AV Linux 23.1 released

Forum Forums General Other Distros AV Linux 23.1 released

  • This topic has 65 replies, 11 voices, and was last updated Mar 15-9:07 pm by puredyne.
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  • #134194
    Member
    puredyne

      Segretario Segreto will not always be able to know what will happen…

      SARL Audiophonics sent a low cost PC-USB filter card but it was already broken. The failure demonstrates that the only melted component is on the external power input. We only powered internally and wrote a complaint via email… This caused Vania’s ASUS motherboard to melt… not bad but (5+5) 10 hours of work requires a week of relaxation for a patient with Fibromyalgia, Multiple Chemical Sensitivity and Electrosensitivity. Today was placed a new order of ~450 euros and ordered to give the grade A converter at the price of grade B to begin to tend to balance the karma.

      In the first chance given to the French, Audiophonics had persisted with cheating and insults. Audiophonics dot FR was used to start an unexplained electrical fire in a flower shop on Edlingerstrasse below the spy’s apartment. Three days later France made the United Nations vote for peace. In case the French insist on being assholes, there is something truly evil about the order. ESS converters have always given the impression of hiding something shady. You have to be careful about commercial beasts… 32 bit 384-768 kHz the definition is fine. There are Chinese and Chinese… for recording it may be better 32 bit or 32 float with Japanese ADC chip also limited to 192kHz… SSL12 with typically colored sound SSL4K we don’t know which ADC chip it has inside, but you will be able to connect a real Original/ Burr/Brown at the ADAT/TOSLINK port… 123dbBB original flagship samples truly extreme low-end. what happens when you connect the optical cable? Will only two external channels go 24 bit or the whole system? in any case there is no isolation for USB/xmos transport and you will need to add external isolation to inject ultra low noise power… in the preamp there is the classic 4000 Series discrete circuit and all those opams can always be modernized with greater detail . for a reasonable price we see decent capacitors onboard and the clock will be a toy… you can’t expect miracles.

      Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

      #134359
      Member
      puredyne

        Segretario Segreto so… already knew what was going to happen. Let’s see what we can do Among other things, SSL should donate 2-3 devices to the Linux Audio community for testing and possible development of Open Source Mixers. SSL does not recommend using a UAC2 compliant device with Linux and Android… is SSL crazy or a bitch? I don’t like the fact of Harrison Mixbus acquisition at all… However SSL12 can work due to physical limit at 192 kHz only in recording or only playback. Two 32/192 kHz channels for playback monitoring should be able to handle it while recording 2-6 channels but it would be safer when recording to set up two different DSPs for recording and playback.

        It must be understood what prevents working at 384 kHz with DIVA in Divine Mode (TempoReale)… does it depend on the current RTirq settings with open source video drivers or the Antix RAM release system that is too reactive?

        N.B. the tube continuously hides (reduces) the number of video views in the Supersonic Seduction channel… are the fascists really fucking terrified of this? ///
        the AP-reply automatically went into your spam folder… never open it (Miss Monique DJ’s Nazi-fascist Ukranian spirit has already been captured)

        * There was confusion, there is only one headphone output with 115db dynamic… ADC and everything else is around 111db… SSL did not answer a specific question and if they don’t say which ADC chip is used, the management marketing is ashamed of the machine.

        Ambiguously the SSL sale writes that the mixer runs at 32 float. So the risk is that there are the same miserable portable recorder converters. Or something very similar toy-105db-ADC, the DSP sums two channels sampled at 32 into a single channel transformed at ~111db 32FP. The suspicion is that it is not the mixer that generates 32 floats. By doing this with an onboard DSP, it becomes a low definition colored/branded digital transport.

        Linux Audio user/producer/master is too conscious for such crap. For this reason SSL will never give machines and sales to Linux Audio Professionals.

        • This reply was modified 1 month, 2 weeks ago by puredyne.

        Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

        #134970
        Member
        puredyne

          Seduzione Supersonica 😛 12 /// It is the first name that came to mind for the latest generation JBVA flagship DIY converter /// The ADC-boards support DSD & 32/768… the USB Digital Transport is capable of transporting 4 channels at 384 kHz (Extreme ADC and/or DAC)… Relly the top engineering with 55 MHz Audio Buffer and 180 MHz OPAMP for extreme musicality and detail FOR ADC and possible Line Driver for mass separation Synthesizers and Studio Gears. ADAT/TOSLINK should not work simultaneously (are methods to enable a maximum of 8+8 internal channels or 8+8 at 44-48 / 4+4 at 96 kHz via ADAT and 2+2 via TOSLINK/SPDIF, so at 192 kHz it will be possible with this DSP to enable only 2+2 external channels or only 4+4 internal channels at the state of the Japanese flagship art)… For Extreme Hi Performance quality it costs like toys in the shop (4 extreme Line Driver channels, 4 ADC channels, USB transport, Buffer, no power supplies). .. With a few decent power supplies it can already run… and upgrade in the future with ~7-9(up11-13 inc. Line Drivers) separated lines of very musical, pure, perfect and expensive HiEnd power supplies… preview photo of the digital transport+buffer to connect 4 internal I2S channels at 32/384 kHz (UAC2 ALSA Compliant).

          * Hello /// Love is Lost

          It seems to remember the DSP had another name in the past… we read the manual carefully ~10 years ago~ SS Run/reflected on the Gnu Linux Liquorix Audio users/producers/masters ~ do not take advantage of chronic fatigue symptom or chronic fatigue syndrome ~ is a typical widespread side effect of all those vaccinated with nanotechnology in the last 20-30 years ~ young people and lately the majority of adults are affected at various levels of chronic fatigue… to keep karma balanced in reality ~ Segretario Segreto is capable of overturning any shit advantageously ~ The nanotechnology contained in vaccines creates chemical/electrical nano networks globally/worldwide (GlobalmenteMondialmente) and with very little energy Segretario Segreto can go around the world in an instant passing through the home of all the tortured/masochists streaming slaves. Over 30% of Italians were scientifically recognized as electrosensitive… as a statistical reflection of patients already diagnosed with Fibromyalgia, Multiple Chemical Sensitivity and Electrosensitivity. In Italy, the most severe patients with Atlas Syndrome aka FSM, MCS and ME are granted disability and retirement. For environmental syndromes and EHS it was hidden by governments and the healthcare system. So EHS is directly linked to MCS. There are officially only very expensive biological medicines (all patients were subjected to underdosing and physical torture) for treatment. Decently Hi Performance Power Supplies, extremely pure Digital Transport and Flagship Converters are also a very powerful electromedical integrations for stabilizing the nervous system and it is really important to take this into consideration.

          So tend to be very careful never to mistake a patient’s extreme fatigue for delusion or insanity.

          If you have already read it, now the new ones will first have to read and understand exactly what is specified in the manual.
          First of all to be able to take the digital/analog purity performance to the extreme when connecting isolators and flagship converters.

          https://www.minidsp.com/images/documents/Product%20Brief-MCHStreamer.pdf
          https://www.minidsp.com/images/documents/MCHStreamer%20User%20Manual.pdf
          https://drive.google.com/file/d/1iWi6IK6nb6y5uAFNQWjUZjOi_6IvD0gH/view

          Cordialmente…

          • This reply was modified 1 month, 2 weeks ago by puredyne.
          • This reply was modified 1 month, 2 weeks ago by puredyne.
          • This reply was modified 1 month, 1 week ago by puredyne.

          Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

          #135234
          Member
          AVLinux

            @puredyne

            If you don’t like AV Linux that’s fine, it’s not everyone’s cup of tea, if you have a problem with systemd or elogind or Enlightenment also fine to be clear I’m not on the antiX forum because I have any opinion about inits I am completely apolitical about all of that stuff so I’m probably too boring to be here… To suggest that AV Linux is for “slow low sample rate people” is absolutely unfounded and ludicrous, the PipeWire setup on AV Linux has capability for any bit depth and sample rate (and therefore equal Audio fidelity) that classic JACK did and of course as always you are completely free to use Ardour or Mixbus or Reaper with their own ALSA backends and circumvent PipeWire completely and operate at the ALSA driver level at whatever sample rate and bit depth your Audio hardware supports.

