antiX-23.1 coming soon [actually, it is AVAILABLE!]

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  • #127341
    Member
    Robin

      Hi all,

      that having the equalizer flat will represent the music as it was meant to be heard when recorded, anything else is a distortion.

      Unfortunately this is a wrong assumption. It’s only true if you listen at the very volume it was recorded (or mixed for). In all other cases the flat equalizer means acoustic distortion. Sounds weird, but is true.

      Let me try to explain in short.

      The basics you need to understand is that the human auditory perception is nonlinear, while your volume slider is linear.

      Graph: https://upload.wikimedia.org/wikipedia/commons/a/a2/Akustik_db2phon.jpg
      Source: Lecture notes “Akustik 2” by J.Blauert, Ruhr-Universität Bochum (RUB)

      It’s all about the different relative sensivity to different sound pressure levels at different frequencies. (stress here on relative)

      Script: http://www.sengpielaudio.com/GehoerrichtigeLautstaerkeregelung.pdf
      Source: Lecture notes “Gehörrichtige Lautstärkeregelung und die Hörempfindlichkeit” by Dipl.-Ing. Eberhard Sengpiel, Universität der Künste Berlin (UdK),

      From the graphs you can read the following:
      The relative sensivity of human hearing at a frequency of 1kHz is the same for a tonal sound with a sound pressure level (SPL) of 90 dB and for an SPL of 30 dB.

      But the graphs are not parallel throughout the full frequency spectrum of interest (20Hz to 20kHz), meaning a SPL reduction by a given dB value is perceived
      differently in different frequencies. As you can learn from the graphs, for low frequencies the sensivity for low sound pressure levels (30 dB-SPL graph) is significantly less than for higher sound pressure levels (90 dB-SPL graph) of same frequency.

      Please note: this is not about the absolute sensitivity which you get from the first diagram (the one from RUB script), but the slight differences in the run of these curves between the different graphs. It’s the differences which represent the relative sensitivity as a function of frequency.

      What consequence does this basic fact of human auditory perception have for playback?

      If you lower the sound pressure for all frequencies by the same dB value, the above said means you get a new mixing of perceived frequency composition (while physically and technically all was reduced perfectly linear throughout all frequencies): For low frequencies you perceive the reduction by a given dB value way more intense than for frequencies around 1 kHz (and to certain degree the same happens again for frequencies around 4 kHz, which is considered to be neglectable in practice). This changes the perceived acoustic colour (Klangfarbe) of the frequency composition (Frequenzgemisch) of the signal if you reduce (or rise) the sound pressure level of all frequencies the same. But precisely that happens, when lowering the volume by a non-compensated volume slider (that’s what you have in antiX*). The result is: The playback sounds unnatural compared with what you’d have heard if you’d have been present at recording set. Instead you’d need to take these characteristics of human hearing into account and lower the frequencies not by a fixed dB value throughout the full spectrum, but a bit less for low frequencies (neglecting the slight 4 kHz dent).

      My core statement is: If you play back something recorded at a distinct volume, you need a frequency correction to get the very same harmonic sound impression when listening to it on a significant lower or higher volume at home than that it was recorded on. In other words: Each recording has precisely one right level of volume it’s to be listened to without frequency correction. Otherwise you’ll not get the very tonal balance as it was intended by the artist and set by the sound engineer in the audio studio.

      What should you know beyond: Many recordings are already run through a frequency compensation for home playback on lower volume level than the original. You can’t know what precisely was done, so it’s try and error with the sliders to get best result for you personally.

      But what should be evident from the above is: a linear equaliser setting for all frequency ranges leads only to a satisfying result if you play back the recording at the very volume it was recorded (or mixed for by the audio engineer in studio). You probably won’t do this, if you know even a philharmonic orchestra can easily reach values of up to 130 dB at stage.

      Sorry if my explanation is possibly hard to understand. For me it’s pretty difficult to express all the technical studio lingo terms in English language, and moreover it’s many many years ago when I’ve worked in the audio studio of our university for some semesters. At that time everything still was analogue. Well, having now everything in digital doesn’t change the physical and psychoacoustical basics.

      But what you hopefully understand from the above is: If you want a linear (in terms of perception) reduction of volume, you must not set the equalizer flat, but taking the above into account and rise (or lower) slightly some frequencies (simplified: low frequencies), amount depending on the difference of the SPL a recording was taken with and the SPL you play it back with. And you’d probably want to compensate for shortcoming of equipment (if any) additionally, and adapt the output to the acoustical characteristics of the room you’re attending in (or even for the characteristics of headphones).

      That’s the theory. Now comes the practical part.

      My suggestion: Set it the way you feel it sounds right for you, and keep in mind the curves you hopefully also have internalised now. They can help you to make reasonable decisions and give an orientation what’s sensible. That’s precisely the way I do it. I also don’t start sophisticated calculations first. The most important part is: listen carefully whether you run into digital (from processing chain) or analog (from output amplifier) clipping. Try to compare in your mind with what you recollect from live events (or from playing an instrument yourself if you have the chance to).

