From brent at keycorner.org Sat Mar 1 18:43:09 2014 From: brent at keycorner.org (Brent Busby) Date: Sat, 1 Mar 2014 12:43:09 -0600 (CST) Subject: [LAU] Dithering...should we dither about it? Message-ID: I've been using triangle dithering when I export at 16-bit in Ardour. Not to start a religious war or anything, but I'd actually be interested in hearing opinions about why that or some other alternative might be better (shaped noise, etc.). How do you usually dither? Does it make a difference? -- + Brent A. Busby + "We've all heard that a million monkeys + Sr. UNIX Systems Admin + banging on a million typewriters will + University of Chicago + eventually reproduce the entire works of + James Franck Institute + Shakespeare. Now, thanks to the Internet, + Materials Research Ctr + we know this is not true." -Robert Wilensky From harryhaaren at gmail.com Sat Mar 1 18:51:40 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sat, 1 Mar 2014 18:51:40 +0000 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: Message-ID: On Sat, Mar 1, 2014 at 6:43 PM, Brent Busby wrote: > How do you usually dither? Does it make a difference? I'll just throw this up, its a good overview of dither types, and *why* it makes a cleaner mix (for humans): http://wiki.xiph.org/Videos/Digital_Show_and_Tell#Dither (that's the accompanying document for this video: http://www.xiph.org/video/vid2.shtml ) I generally export to 32bit float .flac... so no dithering (or burning to CD's :) HTH, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From aiyumi.br at gmail.com Sun Mar 2 01:16:33 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Sat, 1 Mar 2014 22:16:33 -0300 Subject: [LAU] Music Editors for Blind Users? Message-ID: Hi! A visually impaired Linux user here. I recently came across this great thread[1] and decided to write this message. My goal is a little different from the original topic author (generating MIDI via code VS. recording via a keyboard) so I decided to start a new discussion. Here's my situation (warning: very long post!): First of all, I'm just an amateur and a beginner with nearly zero knowledge about music production. I had very little contact with DAWs, even in Windows... The closest I've gotten to a "DAW" that I tried to use was Anvil Studio (many years ago), which wasn't accessible either. The little that I know about MIDI editors and DAWs is from what I read looking for such programs in Linux. I'm using Slackware Linux, with Speakup[2] as my console screen reader (with Espeak[3] through the Espeakup[4] connector, because I don't have a hardware voice synth), and Orca[5] as the screen reader for X (note: I use only ALSA, without Pulseaudio). I'm trying to create some kind of audio setup for quite some time (since around 2010). Back then, I read everywhere that it was needed to recompile the kernel with a realtime patch. Then I found the courage and did just that, but: * I never got JACK to really work. It complained that the soundcard was already being used, probably because of my screen readers which are softsynths (again, I don't have a hardware synth nor a Braille display), and I suspect they take all the soundcard for themselves. JACK only got to start if I used a second soundcard, an M-Audio Fast Track Pro (which I bought because it seemed to work on Linux), but it didn't go too well either. Only much later (last year, 2013) I came across this post http://joegiampaoli.blogspot.com/2011/06/m-audio-fast-track-pro-for-debian-linux.html It seems the Fast Track Pro is already supported on newer (3.X) kernels, but I still didn't test that. * Most audio related apps were (and still are) made with QT GUIs. Though QT's accessibility API exists for a long time (and is well documented[6]), there were no means for Orca to communicate with it yet, so all those apps were completely inaccessible. * With the recompiled kernel, my video driver installation broke (I wasn't even using proprietary drivers or anything). X only showed a blinking cursor on the screen. Orca worked just fine, but it was a problem when a sighted person needed to use the computer. So, I went back to the stock kernel (where the video driver worked) and gave up, for the most part. I settled on doing everything from my keyboard (which requires some sighted assistance), then sending the result as a wav file to my computer, then processing it with SoX[7] (which I use all the time) and/or Ecasound[8]. But recently I began thinking of using sounds from softsynths or other programs (for example, LinuxSampler[9]) instead of my keyboard's internal sounds, and for that, I think it would be best if I could record MIDI directly into my computer. So, the search started again, and I began testing some command line applications (most of which were already mentioned on the thread in [1]): * CuSe[10]: stands for "Cursed Sequencer", and like the name suggests, is Curses based. Interestingly, it was made specifically with visually impaired users in mind. Some features look interesting, like punch in/out, a step sequencer, and (the most interesting of all) a "remote control" like thing, where it's possible to configure key combinations for controlling the software through the hardware synth's keyboard (avoiding the need to run back