            I don’t disagree at all that something as light and resource efficient as antiX could make a phenomenal Audio system and I’m sure many people do use it for that purpose, if you have put something together that works well and better for you than AV Linux or something else that’s great news! I fail to understand why people have to shit on other people’s work to justify something they did themselves…? If your system is good it’s good isn’t it? Does AV Linux or something else have to be crappy for your thing to be good?

            I see you’ve PM’d me, I’ll keep any discussion here please,

            Glen, AV Linux Maintainer

            #135238
            Member
            Robin

              Hi Glen

              To suggest that AV Linux is for “slow low sample rate people” is absolutely unfounded and ludicrous,

              Yes, I can confirm, @AVLinux, your reply is perfectly true. That only low sample rate stuff about pipewire is pure nonsens.

              Does AV Linux or something else have to be crappy for your thing to be good?

              Never! This is something which you should not think of antiX community. If somebody has stated this, it is an individual statement by somebody, not represenative for the community or antiX, as I have perceived it the last years. Sorry you’ve got this impression.

              Btw, antiX itself doesn’t get away any better in the posting you’ve quoted from, so please don’t mind:

              why would antiX be declared stable? Whatever you touch breaks…

              I guess somebody had to vent his frustration, and both, antiX and AVLinux have got theirs…

              I fail to understand why people have to shit on other people’s work to justify something they did themselves…?

              Signed.

              Yours
              Robin
              ———————–

              Edit:
              Btw, @AVLinux, Do you happen to know whether pipewire was optimised for 32 bit CPUs also, not merely for 64 bit, or is there another reason why it produces up to 55% CPU load on a 1,7GHz single core 32 bit CPU, while running on way weaker and slower 64 bit devices without such exorbitant CPU load rates? (These are readings from antiX 23.1 runit 32 bit with pipewire activated.) I guess I’ll have to give AVLinux a try on this 32 bit notebook. Are there 32bit ISOs available?

              • This reply was modified 1 month, 1 week ago by Robin. Reason: Edit

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

              #135250
              Member
              AVLinux

                Hi @Robin, thanks for the comments, I know the antiX community is full of great folks.. 🙂

                Unfortunately AV Linux is 64bit only, when I moved to MX-21/Debian Bullseye I had to make a tough choice to drop 32bit support mostly because less and less Audio and Video software and especially Audio Plugins were being developed for or tested on 32bit. Like many Linux projects I’m a lone spare timer and although I do receive some assistance from the MX developers and Packagers and of course I benefit from the wonderful antiX/MX ISO build toolchain the daily details of AVL are a one-man project so simply keeping a single ISO afloat is often more than I can manage..

                So to your question about PipeWire I have never used or tested it on a 32bit system but it would not surprise me if the PipeWire developers have not done a large amount of testing specifically on 32bit systems so they may not be benchmarking 32bit vs. 64bit regularly. Unfortunately all I have for you is conjecture..

                #135259
                Member
                puredyne

                  For state of the art audio refer only to pure ALSO. JACK & Pipe Wire abstraction layers can only corrupt the stream. All users can learn to hear the difference if they use passive monitors or HiEnd CMOY style headphone amplifiers. Decent streaming performance with Linux Audio required years of extreme hardware strategies.

                  No, AVLinux creates and maintains high-quality streaming consciousness (only in conjunction with WM of professional multimedia performance of ~4MB RAM or more liquid graphics engine… so iceWM or better… only with Wayland will heavy WM for RTLiquid-Audio work ). AVLinux users are completely respectable. We just need to clarify why for ~10 years AVLinux hasn’t created the only protocol for listening and mastering. Why did the first version released in 2023 work and the second not work with UAC2? Why don’t users complain about UAC2 not working… would this prove that all users are suffering from chronic fatigue?

                  If you start LIVE and/or install it you notice that the system does not turn on the audio device. For 10 years AVLinux only ran obsolete RME HDSP. In fact limited to 96 kHz to be able to monitor/sample enough channels. The 192 kHz technology has been discontinued for ~15 years due to excessive distortion of sampled waveforms.

                  And in any case it seems that the UAC2 liquid compatibility and performance problem has mainly affected Liquorix updates throughout 2023.