      From my findings it is important to lower the presets done by pipewire pavucontrol and alsamixer as a very basic step (but that may depend on hardware). With the factory settings pipewire comes with I sense clipping on low output volumes already, somewhere in the processing chain you’ll find a slider set way to high in these cases. Be aware: pipewire resets alsamixer sliders constantly to something different (provoking clipping) unnoticed in the background.

      Essential is:
      move the sliders carefully, minimalist movements can have great effect already. You can use the mouse wheel instead of the pointer on them for fast precision movement.
      – the higher the volume is on which you want to listen, the less you need to raise the deeper frequencies on a recording of a loud sound event,
      – this goes vice versa also: the lower the volume is on which you want to listen a recording of an already low sound event, the less you need to reduce the deeper frequencies.
      – in case you play the recording at the same volume level as it was recorded (or mixed for by the studio engineer) you’ll have to use the flat equalizer setting.
      – beware of too “fat” bass. This can easily happen (but some people like it even that way).
      All this applies only if you want to listen the sound the way it was intended by the artist and/or the audio engineer. And in this case it moreover depends on the room you are attending in, and on the characteristics of your equipment. Pretty much imponderabilia, but you can actually use the graphical equalizer even for compensating to a certain degree many shortcomings of your equipment or the positioning of the speakers, as well as specific audio characteristics of your room. Additional hint: There are some more plugins you can experiment with, e.g. the stereo tools, allowing e.g. even stereophonic base expansion, output phase change or channel swap. Also there are tools for automatic loudness compensation available, then all the above said will be done for you automatically while merely moving the master volume slider.

      In the end counts what settings meet your preferred aural palette, and to find this you’ll have to invest a bit time for experimenting with different setups. There is no right or wrong in your listening habits and predilection. Small slider movements can make great differences, even between totally distorted sound and (for your personal sensation) perfect setup with full clear sound throughout all frequencies. And sometimes a single slider decides whether you get what you head for.

      So a good pair of studio headphones and a flat equalizer is the way I enjoy music these days.

      That’s nothing wrong. Having read the above you know now, that a flat equalizer doesn’t mean you always get the sound as it was recorded, not even with perfectly linear headphones.

      Whether easyeffects, just being installed does at all improve sound quality, without particular tweaking

      Unfortunately not. Maybe we can find a way how to put some reasonable presets in antiX. That’s the future 🙂 For now user has to click the effect tab, and then from the left hand menu the button “add effects”, select the equalizer, and then use it the way I’ve described above.

      —-
      *Remark: We could have loudness compensation for plain alsa and pipewire both in antiX:
      plain alsa: https://github.com/dpapavas/alsaloudness
      pipewire/easyeffects: sudo apt-get install mda-lv2
      This would require some testing and finding out how to create with these programs reasonable presets.
      And then an additional button in volumeicon reading “loudness comp. on/off”.

      In the hope of having eliminated all remaining clearness 😉
      Best regards
      Robin

      P.S. From my screenshot below you can take how minimalist the sliders should be moved. The compensation curve shows the theoretically needed compensation if you want to listen at an SPL of 30 dB without changing it’s acoustic colour to a playback of a recording recorded at an SPL of 90 dB originally.

      Windows is like a submarine. Open a window and serious problems will start.

      #127344
      Member
      olsztyn

        The basics you need to understand is that the human auditory perception is nonlinear, while your volume slider is linear.

        This is an amazing lecture. Thank you very much @Robin!
        I think I start to realize that the music recording we are listening to is never 100% fidelity the same as it was recorded and mixed originally and we can only approximate such fidelity through adjustments if we are very familiar with that piece of classic music. Such adjustments will need to compensate for acoustics of listening environment and audio equipment we use.

        I remember in the past (about five years ago) I found somewhere (and implemented in antiX) some ALSA tweak to adjust resampling rates so ALSA fidelity would be enhanced but I do not remember the details as I later (with later version of antiX) did not bother any more to continue these tweaks. So I am starting fresh now with your lecture, which I am thankful for.

        So could you please clarify in my mind some elements I am still not clear on:
        – As re-sampling is performed on the path from the original recording from analog input to digital, does every such re-sampling gradually degrade fidelity of frequencies and volumes at particular frequencies?
        – For producing the best fidelity of a digital material in antiX setting, you stressed on alsa equlizer being installed and present although pipewire does not use it as equalizer and easyeffects should be used instead. I may have mixed up something here though, so I apologize.
        Could you clarify what exactly needs to be done in that respect, as I am not sure I have clarity on what actually needs to be done in terms of installed components.
        – If pipewire components (assuming this is indeed the case) do in fact reduce fidelity vs. ALSA, would this be due to imperfect design of pipewire components to convey the original fidelity or it is due to the fact that re-sampling is performed on the way and resampling at different rates such as 48000 instead of 44000 (I could be wrong on these freequencies too) contributes to loss of fidelity.

