifically configures it to do so. Which is exactly opposite of how PA is designed. PA latency is caused by PA and nothing else.

[ssokolow]

Good luck using alsa without gstreamer.

Gstreamer is much better than VLC. And ALSA is more than a driver. talk about being not understanding and ignorance.

This is literally the Linux version of saying "Good luck sending audio to your speakers without an MP3/Vorbis/etc. decoder". You've got the dependencies reversed.

1. VLC provides a GUI for controlling things like GStreamer.
2. GStreamer handles decompressing things like MP3 into raw audio.
3. PulseAudio or ALSA dmix handle mixing multiple streams of raw audio into a single stream, doing resampling if necessary.
4. ALSA drivers take that raw audio stream and feed it to the sound card.

One big reason things are such a mess is that, in the beginning (before AC'97 and Intel HDA audio), it was normal for sound cards to accept multiple audio streams and do their own mixing in hardware (and ALSA dmix came later and had noticeable flaws), so programs got in the habit of talking directly to the drivers and that was considered acceptable.

[liam]

My mother always told me never to argue with the stubborn and stupid. If he sincerely thinks that gstreamer is lower level than ALSA, then there's nothing you can do to convince him otherwise.

It's kinda weird to talk about these being higher/lower because gstreamer has multiple hackers.
You can target alsa with gstreamer so it's sorta lower level in that the app would be written to talk to alsa, which pulse would intercept then route (and only route) to alsa, but, pulse is still in the mix of it was installed and running. Tmk, the only ways to get around pa are: disabling/pausing it and using alsa directly, use jack2 with the pulseaudio plugin for jack-dbus (everything goes through jack---this is the current method to have the two sound servers running), or make sure none of your apps are using libpulse and uninstall pa.

[Davidovitch]

EDIT: The default behaviour absolutely needs to be, don't touch the audio stream -ever-, unless the user specifically configures it to do so. Which is exactly opposite of how PA is designed. PA latency is caused by PA and nothing else.

Not an expert here, but to my understanding you have no choice but to re-sample to a frequency that your hardware actually supports. It is not clear which type of cards support hardware mixing [1]. Your audio sources might come in various sample rates, and there is a high probability that at least some frequencies are not supported by your hardware. I assume things get complicated once you have multiple sources (at various sample rates) playing at the same time. If your hardware only supports one input stream at a given sample rate, you'll have to merge and re-sample the various streams into one, I could imagine.

[1] https://en.wikipedia.org/wiki/Hardware_mixing

[starshipeleven]

Play a movie and ask anyone around you to look at a/v sync. I bet you money almost all of them will notice it.

I don't notice any A/V sync issue neither on my tiny crappy laptop here, nor on my desktop that is using a dedicated PCI soundcard.

EDIT: If you don't believe me, just grab a mic.... It's stupid high latency.

Your sound card drivers must suck ass, still not noticing anything. Of course I'm not doing pro audio with pro tools, I'd use Jack for that.

EDIT: Listen to Barrack Obama's Farewell speech.... I bet you that system used PA.... Yeah it was stupid high latency.

Uhm..... no. Badly configured equipment or other limitations. It's not like PA is the only possible source of A/V desync issues IN THE WHOLE WORLD.

[wdb974]

Setup only involves plugging in the USB plug. On my Arch Linux boxes with Gnome3 that was/is the only thing required. Works really well.

Really? Latency and everything are on par with what's advertised on the box/website?

It depends on if your speakers are plugged in via SPDIF or the multiple analog stereo jacks. If the stereo jacks, then you need to plug in to the correct port. If the SPDIF, then it should "just work".

I must've worded myself poorly. I wanted to know if options like LFE crossover frequency and bass redirection exist under Linux. I happen to have a soundcard that sports an 8788 chip as well (I'm assuming that 8188 you mentioned was a mistake) and I just hate have low freqs in my satellites, 'cause they're not made for those. Also, LFE crossover lets me decide how often the subwoofer will be heard. I really dislike hearing the sub whenever someone drops a cigarette butt to the ground in a movie.

[Davidovitch]

Really? Latency and everything are on par with what's advertised on the box/website?

I have to run some latency tests, but I will have to come back to you later for that (remind me in a PM if I forget about it, don't mind running a few tests).