                  There is no other technology to monitor a DAW in real time to perfection or to certify and produce the master with the debian MPD and squeezebox packages.

                  It is a major problem to force users to monitor the DAW with obsolete scrap HDSP type hardware. Like it or not, it is a political problem because fascism is based on ignorance.

                  This thing needs to be investigated carefully. There is a suspicion that AVLinux (first version hidden already in the first month of 2023) and Harrison Mixbus 32C only work at 768 KHz with DIVA in Divine Mode. With AVLinux from early 2023 (three versions ago) and Harrison Mixbus. Should only one person be responsible for AVLinux consciousness? For every 10 hours of physical work, this person needs 10 days of relaxation. She should be paid minimum 3K for every 10 hours of experimentation. Everyone else seems to have much more serious chronic fatigue problems. You need to consume 50-100 euros of medicinal sativa sativa with an extreme excitatory effect every day to combat chronic fatigue.

                  Will we pay for the grass to work with? And what happens if you die? You can promise and guarantee a BIG BANG for everyone’s safety. If you have been in danger of dying every day for 8 years due to no immune system, there isn’t much time.

                  Coincidentally, for all these years AVLinux was opposing the medicinal effect of state-of-the-art streaming. The only thing that can actually keep you alive.

                  Maybe there’s an interesting variation going on. A famous scam blogger in the eurorack scene, released yesterday a new crap module teamed up with the famous audio ignorants Befaco. So Microsoft and Apple streaming ignorance is 100% slave of all Linux Audio users.

                  Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

                  #135312
                  Member
                  Robin

                    For state of the art audio refer only to pure ALSO. JACK & Pipe Wire abstraction layers can only corrupt the stream.

                    All users can learn to hear the difference if they use passive monitors

                    Perfectly true. May be to be added: complemented by linear amplifiers with low harmonic distortion and an adequate sound design of listening room. (and for sure, Neighbours next wall not being too noisy 🙂 )

                    About all the digital processing protocol and hardware related stuff and how to deal with this properly I can’t say or assess anything, having seen and run analogue studio equipment merely some twenty or thirty years ago. But what you state makes perfectly sense to me. You can hear the RFI noise from the default PC internals in audio output signals always. For consumers this is fine in most cases, since signal to disturbance ratio is sufficient in most cases, so average listener won’t notice there is something wrong with the signal. That changes if you aim towards professional audio remastering and editing, then the noise from PC is not acceptable, and actually you need a rt kernel. And yes, then it makes sense not to have any unneeded additional layer on top of pure alsa, which can (not necessarily must) cause the audio streams to break due to cpu is busy with some wayland/desktop-environment-management/pipewire/pulseaudio/or-whatever stuff in a critical moment, ruining the recording or remaster. (To have it perfecly, you’d need to have the PC merely doing all the graphical management tasks, steering a dedicated specialised external audio processing unit, which would solve not only the stream corruption issue when CPU is busy with other tasks, but also the RFI noise issue from the PC processing (at least as long it has a grounded metal case rather than a plastic cover). Having this external audio unit perfectly shielded and its digital inputs filtered for unwanted noises will also protect your signal against signal noise from the modern light bulb replacement at ceiling.

                    Concerning your Supersonic antiX edition: Unfortunately I don’t have all that fine studio level hardware stuff to make use of edition. And actually, I don’t need it either, since I’m not doing audio remastering or recording on a professional level. So for me antiX running with or without pipewire is fine both. You can do nice things with pipewire, there are things like easyeffects, or the graphical patchbay inherited from jack. Not sure whether the occasional crackling noises which the additional pulseaudo layer produces, happening always when CPU is busy elsewhere, can be overcome in a way the PC is still fit for default everyday tasks the PC usage is aimed to for most users. Have never observed these when running plain alsa.

                    Btw, since both of you seem to be experienced in digital audio processing, you may want to have a look at the filter setups in antiX accoustic colours, which is now included in antiX 23.1 . My knowledge about proper curves and chains is faint from long years not having made use of it. So probably there could be something improved with current and fresh know-how. This tool is explicitly not meant as as professional studio level audio processing for remastering or live recording, but as an easy to apply acoustic colour modificator for average PC users, adapting the sound ambiente according to the button pressed, just as you may find it in old tube radios. For checking it out, just run antiX 23.1 from a Live medium, there lives an entry in main menu. And please note, the filtering in the antiX 23.1 included version is not the very best; there is a more recent original (trixie) version 0.4 at my git, providing way better filter results, since in bookworm merely a backport could be applied without breaking core packages of current antiX.