        Thanks again for this informative lecture. I will keep this in my database, so it will not get lost being just in this forum…

        • This reply was modified 5 months ago by olsztyn.
        • This reply was modified 5 months ago by olsztyn.

        Live antiX Boot Options (Previously posted by Xecure):
        http://antixlinuxfan.miraheze.org/wiki/Table_of_antiX_Boot_Parameters

        #127357
        Member
        blur13

          Yes, thank you Robin! You obviously know what you’re talking about. Yet another rabbit hole to explore.

          #127366
          Member
          Robin

            you stressed on alsa equlizer being installed and present although pipewire does not use it as equalizer and easyeffects should be used instead.

            I found the sliders in alsa equalizer are simply without any function when pipewire is activated. So the logical consequence from this is we need adequate means for pipewire.

            Alsamixer-equalizer is to be there for people forced (or preferring) to rely on plain alsa. Pipewire-only users actually don’t need it.

            If pipewire components … reduce fidelity vs. ALSA, would this be due to imperfect design of pipewire components

            I don’t see this. You can have the very same clear sound with pipewire (given your PC can stem the additional load from running it on top of alsa). From what I’ve experienced with pipewire by now it’s all a question of getting all the sliders right. Keep in mind, the audio signal runs through a processing chain, I guess it looks something like that:

            sourceprogram → pipewire(internally: sources → processing → modules → sinks) → alsa(internally: preamp → subchannels.in → master → subchannels.out) → soundchip → amplifier → speakers

            In each link of this chain clipping can happen. If you get poor sound, you need to check all links of your chain. For example:
            – If your audio source program has an independent internal soft volume control (not just linking to system’s master control), make sure this is set in a way the output at this position of processing chain doesn’t run into clipping already, while it’s as high as possible. Play with the slider a bit to get a feeling whether it causes the issue.
            – Next, check the same way in pipewire pavucontrol the setting of the respective source entry on playback tab. Here you’ll find the slider for adjusting the level for this processing step.
            – Then again, check in pipewire pavucontrol the output devices tab. There you’ll find the currently playing device, and again a slider, controlling the signal level in this processing step.
            – If equalizer engaged, it also has it’s own control sliders for input and output. Try rising or lowering them, to find a position for max steam without running into clipping. You can derive it from the level indicator next to the sliders, if they are turning the colour, there’s probably clipping. But you have to listen in the end to make a decision.
            – Next sound runs through alsamixer. So open it and check the slider or sliders of pipewire. There should be a single slider (if running analog or digital stereo duplex preset) or 4, maybe 6 slideres if running pro audio preset). Pipewire raises them to absolute max. Lower them a bit to see whether clipping comes from this chain in processing link.
            – Check the physical volume control knob of your amplifier (in a notebook there might be a wheel of a rotary potentiometer peeping out). Make sure it is lowered to a position the signal doesn’t run the transistors of its output stage into saturation (causing clipping again). Also make sure the signal peaks don’t exceed the capabilities of your speakers, again sound distortion would arise from that.

            Be aware, all links in the chain must be adjusted individually in a way not producing clipping. If your signal is clipped in an early link already, sound will stay distorted even when you lower the volume using the master control in the end of the chain and you won’t see any overload in output level indicator in this case.

            does every such re-sampling gradually degrade fidelity of frequencies and volumes at particular frequencies?

            Not only for particular frequencies, but for all frequencies.

            Resampling (or more precisely: Samplerate conversion) means always a loss of original quality. If you look at the graph found in https://de.wikipedia.org/wiki/Abtastratenkonvertierung (sorry, the respective English language Wikipedia entry doesn’t explain this detail) you’ll understand what happens when converting from one sample rate to another: The algorithm has to interpolate the proper signal information for a point in time for which no information is present in the original sample rate. The software has to make sure to reduce the error in signal rising from this guessing task to a minimum, so you won’t hear it, but a resampled audio signal is never the same quality as the original. Well, it’s bean counting, if the conversion is done properly by the software, the distortion should be minimal and not noticeable when listening. This goes as long some basic conditions are met: The lower one of the sample rates must not be lower than half of the maximal frequency found in signal to be converted (see https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem#Aliasing ). Since we are talking of audio signals 20Hz to 20kHz a lower sample rate of 44,1 kHz meets this condition. But keep in mind: each sample rate conversion means a slight loss in signal quality. These conversion errors are truly relevant for mastering from original recording material, and can be neglected when listening something just for entertainment.
            Btw, this goes the more for lossy storage formats like mp3. It’s equivalent to editing a lossy jpg image and resaving it. Here the lossy compression adds some more errors. That’s why you should generally avoid to resample something from lossy formats.

            If pipewire components (assuming this is indeed the case) do in fact reduce fidelity vs. ALSA, would this be due to imperfect design of pipewire components to convey the original fidelity or it is due to the fact that re-sampling is performed on the way and resampling at different rates such as 48000 instead of 44000 (I could be wrong on these freequencies too) contributes to loss of fidelity.