and forth between the synth and the computer). But it doesn't seem to have some features I want, like adding/editing lyrics to the notes, MIDI CCs and SysEx events. Unfortunately, I was never able to get any sound from it (be it through FluidSynth or my MIDI keyboard). * Midish[11]: already mentioned on the thread in [1], it supports recording and playing MIDI, adding events and SysEx messages, and it has nice filtering features, making it possible to split the keyboard or transform one event into another. It only supports raw MIDI ports and it took me a lot of time to figure that to make it work, I needed to "modprobe snd_virmidi", then (a)connect the virmidi ports to my synth's ports, then configure the inputs and outputs via the "~/.midishrc" file. It was the program I got working more successfully, and I placed my hopes on Midish for a long time, reading and rereading the manual to see if I could make it do what I wanted. I was able to record MIDI via my keyboard (though I wish there was a "remote control" feature or something similar). However, sometimes I also wanted to insert notes manually, and as already pointed out in this message from the thread in [1] http://lists.linuxaudio.org/pipermail/linux-audio-user/2014-February/096120.html that isn't the most intuitive task, and Midish doesn't seem to let me add lyrics or edit notes (only cut/copy/paste measures)... Well, unless I save the song to Midish's format and open it in a text editor, but I don't want to have to memorize which note numbers are which :P. Or unless I'm missing something else. * Midiedit[12]: contrary to Midish, this only edits and doesn't record. It's contained in only one file and is a Perl script. It has two interfaces: a Curses based one and a "text editor" based one. With the Curses interface, it can playback MIDI files (I used Aconnect to hook its ports to FluidSynth and the sounds played), and as the name suggests, edit. Events (notes, CCs, program changes etc.) can be inserted, edited and deleted. They're displayed sequentially in a list, one per line, with their starting position in ticks, duration and channel. They can be navigated by using the Up and Down arrows, then edited (the bottom screen shows the available actions and which keys do what). One thing I love about it is that it plays back the notes when the cursor focuses on them. Another thing is that it treats each note as a single event, by matching the note ons with their respective note offs, which makes managing notes much easier. So far so good, but everything always has some "cons". I wish it could mute or solo tracks, and have a separate view for each track. The search function can be used to jump straight to the notes from a specific channel, and cycle through the results with "n" and "N" (similar to Vi/Vim), but the notes from the other tracks are still there and it gets confusing easily (at least for me). It doesn't seem to have an action implemented on the interface to handle lyrics and text events, but these still can be edited through the "text editor" interface, which is called "dump mode". In this mode, it opens your text editor of choice (the one in the "$EDITOR" environment variable) and represents the MIDI file's contents as readable Perl code (based on the dump output from the MIDI::Perl[13] module). All events (including text events, lyrics and SysEx) can be edited this way, (although obviously it doesn't play back the notes, and we don't have the luxury of the unification of note ons and offs). After saving and closing the editor, the Perl code is processed and the content is written back to the MIDI file. * From the same author of Midiedit, there are other interesting Curses based MIDI tools[14], like Midifade[15], which generates "sliders" for controlling MIDI events through the PC keyboard (on the command line, we specify how many sliders to generate, the ALSA port, the MIDI channel and which event each slider will control), and can, for example, change Ecasound's controller parameters, or the "knob" parameters from a softsynth. That's it for the CLI apps I tested. Midish and Midiedit are the ones I had most "luck" with, but my attempts with them weren't that productive either. My search for an accessible MIDI editor or DAW continues. Recently, I updated Slackware to 14.1. I'm using the stock kernel, no realtime patches or anything yet. First I want to know if I really need it (I read that because of changes in recent kernels, RT patches aren't strictly necessary anymore), and if there's some app that I'm able to use (if not, there's no point to even try compiling a realtime kernel :P). I finally got the QT-AT-SPI[16] bridge to compile, meaning I can access QT GUI apps to some degree now. So, I decided to test some of the most popular Linux DAWs, those QT based audio apps that taunted me since I first read about audio production on Linux (basically, the only options that were available and which I couldn't use). >From the list of features I read, the one that appealed to me the most was MusE[17]: * The possibility of mixing MIDI with audio (samples, loops, vocals etc.). * Instrument definition files for various brands of keyboards that ease the use of those synthesizers, making it so that we don't need to