                    Many thanks!

                    antiX Acoustic Colours

                    —————————
                    P.S.: Please, no personal insults and animosities here. Not against other forum members, nor against other distributions or respins. Otherwise you’ll probably notice the board mods like Brian @Masinick will take action.

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

                    #135370
                    Member
                    puredyne

                      The listener must be really asleep to buy the Reaper… the DEMO sounds terrible 384 khZ performance and miserable 16 BIT with plugins. Does the audio work when I pay? Reaper are the same developers as Winamp, that alone shows how much of an asshole they are and to stay away from that type of audio ignorance. We had tested 1024 sinc upsampling/rendering (11 years ago), the frequency response is fucked. Aurdour and Mixbus’ SRC import is much more consistent. Only the MIDI sequinccer for external instruments makes sense. At that point MUSE Sequencer is better, in realtime the audio playback for pre-listening is superior to Ardor. The DAW must be capable of recording, processing, playing and exporting 32 float streams. Over the last ~35 years, mainly DSD ADC-DAC chips have been produced. For the past 25 years only flagship DSD ADC-DAC. DAW export will need to be converted to DSD with CamillaDSP or something, to be certified for the native stream. 96-192 kHz is ridiculous failure in PCM for waveform warping and converted to the native DSD stream is poor. The ADC should be capable of sampling native DSD256 stereo 23Mbps for digitization of old Vinyl and Tapes or new improvisations and LIVE. In the new millennium PCM playback is discontinued and samplig is virtualizzed only (so 1 bit to PCM SRC integrated in the ADC chip), PCM it is a crazy limitation of computerized DSP technology. For state-of-the-art analog mastering the DAW must be exported to PCM and DSD playback calculated (with CamillaDSP or something similar). Direct DSD playback (without USB) only exists on 8ch RPi BUS Open Source. Rebel Corruption(ignorance) Corrupt fascist streaming of AES and so DSD Ravenna shit must be killed.

                      In the last 10 years the DAW monitoring and listening to audio-playback has only made sense with a dedicated and recognized PCIe-USB-Host for streaming (Buffalo/NEC/ASMEDIA). USB transport must be isolated and reclocked. Thus faithful monitor and playback only exists with AMANERO and extremely expensive isolation integration or popular low-cost I2SoverUSB. There is something commercial but how many Linux Audio users have given 2-3K to RME for two isolated 768 kHz ADC channels in the last 10 years? For that price it’s a total scam.

                      We can get 4 state-of-the-art ADC channels for ~500 euros, including DSP and extremely transparent balanced transmitters. the user must know how to assemble the converters or pay professional consultants. If I have little money, 2-channel modules can be purchased/added over time.

                      To resolve PC noise problems, PC Audio Grade strategies must be followed and studied. Use only Seasonic Hybrid 1000W 350 euro or linear power supply with Startech 20GBps USB card or any other ASMEDIA PCIe card. SSDs corrupt the stream with magnetic fields and power supply noise (it’s easier to use HDDs with a banal Chinese SATA filter costing 20 euros).

                      • This reply was modified 1 month, 1 week ago by puredyne.
                      • This reply was modified 1 month, 1 week ago by puredyne.

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                      #135419
                      Member
                      puredyne

                        It’s a common antiX with LQX + RTirq and some pre-installed audio package… The goal is to increase the operator’s productivity. So dissolve crises and wars…

                        *The most urgent thing was to remove the equalizer and volume control. To kill the obscene integrated pre-amplifier. No one could make or listen to music with that. Not even the Tube… Now the operator can listen to professional 64bit float DSP equalizers.