            I don’t know how pipewire handles this internally. But a conversion should be avoided if not really needed. Sometimes it can’t be avoided: If playing an audio file recorded in 48 kHz sample rate on a device only capable to process 44,1 kHz sampled audio data (or vice versa), you (read: neither pipewire nor plain alsa) have any choice. It must be converted, otherwise you would not get any output to your speakers. It doesn’t count whether alsa or pipewire does the processing. But if both matches already, no conversion up and down should be done, and I don’t think pipewire would do it without need. Edge cases might be if your device could handle 44.100 Hz and 48.000 Hz both, and alsa is set to 44.100 Hz in a configuration file. Then pipewire has to convert even if this was not needed.
            To get an idea what your device is capable of, you can read from the output of the following set of commands:
            aplay -l
            (For the following command the value for x in for hw:x,y is found as first value in the lines starting “Card:x”. The y value is the value named “Device:y”. Example:

            Karte 0: MID [HDA Intel MID], Gerät 0: VT1708S Analog [VT1708S Analog]
            → hw:0,0
            Karte 0: MID [HDA Intel MID], Gerät 1: VT1708S Digital [VT1708S Digital]
            → hw:0,1

            Then enter (replace 0,0 by your findings from above):
            aplay -D hw:0,0 /usr/share/sounds/alsa/Front_Center.wav
            In case it complains about device in use, you need to stop pipewire server for this testing and make sure no other program uses the audio device the same time. Expected output:

            HW Params of device "hw:0,0":
            --------------------
            ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
            FORMAT:  S16_LE S32_LE
            SUBFORMAT:  STD
            SAMPLE_BITS: [16 32]
            FRAME_BITS: [32 192]
            CHANNELS: [2 6]
            RATE: [44100 96000]
            PERIOD_TIME: (62 11888617)
            PERIOD_SIZE: [6 524288]
            PERIOD_BYTES: [128 2097152]
            PERIODS: [2 32]
            BUFFER_TIME: [125 23777234)
            BUFFER_SIZE: [12 1048576]
            BUFFER_BYTES: [128 4194304]
            TICK_TIME: ALL
            --------------------

            The line RATE: [44100 96000] is what you are after. This example means 44,1 kHz and 96 kHz sample rates are supported by the hardware. (Now I wonder a bit myself, since I’m pretty sure the reading from my notebook was RATE: [44100 48000 96000] on older antiX versions. Will have to check once I’ve time for this)
            You can do the same with arecord to learn what audio input sample rates (e.g. from line-in or microphone) your hardware supports:
            arecord -D hw:0,0

            that the music recording we are listening to is never 100% fidelity the same as it was recorded and mixed originally and we can only approximate such fidelity through adjustments if we are very familiar with that piece of classic music. Such adjustments will need to compensate for acoustics of listening environment and audio equipment we use.

            Precisely that’s the gist of it. And it’s not restricted to classic music, it goes for all styles an genres of music. No, let’s say for all kind of audio output to your speakers, regardless whether built-in to a notebook or standalone in your room, connected to lineout of the PC, and also for headphones. Please remember, old tube radios (I still run one of them, I like it’s sound. Sad they’re going to render it inutil by abandoning UKW transmissions) used to have some push switches, reading: “Bass”, “Orchestra”, “Jazz” and “Solo”, along with two turnwheels reading “Treble” and “Bass”. Guess what these where good for?

            Windows is like a submarine. Open a window and serious problems will start.

            #127375
            Member
            PPC

              34% to 53% ?! This is ridiculous for just playing sound from a single source!

              I agree… but I can’t replicate such high values of CPU use in my single core Atom netbook – I’m not using it right now, but I can tell you the extra resource usage I notice is about 16Mb of extra idle RAM being used, and I notice no big change in CPU usage.

              From all my computers: a work desktop (dual Core CPU with 8Gb of RAM), my home PC (a quad core with 4Gb of RAM), my netbook (single core Atom CPU with 1 GB of shared RAM) and my old 32bits HP laptop (with 1 GB of RAM)… only the laptop has good speakers. On all the devices I tested antiX 23 and 23.1, sound is way, way louder than using. Also, I do not notice as much sound degradation when using “software amplification”- as a rule of thumb, I play YouTube videos, usually from the same you-tuber- if I can ear and understand every single word that is said, then I’m pleased with the sound quality. Please notice that I’m only talking about my own personal experience and use case -understanding spoken words.

              I think anticapitalista found the perfect solution for antiX’s audio problems:
              – on 64bits (full edition)- enable pipewire by default, having an easy way to switch back to Alsa, if users so require/need.
              – on 32bits -use just Alsa for sound.

              I was lucky that all my hardware (all my 64bits devices, I’ve not tested antiX 23 on my 32bits laptop) works great with pipewire. But it’s not all a sea of roses: my work desktop (a dell computer that does like to act up) when using antiX 23.1 test ISO 2, does not play sound out of the box, it requires to toggle pipewire off and then back on (sound always worked with test ISO 1, and always working using antiX 23 with the then called “pipewire fix”, that’s now part of antiX 23.X).
              Maybe more important than that, when stress testing on my home desktop pc- running games from the Epic store (those that run on Wine/proton, run great), then suspending then waking up the PC… I sometimes have no sound- “dummy output” comes up on volume application- and, you guessed it, I have to toggle pipewire off and back on, in order to have sound. This may be connected to suspending not working perfectly always, and may not be a pipewire problem.