memorize patch names/numbers and controller numbers. * MIDI recording. * Editing of all types of MIDI events. Besides the visual editor, there's a "event list editor" (I suppose it's something like in Midiedit), so maybe it could work for me. * A few internal instruments (some synths and a FluidSynth frontend). * DSSI support. * (apparently) native VST support too (and I assume, a few Windows VSTs through DSSI-VST). * And the "remote control" feature :D! Of course there's the multitude of functions that depend on the mouse (which I can't use), but it seems to have customizable shortcuts. It's a graphical application, obviously very far from being text-based, but I read that it also has some Python bindings, so maybe a few things can be scripted...? However, sadly, all isn't as good as it sounds. I installed MusE 2.1.2 and opened the interface. The menus work fine, but the screen reader has problems reporting the interface elements (for example, it says "spinbox" or "checkbox" but doesn't specify wish item that control corresponds to. The "MusE Settings -> Global Settings" screen is full of these examples and unreadable through the screen reader). I just took a quick "look" at the interface to have an idea of what the screen reader could read. I didn't test recording or anything yet, mainly because of the aforementioned problem with JACK, nor could sort out the setup part about "/dev/rtc"... While we're on the subject of MusE, a question for the "MusErs": if I don't use JACK (or use the "-a" switch), won't MusE ever play anything through the PC's speakers (imported audio, internal instruments, FluidSynth soundfont sounds)? Or will it just be unable to communicate with JACK aware applications? I'm thinking if I'll insist on MusE for a bit more, and if I do, maybe I'll contact the developers about the accessibility issues. Based on my limited knowledge of MIDI editors and DAWs which I acquired from what I've tested or read about so far, desired features for my dream DAW would be: * Curses based interface. * Both JACK and some form of ALSA Seq support (even if it's only with raw MIDI ports, or some workaround involving sending commands through Amidi :P). JACK would be for more complicated setups. ALSA would be for when I only want to do some quick MIDI editing and don't want to use a second soundcard just because of JACK (I need my screen reader, so I can't turn it off to free the first card). * Playback of MIDI files. * Support of a metronome click, and a "count in" feature for recording. * A "remote control" feature, or a way to start recording (or the count in) only when I press a key on my hardware instrument. * Each note shown as only one event, by matching the note on with it's respective note off. This way we don't need to struggle scrolling through the events trying to match the ons and offs. * Possibility of inserting/editing/deleting all types of events, including text events, lyrics and SysEx messages. * A way to select a track to be "active". * A separate list of MIDI events for each track (only showing the contents of the "active" track), where events can be browsed with arrow keys and be edited or deleted. Also, the possibility of inserting new events through the interface (without using the MIDI keyboard). * Playback of notes when the cursor focuses on them. * Ability to mute and solo tracks. * Support for instrument definition files, to ease the use of hardware synths (maybe some softsynths too). With that, I'd be able to select the patches and controllers specific to my hardware synth directly from the computer and with help of my screen reader, reducing the need of asking for sighted assistance to know the patch names. * DSSI support, or the possibility of interacting with some command line based DSSI host. * The ability to assign the MIDI channel and an instrument (ALSA/JACK/whatever ports and patch names) to each track (or is this a work better suited to a "patchbay" like app?). * Rendering of MIDI tracks to audio with the used synths' sounds. * Customizable keyboard shortcuts. * Some form of scripting support, like Perl or Python or Lua bindings, and the ability to assign user scripted functions to some keybinding in the editor. Some more complex features: * The ability to have both MIDI and audio in the same project. The audio doesn't need to be editable (just "importable"), only needs to have its starting position adjustable to be in sync with the MIDI tracks (triggered via some MIDI event maybe?). * Recording of MIDI while playing back the audio tracks, keeping them in sync. * Maybe a pattern editor, where the lists of events become a phrase, chains of phrases make patterns and chains of patterns make a song (but maybe it's too complex and is a discussion for another day :P). And that was my utopic idea for an accessible Linux DAW. I don't know how hard it is to implement all of that, specially the "MIDI alongside audio" part. Unfortunately I understand nothing about audio programming (honestly, I already tried :P) and very little about music production, so a lot of features I pointed out might be complete nonsense, or might be accomplished in a much better way (if anyone knows a way, preferably via command line, please let me know). As for the "patchbay" (with "jack_connect" etc.), I think even a simple Shell Script with Dialog based menus would be fine. As for post processing and mixing (applying LADSPA/LV2 effects etc.), it can be left to some other app (Ecasound for example) after all tracks are rendered to audio. And last but not least, on the thread in [1], I found Teqqer[18]. I couldn't test it yet because of the JACK problem (I seriously need to sort that out), but I took a look at the default config variables in the Python source code, and it looks like scripting support and customizable shortcuts are mostly taken care of. From what I understand from the shortcuts, the interface for inserting/editing MIDI events is a bit similar to Midiedit (Up/Down to cycle through events, although note ons and offs are separate) and it seems to have one view for each track (Left/Right to cycle through tracks), and it even supports patterns! :D So, minus the recording part (from what I understood, it wasn't made for that), the lack of ALSA (without JACK) support, I think it has the potential to become something like what I'm looking for (plus, I always wanted to try a tracker, but never found a "screen reader friendly" one). I'll definitely be watching Teqqer to see where it goes! While my dream DAW doesn't come true, does anyone know a command line way to record MIDI while simultaneously playing an audio file and keeping both in sync (even if it involves JACK)? I think Ecasound can be setup to do the opposite, playing back a MIDI file while recording audio (but it doesn't support MIDI recording, just playback... Or does it? :P). Well, that's it. Sorry for the overly long post. Thanks for your time! [1]: http://lists.linuxaudio.org/pipermail/linux-audio-user/2014-January/095910.html [2]: http://www.linux-speakup.org/ [3]: http://espeak.sourceforge.net/ [4]: https://github.com/williamh/espeakup [5]: https://wiki.gnome.org/Projects/Orca [6]: http://qt-project.org/doc/qt-4.8/accessible.html [7]: http://sox.sourceforge.net/ [8]: http://nosignal.fi/ecasound/ [9]: http://www.linuxsampler.org/ [10]: http://pi4.informatik.uni-mannheim.de/~haensel/cuse/index_en.html [11]: http://www.midish.org/ [12]: http://www.pjb.com.au/midi/midiedit.html [13]: http://search.cpan.org/perldoc?MIDI [14]: http://www.pjb.com.au/midi/index.html [15]: http://www.pjb.com.au/midi/midifade.html [16]: http://projects.kde.org/qtatspi [17]: http://www.muse-sequencer.org/ [18]: https://github.com/fps/teqqer -- ____________________ Blog: http://aiyumi.warpstar.net/ From gnome at hawaii.rr.com Sun Mar 2 02:43:40 2014 From: gnome at hawaii.rr.com (david) Date: Sat, 01 Mar 2014 16:43:40 -1000 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: Message-ID: <53129ADC.20806@hawaii.rr.com> On 03/01/2014 08:51 AM, Harry van Haaren wrote: > On Sat, Mar 1, 2014 at 6:43 PM, Brent Busby > wrote: > > How do you usually dither? Does it make a difference? > > I'll just throw this up, its a good overview of dither types, and *why* > it makes a cleaner mix (for humans): > http://wiki.xiph.org/Videos/Digital_Show_and_Tell#Dither > > (that's the accompanying document for this video: > http://www.xiph.org/video/vid2.shtml ) > > I generally export to 32bit float .flac... so no dithering (or burning > to CD's :) Hmm, I thought FLAC only did 24-bit??? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From harryhaaren at gmail.com Sun Mar 2 03:38:07 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sun, 2 Mar 2014 03:38:07 +0000 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: <53129ADC.20806@hawaii.rr.com> References: <53129ADC.20806@hawaii.rr.com> Message-ID: On Sun, Mar 2, 2014 at 2:43 AM, david wrote: > On 03/01/2014 08:51 AM, Harry van Haaren wrote: >> I generally export to 32bit float .flac... so no dithering (or burning >> to CD's :) > Hmm, I thought FLAC only did 24-bit??? I think the FLAC spec says it will handle anything from 4-32 bit-depth: https://xiph.org/flac/faq.html#general__samples That said, Audacity only has FLAC export options of 16 & 24 bit depths. Ardour supports 8, 16 and 24. Still no 32 bit float support (at application level). I should correct my previous statement though: I *thought* I exported 32bit float: but it turns out they're 24bits (from Ardour3), dithering set to None. And cropping the resulting output in Audacity and exporting was to 16-bit PCM, so I was actually doing this all wrong (no dithering, 32 -> 24 -> 16). A better workflow would be to: A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity 16bit in, crop, 16bit out B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, crop, export 16bit (with dither). The important part being to not dither twice, since then you'll be adding noise to the signal twice! I'll be using option A above from now on I think, since it involves less bit-depth changes. Living and learning :) -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sun Mar 2 03:47:22 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 02 Mar 2014 04:47:22 +0100 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: <53129ADC.20806@hawaii.rr.com> Message-ID: <1393732042.596.155.camel@archlinux> On Sun, 2014-03-02 at 03:38 +0000, Harry van Haaren wrote: > A better workflow would be to: > > A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity 16bit > in, crop, 16bit out > > B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, > crop, export 16bit (with dither). > > > The important part being to not dither twice, since then you'll be > adding noise to the signal twice! > > > > I'll be using option A above from now on I think, since it involves > less bit-depth changes. > > Living and learning :) When you need to add noise, consider to use proprietary noise :S. -------- Oops,I used the wrong account: Forwarded Message -------- From: Ralf Mardorf To: linux-audio-user at lists.linuxaudio.org Subject: Re: [LAU] Dithering...should we dither about it? Date: Sun, 02 Mar 2014 00:50:58 +0100 Mailer: Evolution 3.10.4 On Sat, 2014-03-01 at 18:51 +0000, Harry van Haaren wrote: > I generally export to 32bit float .flac... so no dithering (or burning > to CD's :) That's good. I keep my recordings as 48 KHz 32bit float PCMs, but I would be willing to use FLAC too, assumed somebody would be interested in my recordings and I would be willing to share them. However, what is dithering ;)? What are CDs? I remember that a long, long time ago, when [... long story ...] there were special "noise algorithms" available. It wasn't just steady noise by what ever waveform. IIRC it was noise, that only was added to low audio signal levels. Non-free-open-source-noise ;). From paul at linuxaudiosystems.com Sun Mar 2 03:51:16 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sat, 1 Mar 2014 22:51:16 -0500 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: Your post was very long and my response will not directly address what you wrote. I am quoting something I wrote a couple of years ago about interfaces for this blind people. It makes a single but I think critical point. ------------- years ago someone paid me to do a text-based UI for ardour. it was centered on very efficient use of the keyboard and using a screen-reader. the code probably still exists. i don't think it was very successful, partly for the reasons identified in the text you sent. but i think there is a more important reason. working with audio tends to involve the use of the screen to act as a kind of memory. there are a ton of parameters in play, and its a huge barrier if you constantly need to remember what they are all set to. the 2d expanse of the screen represents a kind of 2nd level cache of this information, where a sighted person can simply glance around and discover what they need to know about the current state of things. reproducing this functionality without the information-dense medium that the screen represents is a HUGE challenge. i've thought about it on and off every since the "ksi" interface for ardour was done. i have no ideas on how anyone could make progress on this. i think its a very interesting, very, very hard problem. i have no time to work on it. as a practical note, if someone wants to do something like this, it would obviously be quite likely that basing their efforts on an open source tool is likely to offer a lot of possibilities that are simply not available when using closed source tools. ----------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Sun Mar 2 04:18:59 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sat, 1 Mar 2014 23:18:59 -0500 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: <53129ADC.20806@hawaii.rr.com> Message-ID: On Sat, Mar 1, 2014 at 10:38 PM, Harry van Haaren wrote: > > I think the FLAC spec says it will handle anything from 4-32 bit-depth: > https://xiph.org/flac/faq.html#general__samples > flac cannot handle 32 bit *floating point*. the link you provided reads: "FLAC supports linear PCM samples with a resolution between 4 and 32 bits per sample. FLAC does not support floating point samples" That said, Audacity only has FLAC export options of 16 & 24 bit depths. > Ardour supports 8, 16 and 24. Still no 32 bit float support (at application > level). > Ardour supports 32 bit float export to file formats that allow it. FLAC does not. Note that technically WAV does not allow 32 bit float. You really need "WAVEX" for that. Sensible audio file I/O libraries can handle it, however, even though technically it violates the WAV specification. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sun Mar 2 04:31:22 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 02 Mar 2014 05:31:22 +0100 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: <1393734682.596.171.camel@archlinux> On Sat, 2014-03-01 at 22:51 -0500, Paul Davis wrote: > working with audio tends to involve the use of the screen to act as a > kind of memory. there are a ton of parameters in play, and its a huge > barrier if you constantly need to remember what they are all set to. True for most of us, who have the ability to verify memory by taking a look to a screen or notes written on a paper. The mnemonics of blind audio engineers likely will work a little bit different. I guess the "weak" point is "editing". When I was young we used analog recording and IMO the missing abilities to edit audio productions wasn't a disadvantage. I prefer music recordings without editing. To push a record and stop key can be done by blind and seeing engineers. Mixing is another "issue". I'm not blind, but I anyway prefer an