                        We have seen and tested patchbays for Pipe Wire and you can do everything better in realtime by skipping that layer with Jack(d)\\\

                        Is the power supply noisy and does the motherboard have miserable switching voltage regulators? There will be devastating RFI issues for the stream. SSD produces devastating RFI and EMI. CPUs above 65W seem to tend to break up the stream with noise. Maybe 35W is even better but worth trying and maybe it can only do small media players.

                        Sotm sells an audio grade motherboard for ~700 euros, chip set and intell cpu… always boycott intel to death.

                        All power supplies with RFI (noise) above 10mV destroy the stream.

                        We found an exceptional old stock motherboard for 93 euros. There is a load of transtors onboard to clean the RAM power supply… there are too many and this does not guarantee long life and in any case it should work for at least 10 years… so by installing 2 RAM Filers the CPU Noise cleaning effect is extreme . Low-power CPU, passive heatsink or CPU-FAN-Filter is also indispensable.

                        Attached is the photo of dead Zio Vania’s fused and brused motherboard (it doesn’t work anymore thanks to the audiophincs dot fr scam and the new one sounds much better)… RAM Filter detail… PDF file with SSD measurements

                        ** chaos added… Enigizer Orgon Systems is one of the best synthesizers in history… I’m interested because Leopoldo is directly connected… killed or induced to commit suicide because after 10 years he stopped consuming “roba¨. When Leo was still aged 18-20, someone had decided he should command the freetek scene e tribes in north-eastern Italy, and so on… AVeva viso e nasa come Asburgo/Mani…

                        *** The BUS Filter is very popular in the audio community. ~10 years ago it was common to install two with the sound card in the center. Noise only from 3.3V and 5V power supplies is filtered. So much improves the quality of standard optical and coaxial transport. The stream from an HDSP sounds much more relaxed, so focused. In recent years it was used a lot to clean the 3.3V power supply of USB-Host. In this case (20Gbps), only the USB clock for transport is improved. Convenient because it is inconvenient to add capacitors near the clock. It’s not easy. SATA Filter with SSD is not enough, a filter is essential to separate interference with PSU also at the input. SSD must be enclosed in aluminum or metal box(es) to block ~60cm of radiation.

                        • This reply was modified 1 month, 1 week ago by puredyne.
                        • This reply was modified 1 month, 1 week ago by puredyne.
                        • This reply was modified 1 month, 1 week ago by puredyne.

                        Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

                        #136378
                        Member
                        AlexUD

                          Hi guys!

                          this is my very first time posting here so please forgive me if my comment and questions are too stupid.

                          I’m pretty much a rookie here, been using Antix since 2021 on my laptop, for at 1.5G of RAM I can hardly use anything else, tried many distros but Antix is the only one that works smoothly. in my other PC I was running Windows 10, but with 4G of RAM that was SLOW, and the PC is the one I use for my freelancer efforts, which are 100% remote, I do some industrial and graphic design (I use FreeCAD and Blender), some video recording and editing (which doesn’t need to be such a great audio) and yes some musical arrangements for the which I use Musescore, but what I really need good audio for is for remote interpreting, which I used to do audio only but increasingly am demanded to do in video too.

                          That’s when I decided to leave Windows for it was too slow, and when interpreting for physicians or in courts and I have them speaking as fast as they possibly can, in noisy environments and too proud to offer repetitions I need to catch every word and do so very clearly, so in Windows, with my poor little RAM, in a video session both video and audio were choppy or laggy, so I moved to Linux too, initially took a chance with MX (For I love Antix), both XFCE and Fluxbox, honestly both were awesome when compared to Windows, I was even able to install newer versions of both freecad and Musecore because RAM was no longer an issue, and the PC was basically flying rather than running.

                          However, the audio was still laggy when interpreting, both in video and only audio (which didn’t happen on Windows), video was perfect, never choppy, automatically going for higher resolutions, that’s why I went from XFCE to Fluxbox, I thought going for an even lighter option was the answer, but the same kept on happening, now, let’s be clear, it was not horrible choppy or laggy, but again, I needed audio to be flawless and I just had to assume it wasn’t a RAM issue, and being pretty ignorant I researched online and read about audio in linux, and learned many things, for instance, my Antix laptop, with ALSA it was really easy to choose, for instance the ringing of the phone to be on the speakers and the actual call audio happen on my headset, but it was hard to do the same thing on MX, that’s when I learned about pipewire (I was only able to see one device instead of the 3 I have plugged but this is because audio was centralized in pipewire), I tried managing it’s settings but it was clear I had no idea what I was doing, so I simply uninstalled it and I saw great improvement, let’s say, with pipewire I had 80% of the quality I expected on my calls, but without pipewire it went up to 90%, so my simple minded and probably miope conclusion is pipewire=bad.