              In all, if pipewire works with you hardware, I think it’s perfect, to help make antiX an even easier to use OS- all sound problems can literally be solved with 2 flicks of a switch… (that is so easy that even non tech minded users can understand: Is sound giving you problems? Turn the sound “thingie” off and then turn it back on. People are used to doing that since they first started using computers).

              Personal note:
              I’ll have to save, in order to read carefully Robin’s posts about audio – they are great, in all senses of the word.

              P.

              #127410
              Moderator
              Brian Masinick

                Regarding the audio specifics, our friend @Robin either knows, or has researched, far more than I have ever known specifically about computer-based audio systems.

                I’ve listened to some really good systems over the years. Once upon a time there used to be a company called Audio Technica that sold and supported some really good audio systems, and from two things – the specs of the equipment and the actual sound produced by various models – but without really knowing anything else, my EARS could detect some great stuff from the best equipment.

                I’ve never been able to own a truly premium audio system; perhaps the best thing I have right now are a pair of Anker Soundcore Life Q10 headphones. None of my systems, particularly my laptop models have very good sound coming out of their speakers; when I actually want to listen to something musical I put on the Anker headphones; these are also relatively modest priced (and now out of date) headphones, but they charge via USB and they can operate either from Bluetooth or from a directly connected cable. From these I can obtain sound that is definitely pleasant; to claim they are an audiophile’s dream would be a gross exaggeration. They are enough to make listening occasionally to music a pleasant experience.

                I defer to others, especially @Robin, when it comes to anything beyond this!

                --
                Brian Masinick

                #127415
                Member
                olsztyn

                  But it’s not all a sea of roses: my work desktop (a dell computer that does like to act up) when using antiX 23.1 test ISO 2, does not play sound out of the box, it requires to toggle pipewire off and then back on (sound always worked with test ISO 1, and always working using antiX 23 with the then called “pipewire fix”, that’s now part of antiX 23.X).

                  Looks to me something easy to fix. From symptoms it looks like pipewire is not properly initialized upon start, but it gets properly initialized via flipping that switch. You could check if all pipewire components are running upon start and single instance each.
                  As published by abc-nix, the pipewire itself should be started from session sturtup and pipewire-pulse and wireplumber should be started in turn by pipewire, these specified in pipewire configuration file.

                  I’ll have to save, in order to read carefully Robin’s posts about audio – they are great, in all senses of the word.

                  This is what I am doing so as not to lose this particularly valuable knowledge.

                  Live antiX Boot Options (Previously posted by Xecure):
                  http://antixlinuxfan.miraheze.org/wiki/Table_of_antiX_Boot_Parameters

                  #127661
                  Member
                  PPC

                    I have antiX 23.1 test iso 2 installed and I just installed several dozens of updates!

                    – IceWM was updated!
                    – Control Centre was updated to include the newer (yad) IceWM Control Center (that does not include the latest localization, at least not in pt-pt)
                    – Control Centre now includes the possibility to install and use a Pipewire equalizer (that, for some reason fails to install on my pc, the terminal window just remains open, with nothing happening after I confirm I want to install the packages)
                    – Control Center now has tooltips for all entries (not all tooltips are localized in pt-pt)
                    – antiXradio updated automatically – great app! I’m happy I had a small part in helping develop it (no pt-pt localization were installed, though).

                    antiX is becoming more feature rich, but still without “bloat”. Pipewire, if the equalizer installs and works, will be very nearly perfect (for those lucky enough to have hardware that works with it).

                    Edit: the “locate” command is not working (in antiX 23.1 installed version). I even tried running plocate-build

                    P.

                    • This reply was modified 4 months, 3 weeks ago by PPC.
                    • This reply was modified 4 months, 3 weeks ago by PPC.
                    #127888
                    Member
                    Robin

                      Here comes another one: antix-pipewire-goodies, fresh from workbench. Paint not completely dry yet. I sense this is necessary to master pipewire for antiX users.

                      I’ve added some more presets I’ve cobbled together from my knowledge about (analog) audio processing chains, equaliser curves and effect devices.

                      Just check out: antiX-pipewire-goodies. Designed to tame this mighty beast named pipewire a bit more again. Once installed you’ll find it in antiX main menu, section multimedia, named “antiXAPS”. If you have better proposals for this naming, just bring it on.

                      @anticapitalista: If you like it, feel free to merge it to your existing antiX-pipewire-extras, probably no need to have two of them around. Or on contrary possibly, keep it separate so the modules can be installed and updated independently of each other. Do it the way you feel it’s best in case you want it at all. I’ve merely put it into a package for people easily to check out and test here. It’s not on gitlab by now. GPL v.3 as always.