analog mixer over digital mixing with a mouse. Is there really the need for blind (and seeing) engineers to have all the options that a GUI is able to provide? From gnome at hawaii.rr.com Sun Mar 2 05:38:21 2014 From: gnome at hawaii.rr.com (david) Date: Sat, 01 Mar 2014 19:38:21 -1000 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: <5312C3CD.2090506@hawaii.rr.com> Yet we have a member of the list who is blind and works with audio ... On 03/01/2014 05:51 PM, Paul Davis wrote: > Your post was very long and my response will not directly address what > you wrote. I am quoting something I wrote a couple of years ago about > interfaces for this blind people. It makes a single but I think critical > point. > > ------------- > > years ago someone paid me to do a text-based UI for ardour. it was > centered on very efficient use of the keyboard and using a screen-reader. > > the code probably still exists. i don't think it was very successful, > partly for the reasons identified in the text you sent. but i think > there is a more important reason. > > working with audio tends to involve the use of the screen to act as a > kind of memory. there are a ton of parameters in play, and its a huge > barrier if you constantly need to remember what they are all set to. the > 2d expanse of the screen represents a kind of 2nd level cache of this > information, where a sighted person can simply glance around and > discover what they need to know about the current state of things. > > reproducing this functionality without the information-dense medium that > the screen represents is a HUGE challenge. i've thought about it on and > off every since the "ksi" interface for ardour was done. i have no ideas > on how anyone could make progress on this. i think its a very > interesting, very, very hard problem. i have no time to work on it. > > as a practical note, if someone wants to do something like this, it > would obviously be quite likely that basing their efforts on an open > source tool is likely to offer a lot of possibilities that are simply > not available when using closed source tools. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From len at ovenwerks.net Sun Mar 2 05:36:14 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 1 Mar 2014 21:36:14 -0800 (PST) Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: On Sat, 1 Mar 2014, Aiyumi Moriya wrote: > A visually impaired Linux user here. I recently came across this great > thread[1] and decided to write this message. My goal is a little > different from the original topic author (generating MIDI via code VS. > recording via a keyboard) so I decided to start a new discussion. > Here's my situation (warning: very long post!): OK > > I'm using Slackware Linux, with Speakup[2] as my console screen reader > (with Espeak[3] through the Espeakup[4] connector, because I don't > have a hardware voice synth), and Orca[5] as the screen reader for X > (note: I use only ALSA, without Pulseaudio). As you require the audio reading tools, you need to do one of two things... use two audio devices one for the screen reader and one for audio i/o and jack. Or use pulse and jackd(bus) bridged together. Lots of people don't like pulse for political/personal/performance reasons. It seems pulse just ignores dropouts in audio, but that probably doesn't matter in your case as you would only use it so your screen could talk to you. Pulse uses about twice the cpu that jack does at any latency setting (so far as I can tell on my machine). Sorry, there is a third option. You can create an alsa device that is really just connected to a jack port. This uses less cpu than pulse. You have to have all this set up before the screen reader starts. You will want to tell the system this is the default audio card. I haven't used this method because I just use pulse the few times I need the functionality and turn it off otherwise. But you may be able to set the extra alsa device before hand. Others on the list use this method all the time and can tell you better than I. > I'm trying to create some kind of audio setup for quite some time > (since around 2010). Back then, I read everywhere that it was needed > to recompile the kernel with a realtime patch. Then I found the > courage and did just that, but: Not so sure about slackware's stock kernels, but the patch is not really needed so long as the preempt switch is on. In ubuntu it is the -lowlatency kernel and uses all the same video drivers as stock. Slackware may have something similar, but even if not, rolling your own kernel on the same source tree with no patch but the option changed should work with the stock video too. You may also be able to use a command line switch at system start to turn this on at boot time. It would be possible to have two grub menu options one to start with it enabled and another without... quite honestly I run with it turned on all the time just fine so I would just default it that way. It is important no matter what kernel you use to have no USB ports sharing interupts with your sound card. > * Each note shown as only one event, by matching the note on with it's > respective note off. This way we don't need to struggle scrolling > through the events trying to match the ons and