PulseAudio 15 Lands mSBC Codec Support To Enable Bluetooth Wideband Speech

[You-]

I don't understand why there is such a rush to bury Pulseaudio. The truth is Pipewire can´t replace Jack or Pulseaudio at this moment.
Maybe in the future, but not for now. Even Pipeware devs admit it: https://gitlab.freedesktop.org/pipew...commend-to-use

It says:

We recommend that you continue to use PulseAudio, JACK and ALSA APIs for now.

Pipewire implements those APIs.

As for ready and replacement, we are still in early days. For many use cases it is ready and with Fedora we will see whether they consider it ready soon enough and even then feedback after release of Fedora 34 will be crucial.

I dont hate pulseaudio - it was needed and a boon. But in the early days it was also misconfigured by default on Ubuntu, causing many problems. however it has raised the tide high enough that a replacement doesnt have to step on the same mines it had to. Even those who cant wait to move to Pipewire should be grateful for what Pulseaudio did and accomplished.

[schmidtbag]

Not relevant for a desktop user listening to music, since as you say, we cannot hear that. But, for a general audio architecture it is important to handle audio above what we can hear, since there are pro users mixing audio. And Pipewire is also aiming for the pro users with its Jack support.

I can't imagine why any pro user would care to hear something that high pitched. However, seems there's a bit of a misunderstanding here:

How is this relevant for ordinary desktop users? You can't hear anything past 22 kHz.

You're confusing pitch with sample rate. The average person very much can hear past a 22KHz sample rate, but very few adults can (let alone want to) hear past 22KHz pitch.

[ssokolow]

I have never had any problems with Pulseaudio in the last decade, so I plan to continue using it until Pipewire has been proven to be a real improvement without losing any Pulseaudio feature.

My brother called me up two days ago for a consult on how to diagnose crackling from his Blue Yeti under Linux. It was absent when run with pasuspender.

Also, I haven't re-tested them but, in 2016, I had to add launcher scripts with PULSE_LATENCY_MSEC=60 to various games (eg. FTL) to keep them from crackling on my system and one of the pieces of advice I give when crackling is encountered in an app is to try switching from the PulseAudio backend to using the ALSA backend and letting the libalsa-to-pulseaudio shim do the work.

[josmat]

Hi everyone. Long time Phoronix and Phoronix Forums reader here (8+ years), but just subscribed to comment on this since I work sometimes with pro audio.

I can't imagine why any pro user would care to hear something that high pitched.

Pro audio users don't use high sample rates primarily because they want to hear high frequencies, but because higher sample rates gives the audio more resolution for effects and processing. For the same reason they prefer higher bit rates: Audacity uses 32-bit float, and many commercial softwares use 64-bit float. In the final export everything is resampled to 44.1/48.0 khz 16-bit.

But there are people who say they can hear differences in audio quality with high sample rates, provided that high-end audio devices (dac, amp and headphones) are used. This is not because they can hear higher frequencies that the majority of people, but I think it's because downsampled audio can have artifacts that higher sampled audio have not (or have less).

[gufide]

How is this relevant for ordinary desktop users? You can't hear anything past 22 kHz.

The human eye cannot see past 30 fps. You should put your screen to that to save power. Either way, you won't see the difference. 120Hz and above are just marketing gimmicks, there is no such thing as "lower latency" or "increased responsiveness" or even "motion blur". Just send me all your screens, I'll send you back some that are at the correct and only true 30 fps.

[wertigon]

The human eye cannot see past 30 fps. You should put your screen to that to save power. Either way, you won't see the difference. 120Hz and above are just marketing gimmicks, there is no such thing as "lower latency" or "increased responsiveness" or even "motion blur". Just send me all your screens, I'll send you back some that are at the correct and only true 30 fps.

Actually, the human eye nerves samples at a frequency between 300 - 1000 Hz, so according to NyQuist Theorem, perfect refresh rate should be at least 2000 Hz. However, diminishing returns make the sweet spot around 150 Hz - any higher simply isn't worth it.

This means according to NyQuist, that the perfect FPS should be 300 fed to a 150 Hz screen fed to a human with perfect vision. If you want the perfect VR experience though, you want 4000 FPS on a 2 kHz screen.

[Xaero_Vincent]

I just switched back to Pulseaudio from Pipewire to test out an app called Soundux that can pass audio into apps like Discord while still using the microphone, which normally has sound streaming problems under Linux.

[royce]

How is this relevant for ordinary desktop users? You can't hear anything past 22 kHz.

Nyquist.

You're confusing audio frequency with sampling rate.

[royce]

For the same reason they prefer higher bit rates: Audacity uses 32-bit float, and many commercial softwares use 64-bit float.

You're confusing here resolution with bit rate. Bit rate is a measure of how much information you're using per unit of time to represent data. Whereas 32 or 64 bit applied to audio is the resolution of each sample.

[josmat]

You're confusing here resolution with bit rate. Bit rate is a measure of how much information you're using per unit of time to represent data. Whereas 32 or 64 bit applied to audio is the resolution of each sample.

I don't think I confused the two concepts, I just wanted to say the two things (sample rate and bit rate) increase audio resolution (in different ways), and that this is the reason why pro audio users prefer audio with high sample rate ("time resolution"?) and high bit rate ("sample resolution"?).