                      Please report all errors, issues and missing dependencies, which might cause to fail some of the expected filters, so anticapitalista can set them properly, just in case. I do hope I have extracted from my experiments all what actually counts. Consider the filter curves merely roughly cobbled together by now, they can and will be improved bye and bye in future updates.

                      What is in the package: This goodies package adds some dependencies and puts some easyeffects preset files I’ve tailored for antiX in place. Some of them are based on what I found at
                      git clone https://github.com/JackHack96/EasyEffects-Presets
                      And then it brings a script file I’ve written, providing some shortcut buttons to avoid user has to click through all the rich featured menus, tabs and pulldowns of easyeffect just to switch antiX to another acoustic colour. Fine tuning can be done in easyeffects program window if needed.

                      Best regards
                      Robin

                      P.S.: For now it borrows the window icon and desktop symbol from antiXradio. Will get it’s own icon soon.

                      —-
                      Zip archive with installer package is attached below screenshot

                      Windows is like a submarine. Open a window and serious problems will start.

                      #127897
                      Member
                      PPC

                        added some more presets I’ve cobbled together from my knowledge about (analog) audio processing chains, equaliser curves and effect devices.

                        Hum… I think all that knowledge may even make Xunzi come to the “dark side” and try Pipewire again? I hope so. I really do think Pipewire (despite using a tiny bit more resources than Alsa, but, like it’s said in John Wick 4 – “such is life”) solves almost every single audio problem antiX usually faces.
                        As usual, nice and very inspired idea, Robin! Many thanks.

                        P.

                        #127943
                        Moderator
                        caprea

                          Edit: the “locate” command is not working (in antiX 23.1 installed version). I even tried running plocate-build

                          sudo updatedb.plocate
                          fixed it here

                          @Robin,I have encountered a strange problem with the new option in Control Center > Hardware > Equalizer for pipewire
                          I can’t remmber if this button came through an update or If I installed it manually from the forum.I think it came through updates.
                          A small window pops up asking me if I want to install easyeffects, if I say yes a terminal opens, so far so good,

                          Hit:1 http://security.debian.org bookworm-security InRelease
                          Hit:2 http://deb.debian.org/debian bookworm-backports InRelease        
                          Hit:3 http://ftp.de.debian.org/debian bookworm-updates InRelease       
                          Hit:4 http://ftp.de.debian.org/debian bookworm InRelease                       
                          Hit:5 http://ftp.halifax.rwth-aachen.de/mxlinux/packages/antix/bookworm bookworm InRelease
                          Reading package lists... Done
                          Reading package lists... Done
                          Building dependency tree... Done
                          Reading state information... Done
                          The following additional packages will be installed:
                            calf-plugins libebur128-1 libfftw3-double3 libfluidsynth3 libfmt9 libgsl27
                            libgslcblas0 libhwloc15 libinstpatch-1.0-2 libsigc++-3.0-0 libtbb12
                            libtbbbind-2-5 libtbbmalloc2 libzita-convolver4 lsp-plugins-r3d-glx
                            timgm6mb-soundfont
                          Suggested packages:
                            ladish libfftw3-bin libfftw3-dev gsl-ref-psdoc | gsl-doc-pdf | gsl-doc-info
                            | gsl-ref-html libhwloc-contrib-plugins fluid-soundfont-gm
                          Recommended packages:
                            libhwloc-plugins
                          The following NEW packages will be installed:
                            calf-plugins easyeffects libebur128-1 libfftw3-double3 libfluidsynth3
                            libfmt9 libgsl27 libgslcblas0 libhwloc15 libinstpatch-1.0-2 libsigc++-3.0-0
                            libtbb12 libtbbbind-2-5 libtbbmalloc2 libzita-convolver4 lsp-plugins-lv2
                            lsp-plugins-r3d-glx timgm6mb-soundfont
                          0 upgraded, 18 newly installed, 0 to remove and 0 not upgraded.
                          Need to get 24.9 MB of archives.
                          After this operation, 70.8 MB of additional disk space will be used.
                          Do you want to continue? [Y/n] Y
                          

                          And there it hangs, nothing happens anymore.Does someone else see this?
                          It’s antix23 base here, I installed pipewire manually.

                          —————————————————

                          There’s a problem with the installation of antix-pipewire-goodies.deb

                          $ LANG=C sudo apt install '/home/helga/Downloads/antix-pipewire-goodies.deb' 
                          Reading package lists... Done
                          Building dependency tree... Done
                          Reading state information... Done
                          Note, selecting 'antix-pipewire-goodies' instead of '/home/helga/Downloads/antix-pipewire-goodies.deb'
                          Some packages could not be installed. This may mean that you have
                          requested an impossible situation or if you are using the unstable
                          distribution that some required packages have not yet been created
                          or been moved out of Incoming.
                          The following information may help to resolve the situation:
                          The following packages have unmet dependencies:
                           antix-pipewire-goodies : Depends: easyeffects (>= 7.0.1) but 7.0.0-1 is to be installed
                          E: Unable to correct problems, you have held broken packages.
                          

