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PipeWire 1.0 Released For Managing Audio/Video Streams On The Linux Desktop

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  • #21
    For people interested in learning about digital media, I strongly recommend 'Monty' Montgomery's videos on the topic on this page:

    Xiph.org: Video

    or, more directly:

    Episode 1: A Digital Media Primer for Geeks
    This first video from Xiph.Org presents the technical foundations of modern digital media via a half-hour firehose of information. One community member called it "a Uni lecture I never got but really wanted."​
    Episode 2: Digital Show & Tell
    The second video from Xiph.Org explores multiple facets of digital audio signals and how they really behave in the real world. Sampling, quantization, dither, band-limiting, and vintage bench equipment all in one video!
    The 'stairstep' is a lie. If your understanding of a reconstructed audio waveform from a digital encoding includes a stairstep in the final output, your understanding is wrong.

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    • #22
      Originally posted by Quackdoc View Post

      This isn't how it works, there are plenty of hardware that have noticable audio quality degradation if they are running below their optimal sample rate. and even if you have a dac that could output right, there are plenty of reasons to want a higher sample rate input/output, recording music for instance, should always be done at higher sample rates.
      Then that depends on the DAC hardware, as already said in another comment. On my DAC, there is no audible difference between sample rates (it's NAD D1050 if you're interested). Like I said in my previous comment, there might be small difference only if a high resolution source is downsampled to 44100hz / 48000hz in software.

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      • #23
        Originally posted by Danny3 View Post
        Wonderful!
        The only other thing I want from it is to obtain the battery stats from devices like speakers and headphones connected over Bluetooth and report it back.
        Some deskto environments already show battery levels for devices like mice and game controllers connected over Bluetooth and it would be nice if they could show it for speakers and headphones too with the help of PipeWire.
        PulseAudio can get this info from version 16:
        The freedesktop.org project announced the release of the PulseAudio 16.0 sound server with some new features. Here's what is new.


        So, I think PipeWire should be able to get it too.
        I know you're a Debian user as I am so maybe you have to do the same as I had to do. Maybe it's configured differently in other distributions or maybe not but here I had to enable experimental features for the bluetooth stack:

        In /etc/bluetooth/main.conf I had to set Experimental to true so the stack processes battery information from my bluetooth headset.
        It works perfectly so I wonder when this is not experimental anymore. I did not experience any diametral effects enabling.


        Edit: Oh, boy. I've just seen you already got that answer several times But, yeah, what the others have said....
        Last edited by reba; 27 November 2023, 02:48 AM.

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        • #24
          "Look, spotted the delusional guy who lives in his non-scientific bubble."

          I've been accused of many things but that's a first - rest assured I'm at least as aware of all the "theory" involved as you are (not only do I have four years of college-level physics under my belt, my aspie sigint dad taught me electrical theory when I was six).

          "So how high can you hear in frequency huh?​"

          Make that far more aware: as stated by reavertm above, Sampling rate != Sound frequency. The upper range of my hearing (which, incidentally, I have to have tested every two years - and the sensitivity of which always blows the mind of the person testing it) doesn't enter into the discussion.

          I'm the farthest thing from a "deluded wannabe audiophile" - I run twenty-year-old Class A/B amps (the butchered sound of Class D quite literally makes me mildly queasy) through no-name cables powering thrift store speakers - albeit good thrift store speakers like B&W's, Paradigms, DefTechs and Klipsch References (I had the good fortune of living and working in a ski area for the past ten years where Texas oil money regularly donates the unused electronics in their multi-million dollar chalets to charity).

          You probably won't be inclined to acknowledge the fact but we don't all have equally-mediocre nervous systems: not only do I have amazing hearing, my sense of time is far more fine-grained than that of most people (I can entertain a dozen different thoughts just while you're struggling to complete a sentence). I used to be utterly unable to listen to uncompressed 16bit recordings of Upside Down by Diana Ross because of the artifacts; it's only now that I'm able to enjoy (far-smoother) 24bit recordings of it that it's not jarring to my ears.​

          "To me what "sounds good" is something that reproduces the exact input signal it's fed, as close as possible."

          You hit the nail on the head in spite of yourself; the above is all that matters and represents the entirety of the approach I stick to.