                          then trying to polish it a little more I was pondering about using a low latency kernel but while doing that research I found AV Linux already with Liquorix, so I gave version 21 a try and WOW, that’s what I’m using now, out of the box it was 95% what I wanted, all I needed to do was disabling XFCE’s compositor (learned about that while researching DE’s for Antix that they offer a XFCE with no compositor) and it is now PERFECT. It has basically every simple piece of software I use already included except freecad and the server for my remote mouse and keyboard, so this distro is not only for audio or musical people, it can be a SERIOUS option for remote workers who video conference a lot and for designers too (AV’s creator does say it is if for creators and he means it).

                          but now I worry because tinkering with XFCE was really simple (performed all automation and efficiency tasks I needed in few minutes and even had the time to make it look nice, not that it needed much work,the orange colors look great, it was almost the exact palette I was using on my Antix JWM) but I read Enlightenment isn’t, however, being a lighter option I was also excited about the change, but really what worries me more is going back to pipewire (which in AV21 was optional and not default), I feel my system is perfect as it is now, I am going to purchase a new bigger PC and I will be trying AV23 but I am afraid I will have to thinker much more with that one and may not get the same flawless results.

                          now why don’t I use Antix for my needs? I mean, pure ALSA is cool, however there are some things I don’t know how to do, I understand some of my issues can be fixed by writing a script (which I don’t really know how to do, again, I’m pretty much a guy who is just getting my toe tips wet in the Linux sea), but every time I restart my laptop I have to, for instance, set my headset’s volume again, because the preferences are not saved, or also I don’t know yet (this I did try) how to have some programs starting at startup, I mean I know Antix totally satisfies my audio and speed needs or rather it goes well beyond my needs, but I can’t yet perform all the automation I usually use and that is so easy to do with AV (and MX) but then, with Antix, the only issue is comfort, like I have to set my speakers preference every day, or for instance, after a power outage, setting my clock again (not always, just some times) which is rather simple but those are annoying steps I don’t have to walk with AV.

                          So it is pure laziness, me not using Antix as my job drive.

                          now I might be that pipewire is so awesome and that’s why lots of people in the comments in distrowatch were like “pipewire is THE way to go” and AV21 I even read criticism for not having pipewire by default, but that was the cherry on the cake that made me try it in the first place, basically MX but no pipewire and Liquorix and many other perks for content creators, sounded perfect, maybe I’m just too stupid to properly use pipewire, maybe I need to study more how to properly use Antix, I mean my laptop running JWM Antix, I just use it for web browsing and writing documents or spreadsheets, and remote document storage, pretty sure it has lots of potential that I have yet to tap into, but the learning curve seems steeper for me, previously I had used Fedora (leanred it in school but never really used it),tried Mint and Zorin, all very user friendly IMO but not nearly as lean as I needed them to be, tried linux lite, puppy, lubuntu, but those had driver issues and they are not as light really, LXLE and Antix were the only options that really worked on my laptop, but Antix is superior.

                          That’s all the experience I have when it comes to linux distros. So my questions:

                          to those who already tried AV23 and getting an idea of my inexperience level, do you think I’ll have trouble getting it up and running?
                          has anyone used AV for video conferencing? or MX even? any one experienced laggy choppy audio? if so, was it pipewire? any other thing that might have been the problem if not pipewire?
                          maybe you need more details, the PC is Atom D525 1.8Ghz and 4 of RAM, running AV21 smoothly but never got the audio right with MX23, my internet conection is 300Mbps (both upload and download), ping is 6ms and jitter is 3ms, so I guess my internet service is pretty good. MY whole system, while on a video call uses 1.5G of RAM so again, I think RAM is not the issue. Maybe my processor but nothing I can do about that one, and the situation did improve with no pipewire.