                          7.0.1 would be the one from backports.

                          • This reply was modified 4 months, 3 weeks ago by caprea.
                          • This reply was modified 4 months, 3 weeks ago by caprea.
                          #127957
                          Member
                          Robin

                            antix-pipewire-goodies : Depends: easyeffects (>= 7.0.1) but 7.0.0-1 is to be installed

                            7.0.1 would be the one from backports.

                            Hi @Caprea,
                            as you’ve said, you can get this version when activating debian backports repo. This is a hard dependency here since the 7.0.0-1 has some bugs causing the script to fail when trying to file the switching commands to easyeffects (while current 7.1.3-1 from sid fails in it’s current state to load the needed filters properly.)

                            I can’t remmber if this button came through an update

                            Yes, and now. The button was present always in antiXcc, but had a bug in it’s test condition so it never showed up. While fixing this, I crafted it to handle new pipewire in one go, and so the button has some magic in it now: If running on a pure alsa system it reads “Alsa equalizer”, but running on a pipewire activated system it reads as you have described. Depending on whether the equalizer is installed already or not, it starts either the easyeffects or asks user to install missing packages first.

                            Need to get 24.9 MB of archives.
                            After this operation, 70.8 MB of additional disk space will be used.
                            Do you want to continue? [Y/n] Y
                            And there it hangs, nothing happens anymore.

                            Why does the installer doesn’t proceed for you?
                            From what you’ve posted above I have the impression it’s apt that hangs on processing the installation or waiting for the files to drop in. I don’t see anything in acc what could cause it. Is it possible your internet connection failed the very moment apt tried to download the 24 MB of package files?

                            Best regards
                            Robin

                            Windows is like a submarine. Open a window and serious problems will start.

                            #127967
                            Moderator
                            caprea

                              what could cause it. Is it possible your internet connection failed the very moment apt tried to download the 24 MB of package files?

                              Of course I have already ruled out the most obvious causes. The problem repeats itself when I now boot the system and try the installation through the button again.

                              and so the button has some magic in it now: If running on a pure alsa system it reads “Alsa equalizer”, but running on a pipewire activated system it reads as you have described. Depending on whether the equalizer is installed already or not, it starts either the easyeffects or asks user to install missing packages first.

                              This all works nicely.Except that it then hangs here during installation. Probably once again one of the phenomena that only occur on my system.

                              #127971
                              Member
                              Robin

                                I have already ruled out the most obvious causes.

                                Then let’s have the mechanism itself on trial, something on antiX base could cause it to fail. Here on antiX 23 full it works flawlessly, on 32 bit and on 64 bit both. I have designed it to be not critical on behalf of timing.

                                Probably once again one of the phenomena that only occur on my system.

                                It could be an issue with the mechanism not working for some reason on base. Does it work for you on antiX full?

                                I installed pipewire manually.

                                This doesn’t count obviously, the switch has sensed pipewire’s presence properly as designed, otherwise you’d have been presented with the Alsa-EQ.

                                Up to the point where the installer starts everything looks good to me from your description. Then let’s start at that very point.

                                At this point a subroutine sets a lockfile in /dev/shm (you can check for it, it is named acc-install-lock-(followed by the process ID number)
                                – So the first question is: does antiX base provide the /dev/shm directory? Can a default user write to it in antiX base?
                                The lockfile makes sure the control center is reloaded not before the installer has completed, so the logic behind the button will detect the freshly installed easyeffects and allows it to come up instead of again asking for installation.

                                Could you please check whether the lockfile /dev/shm/acc-install-lock-(followed by the process ID number) is written and removed properly while the issue occurs? If this file is sticky after install was completed, antiXcc main window won’t come back. What happens if you manually remove it while the install window is still open?

                                But it rather looks to me that apt never completes it’s task for some strange reason.
                                So please ask in a terminal window while the installer seems to wait for the 24 MB to drop in, whether it possibly has done it’s work already in background silently without letting us know:

                                which easyeffects
                                apt-cache policy easyeffects

                                If nothing comes back from this, your system is chewing on the command sequence:

                                acc_install_lock="/dev/shm/acc-install-lock-$$"
                                touch "$acc_install_lock"; gksu apt-get update && gksu apt-get install easyeffects lsp-plugins-lv2; rm "$acc_install_lock"; echo done.;

                                So next question rising from that is: Is gksu present on a base antiX?

                                And: Does the sequence work for you if feeding it manually to a command line?

                                Depending on your findings from these questions we can proceed in singling out the culprit.

                                Windows is like a submarine. Open a window and serious problems will start.

                                #127973
                                Moderator
                                caprea

                                  Ok, before I saw your post I made a second test.
                                  Installed again clean antix23 base on new partition, updated and full-upgraded the system, installed pipewire again manually, rebooted. The new button is in CC,. choosed again yes to install,typed in my password when prompted , terminal opened, typed Y and the terminal hangs again.
                                  So it is excluded that this is a problem caused by me due to some strange configuration I did to the system before.