offs. All the gui based midi editors do this, so it is not an unreasonable request. Notes have a start time and length rather than a start time and an end time. > * Possibility of inserting/editing/deleting all types of events, > including text events, lyrics and SysEx messages. Just work to add it. But remember most authors do this stuff as a hobby too. Same with most of the things you want. > Some more complex features: > * The ability to have both MIDI and audio in the same project. The > audio doesn't need to be editable (just "importable"), only needs to > have its starting position adjustable to be in sync with the MIDI > tracks (triggered via some MIDI event maybe?). > * Recording of MIDI while playing back the audio tracks, keeping them in sync. Nama works well with midish and plays them synced. I don't think it triggers from midi events, but uses the same transport. triggering audio to midi events is really extended sample playback, so one of the sample players may do this already. nana's web page is a bit old and the author has done more work than the web page would sugest. The web page is 2009, but the latest update is 2013. > production, so a lot of features I pointed out might be complete > nonsense, or might be accomplished in a much better way (if anyone > knows a way, preferably via command line, please let me know). There are some other blind people on here who do wonderful production from commandline. It can be done. I think Julien ( http://juliencoder.de/nama/ ) does mostly keyboard to audio using nama, but some of his stuff uses midi recording too (I think). -- Len Ovens www.ovenwerks.net From joelz at pobox.com Sun Mar 2 06:51:15 2014 From: joelz at pobox.com (Joel Roth) Date: Sat, 1 Mar 2014 20:51:15 -1000 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: <20140302065115.GA9973@sprite> Aiyumi Moriya wrote: > Hi! Hi Aiyumi, > A visually impaired Linux user here. I recently came across this great > thread[1] and decided to write this message. My goal is a little > different from the original topic author (generating MIDI via code VS. > recording via a keyboard) so I decided to start a new discussion. > Here's my situation (warning: very long post!): > I'm trying to create some kind of audio setup for quite some time > (since around 2010)... but: > * I never got JACK to really work. JACK is pretty useful. You may be able to configure /etc/asoundrc or $HOME/.asoundrc so that ALSA I/O is routed to JACK[1]. > While my dream DAW doesn't come true, does anyone know a command line > way to record MIDI while simultaneously playing an audio file and > keeping both in sync (even if it involves JACK)? Nama[2] is a command-line application that uses Ecasound for recording and editing audio. Although it is currently oriented toward audio, Nama can send commands to a midish process, and has been used with midish and a2jmidi[3] for combined audio/MIDI recording under JACK. A simple hack for starting audio and MIDI in sync was to put midish and ecasound commands in the same line: nama> midish-command r; start # For midish, "r" means "record" I expect we'll eventually add in a MidiTrack class that would handle MIDI recording and playback. For now, you would have to issue all the midish commands yourself. Joel > Well, that's it. Sorry for the overly long post. Thanks for your time! 1. http://jackaudio.org/routing_alsa 2. https://freeshell.de/~bolangi/cgi1/nama.cgi/00home.html http://github.com/bolangi/nama 3. http://home.gna.org/a2jmidid/README The need for this appears to be partially if not fully obsoleted by the latest jack1: http://comments.gmane.org/gmane.comp.audio.jackit/27914 -- Joel Roth From ralf.mardorf at rocketmail.com Sun Mar 2 08:48:44 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 02 Mar 2014 09:48:44 +0100 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: <5312C3CD.2090506@hawaii.rr.com> References: <5312C3CD.2090506@hawaii.rr.com> Message-ID: <1393750124.596.186.camel@archlinux> A question to the OP, sorry, your mail was to long to read right now, perhaps I'll read it later. Do you prefer speech synth or braille? On Sat, 2014-03-01 at 19:38 -1000, david wrote: > Yet we have a member of the list who is blind and works with audio ... IIRC we have more than one member. Julien unfortunately for us, fortunately for him, is on an adventure. IIRC there are some other members visually impaired. Apart from Julien I only remember one other member too, but not his name. On musician forums there are likely several blind users, because even Linux GUIs oven provide text based configs, blind user are able to use. IIRC using Linux audio is more comfortable when the blind users prefer braille over speech synth. A blind formal member of this community: http://juliencoder.de/ I had some phone calls with him, he's very smart and has got knowledge about the things he's talking about. I never read the interview with Julien, perhaps there is some useful information too. http://www.zthmusic.com/julien-claassen/ Regards, Ralf From nettings at stackingdwarves.net Sun Mar 2 08:56:36 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 02 Mar 2014 09:56:36 +0100 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: <53129ADC.20806@hawaii.rr.com> Message-ID: <5312F244.5020709@stackingdwarves.net> On 03/02/2014 04:38 AM, Harry van Haaren wrote: > On Sun, Mar 2, 2014 at 2:43 AM, david > wrote: > > On 03/01/2014 08:51 AM, Harry van Haaren wrote: > >> I generally export to 32bit float .flac... so no dithering (or burning > >> to CD's :) > > Hmm, I thought FLAC only did 24-bit??? > I think the FLAC spec says it will handle anything from 4-32 bit-depth: > https://xiph.org/flac/faq.html#general__samples > That said, Audacity only has FLAC export options of 16 & 24 bit depths. > Ardour supports 8, 16 and 24. Still no 32 bit float support (at > application level). 