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          • #25
            PipeWire is definitely an improvement over Pulse, but it still has issues with USB DACs that have both an output and input component. Specifically, it has issues with activating both the output and input at the same time. There are multiple bugs about this on the git repo, one of which I filed myself weeks ago but has 0 comments on it.

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            • #26
              Originally posted by Type44Q View Post
              The upper range of my hearing (which, incidentally, I have to have tested every two years - and the sensitivity of which always blows the mind of the person testing it)
              The only time I had the frequency I can hear tested by a medical professional is when I was suffering from high-blood-pressure-induced pulsatile tinnitus. And that procedure is irritating as hell. Wtf are you doing to warrant that done every 2 years?
              Last edited by BreachScrambler; 26 November 2023, 02:31 PM. Reason: Phoronix editor sucks on mobile.

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              • #27
                Originally posted by reavertm View Post
                Sampling rate != Sound frequency
                Yeah, sampling rate is just the upper limit of frequency, and that's literally it.

                Every DAC in the past 30 years or more filters the conversion through capacitors so the result is a perfect sinusoidal analog wave. The only thing higher sampling rate does is enable it to have higher frequency content available. But you, as a human, probably won't even hear above 18k Hz, much less 21k Hz (limit of 44k sampling rate). Highest frequency that can be output by a given sampling rate is half of it.

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                • #28
                  Originally posted by Type44Q View Post
                  "Look, spotted the delusional guy who lives in his non-scientific bubble."

                  I've been accused of many things but that's a first - rest assured I'm at least as aware of all the "theory" involved as you are (not only do I have four years of college-level physics under my belt, my aspie sigint dad taught me electrical theory when I was six).
                  "Theory"?

                  Did you measure it?

                  Show us your "analog" wave from the DAC being "stair stepped". Measure it, like Xiph.org did. All you're doing is "theory".

                  See this post if you're curious and want to know facts instead of living in delusion. Scientific facts, that is. You know, science is based on measurements.

                  Originally posted by Type44Q View Post
                  "So how high can you hear in frequency huh?​"

                  Make that far more aware: as stated by reavertm above, Sampling rate != Sound frequency. The upper range of my hearing (which, incidentally, I have to have tested every two years - and the sensitivity of which always blows the mind of the person testing it) doesn't enter into the discussion.

                  I'm the farthest thing from a "deluded wannabe audiophile" - I run twenty-year-old Class A/B amps (the butchered sound of Class D quite literally makes me mildly queasy) through no-name cables powering thrift store speakers - albeit good thrift store speakers like B&W's, Paradigms, DefTechs and Klipsch References (I had the good fortune of living and working in a ski area for the past ten years where Texas oil money regularly donates the unused electronics in their multi-million dollar chalets to charity).

                  You probably won't be inclined to acknowledge the fact but we don't all have equally-mediocre nervous systems: not only do I have amazing hearing, my sense of time is far more fine-grained than that of most people (I can entertain a dozen different thoughts just while you're struggling to complete a sentence). I used to be utterly unable to listen to uncompressed 16bit recordings of Upside Down by Diana Ross because of the artifacts; it's only now that I'm able to enjoy (far-smoother) 24bit recordings of it that it's not jarring to my ears.​

                  "To me what "sounds good" is something that reproduces the exact input signal it's fed, as close as possible."

                  You hit the nail on the head in spite of yourself; the above is all that matters and represents the entirety of the approach I stick to.
                  No, you have no idea what you're talking about.

                  There is zero subjective opinions or "golden ears" in this discussion. You can measure the output analog wave from a DAC. That's a scientific fact and is not subjective. Until you do, you don't know anything.

                  And I already answered the sampling rate != frequency in my post above. It's directly related, and that's all higher sampling rate does. That's all it does.

                  Every DAC worth its salt will output sinusoidal analog waves up to sampling rate/2, because it's easy to do so. Period.
                  Last edited by Weasel; 26 November 2023, 02:51 PM.

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                  • #29
                    This project rocks indeed. It's like having coreaudio on Linux.

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                    • #30
                      Originally posted by Weasel View Post
                      "Theory"?
                      Every DAC worth its salt will output sinusoidal analog waves up to sampling rate/2, because it's easy to do so. Period.
                      Actually it is not that easy to do. That will require an infinitely steep anti aliasing filter...

                      So the better DACs will instead up-sample the signal and apply digital filtering to relax the transition band requirements and thereby the phase-shift of the analog filter.

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