                                  does antiX base provide the /dev/shm directory? Yes

                                  helga@antix1:/dev/shm
                                  $  ls -al 
                                  insgesamt 0
                                  drwxrwxrwt  2 root  root   100  4. Jan 13:20 .
                                  drwxr-xr-x 16 root  root  3960  4. Jan 13:11 ..
                                  -rw-r--r--  1 helga helga    0  4. Jan 13:13 acc-install-lock-3612
                                  -rw-r--r--  1 helga helga    0  4. Jan 13:20 acc-install-lock-5570
                                  -rw-r--r--  1 root  root     0  4. Jan 13:11 .tmpfs

                                  What happens if you manually remove it while the install window is still open?
                                  Nothing

                                  please ask in a terminal window while the installer seems to wait

                                  $ which easyeffects
                                  helga@antix1:~
                                  $ apt-cache policy easyeffects
                                  easyeffects:
                                    Installiert:           (keine)
                                    Installationskandidat: 7.0.0-1
                                    Versionstabelle:
                                       7.0.1-1~bpo12+1 100
                                          100 http://deb.debian.org/debian bookworm-backports/main amd64 Packages
                                       7.0.0-1 500
                                          500 http://ftp.de.debian.org/debian bookworm/main amd64 Packages
                                  

                                  is gksu present

                                  $ apt-cache policy gksu
                                  gksu:
                                    Installiert:           2.1.0antix1
                                    Installationskandidat: 2.1.0antix1
                                    Versionstabelle:
                                   *** 2.1.0antix1 500
                                          500 http://ftp.halifax.rwth-aachen.de/mxlinux/packages/antix/bookworm bookworm/main amd64 Packages
                                          100 /var/lib/dpkg/status
                                  

                                  Same result from terminal, blinking cursor , terminal hangs

                                  helga@antix1:~
                                  $ acc_install_lock="/dev/shm/acc-install-lock-$$"
                                  touch "$acc_install_lock"; gksu apt-get update && gksu apt-get install easyeffects lsp-plugins-lv2; rm "$acc_install_lock"; echo done.;
                                  OK:1 http://deb.debian.org/debian bookworm-backports InRelease
                                  OK:2 http://security.debian.org bookworm-security InRelease                    
                                  OK:3 http://ftp.halifax.rwth-aachen.de/mxlinux/packages/antix/bookworm bookworm InRelease
                                  OK:4 http://ftp.de.debian.org/debian bookworm-updates InRelease
                                  OK:5 http://ftp.de.debian.org/debian bookworm InRelease
                                  Paketlisten werden gelesen… Fertig
                                  Paketlisten werden gelesen… Fertig
                                  Abhängigkeitsbaum wird aufgebaut… Fertig
                                  Statusinformationen werden eingelesen… Fertig
                                  Die folgenden zusätzlichen Pakete werden installiert:
                                    calf-plugins libadwaita-1-0 libcairo-script-interpreter2 libcloudproviders0
                                    libebur128-1 libfftw3-double3 libfluidsynth3 libfmt9 libgraphene-1.0-0
                                    libgsl27 libgslcblas0 libgtk-4-1 libgtk-4-common libhwloc15
                                    libinstpatch-1.0-2 libsigc++-3.0-0 libtbb12 libtbbbind-2-5 libtbbmalloc2
                                    libzita-convolver4 lsp-plugins-r3d-glx timgm6mb-soundfont
                                  Vorgeschlagene Pakete:
                                    ladish libfftw3-bin libfftw3-dev gsl-ref-psdoc | gsl-doc-pdf | gsl-doc-info
                                    | gsl-ref-html gvfs libgtk-4-media-gstreamer | libgtk-4-media-ffmpeg
                                    libhwloc-contrib-plugins fluid-soundfont-gm
                                  Empfohlene Pakete:
                                    libgtk-4-bin libhwloc-plugins
                                  Die folgenden NEUEN Pakete werden installiert:
                                    calf-plugins easyeffects libadwaita-1-0 libcairo-script-interpreter2
                                    libcloudproviders0 libebur128-1 libfftw3-double3 libfluidsynth3 libfmt9
                                    libgraphene-1.0-0 libgsl27 libgslcblas0 libgtk-4-1 libgtk-4-common
                                    libhwloc15 libinstpatch-1.0-2 libsigc++-3.0-0 libtbb12 libtbbbind-2-5
                                    libtbbmalloc2 libzita-convolver4 lsp-plugins-lv2 lsp-plugins-r3d-glx
                                    timgm6mb-soundfont
                                  0 aktualisiert, 24 neu installiert, 0 zu entfernen und 0 nicht aktualisiert.
                                  Es müssen 30,0 MB an Archiven heruntergeladen werden.
                                  Nach dieser Operation werden 93,0 MB Plattenplatz zusätzlich benutzt.
                                  Möchten Sie fortfahren? [J/n] J
                                  
                                  • This reply was modified 4 months, 3 weeks ago by caprea.
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