                          my last doubt is, about my laptop, maybe I’m doing something totally wrong in Antix, some basic stupid mistake and that’s why I have to arrange my sound volume every day or I can’t make my remote keyboard’s local client to start at startup, what I tried was control center > desktop > edit jwm settings > startup and tried adding my Unified Remote Server to the list. The other thing I tried was control center > system > choose startup services > add, but can’t get the service to show in the list of items to add.

                          not sure either how I should fix the volume of my speakers and headset so as to not having to adjust it each session. Maybe I need to install pulseaudio? besides the alsa mixer. Not sure, I might be speaking nonesense. Antix base is installed in my hard drive, with runit.

                          sorry for such a lengthy post, and thank you everyone developing both Antix/MX and AVLinux and the people in these forums, I did lose my job because of covid and needed to totally change career path and doing so online for the first time ever and could not possibly have done it without you guys, with my crappy devices and rusty IT knowledge I needed a miracle to start working remotely and you guys are to thank for that miracle.

                          #136380
                          Member
                          PPC

                            antiX always uses the same default audio level, that is set in desktop-session.conf.
                            To edit that value: antiX menu > Control Centre > Session Tab > “Green rocket” icon > “desktop-session.conf” > Look for the “STARTUP_SOUND_LEVEL=” around line nr 107 and change the value to the one you want. Save the file. when you restart, the volume should always be at the level that you set it in, on that config file.

                            P.

                            #136386
                            Member
                            Robin

                              how to have some programs starting at startup,

                              Just add them to ~/.desktop-session/startup file, and make sure to append an ampersand to make a coprocess from them so the script can continue with the next startup tasks.

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

                              #136391
                              Member
                              puredyne

                                No, no, no miracle for you… The sound card has been an obsolete and scrappable concept for over 10 years. For audio, USB is used only as a transport with modern PC mass separation strategies. Telephone and musescore is consumer audio. They already work with the onboard codec. For audio applications all laptops, iMacs and similar integrated devices can be scrapped… For the techniques already explained above. The audio user must use an Audio Grade PC if I’m too slacker to build it, I will be able to buy ready made for 7K online. Normal laptops and normal PCs are only good for lo-fi communication (audio stream is broken). Creating a system capable of making Liquid Audio Player is the most difficult job in the IT sector, it is only for experts and interested in listening and producing/mastering music.

                                Preview photo of control panel in eurorack format for Analog To Digital Domain.

                                We are recycling (French) ACX Synth LINE DRIVER front panels. 4+4 modules will be joined together with rear shield plate (6HP 290mm *8=2320mm width) With just 4x 384 kHz channels you can record a session of eight already calibrated volumes/attenuation but the switch on the black panel also works as MUTE and permits the insertion on the FLY of infinite input voices.

                                Here there are no OUTs in the panel! The output is behind on the Line Driver Board and goes directly to the 4 ADC channels. The Red TRS is Direct BAL Input and excludes the UNBAL minijack and attenuator by inserting a Balanced JACK. It will be possible to connect BAL HiEnd Attenuator modules to the RED… So the black has the internal BAL Output of both panels on a MUTE/SWAP Switch and only one TRS-Jack Input is always BAL/Attenuator…

                                *Flagship British/Japanese Engineering ADC are capable of 32 bit 758 kHz and DSD256. It is a current limit of the American DSP 4 channels 32/384 ADCs.

                                Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

                                #136400
                                Member
                                puredyne

                                  To learn to listen to music I will have to deactivate pipewire, remove the antiX equalizer package. The volume control should automatically disappear from the taskbar.

                                  If the operator uses decent or professional DSP over USB, ALSAMIXER should appear as it does in the snapshot.

                                  Just do it…

                                  Avanguardia Tempo Reale ~ VÄRÏÄ ~23Mbps~ (((STËRËÖ))) 🔊 🗲ëdüzïönᶒ 🗲üpërsönïcᶏ 🔊 ~ ⏻ 🗲ïstëmᶏ ☉përätïv☉ ⏻ ~☽👑🦅☼ ᵵᶚᶆᵽ☉ 𝖗ᶚᶏᶅᶒ ☼🦅👑☾

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