24bit integer is equivalent to 32bit float in terms of resolution, and pretty much identical as long as the float samples are clamped between -1.0f and 1.0f. > I should correct my previous statement though: I *thought* I exported > 32bit float: but it turns out they're 24bits (from Ardour3), dithering > set to None. And cropping the resulting output in Audacity and exporting > was to 16-bit PCM, so I was actually doing this all wrong (no dithering, > 32 -> 24 -> 16). > > A better workflow would be to: > A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity 16bit > in, crop, 16bit out > B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, > crop, export 16bit (with dither). > > The important part being to not dither twice, since then you'll be > adding noise to the signal twice! actually, you *must* dither at every truncation step - if you don't, you will lose information _and_ introduce signal-dependent requantisation noise, which can never be removed again. so the best approach is to only reduce the wordlength once, at the very end of the chain, before going to CD. some mastering people (bob katz among them) even go as far as demanding dither at every level control in the chain. however, if you are exporting from JACK's native 32bit float to 24bit int _and_ you make sure that there is no sample larger than full-scale (floats are funny :), then there is no actual loss of information, and no dithering is required. all you do is map the 23 mantissa bits and the sign bit of the float to the 24 bits of the integer. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From self at thorstenwilms.com Sun Mar 2 09:54:01 2014 From: self at thorstenwilms.com (Thorsten Wilms) Date: Sun, 02 Mar 2014 10:54:01 +0100 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: <53129ADC.20806@hawaii.rr.com> Message-ID: <5312FFB9.9070805@thorstenwilms.com> On 03/02/2014 04:38 AM, Harry van Haaren wrote: > > A better workflow would be to: > A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity 16bit > in, crop, 16bit out > B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, > crop, export 16bit (with dither). An even better workflow would use the "Trim silence at start" and "Trim silence at end" options of format profiles, reachable via the Export dialog ;) The whole thing is designed for not having to edit the files at all, after export. -- Thorsten Wilms thorwil's design for free software: http://thorwil.wordpress.com/ From lists at parisson.com Sun Mar 2 12:29:11 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Sun, 02 Mar 2014 13:29:11 +0100 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <5307D147.6030509@parisson.com> References: <20140129024408.GA3961@ordinator> <52EA15A5.8070107@autostatic.com> <20140130141433.GB4676@ordinator> <5307D147.6030509@parisson.com> Message-ID: <53132417.2040305@parisson.com> Hi Lewis, Anything new with your tests? Did you try the kernel parameters below? G On 21/02/2014 23:20, Guillaume Pellerin wrote: > Hi Lewis, > > I know I'm late again, sorry > >> No luck so far, and the mystery continues. Are there specific kernel >> configuration options which you would advise disabling? Here is a >> snippet from my current /proc/config.gz which shows some settings >> related to USB: >> >> # -------------------------------------------- >> # >> # Miscellaneous USB options >> # >> CONFIG_USB_DEFAULT_PERSIST=y >> CONFIG_USB_DYNAMIC_MINORS=y > > yes, try: > CONFIG_USB_DYNAMIC_MINORS is not set > CONFIG_USB_SUSPEND is not set > > Guillaume > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > From murks at tuxfamily.org Sun Mar 2 12:34:11 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sun, 2 Mar 2014 13:34:11 +0100 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: References: <53129ADC.20806@hawaii.rr.com> Message-ID: <20140302133411.6ef47f54@eeyore.mozart.uni-klu.ac.at> On Sun, 2 Mar 2014 03:38:07 +0000 Harry van Haaren wrote: > On Sun, Mar 2, 2014 at 2:43 AM, david wrote: > > On 03/01/2014 08:51 AM, Harry van Haaren wrote: > >> I generally export to 32bit float .flac... so no dithering (or > >> burning to CD's :) > > Hmm, I thought FLAC only did 24-bit??? > I think the FLAC spec says it will handle anything from 4-32 > bit-depth: https://xiph.org/flac/faq.html#general__samples > That said, Audacity only has FLAC export options of 16 & 24 bit > depths. Ardour supports 8, 16 and 24. Still no 32 bit float support > (at application level). > > I should correct my previous statement though: I *thought* I exported > 32bit float: but it turns out they're 24bits (from Ardour3), > dithering set to None. And cropping the resulting output in Audacity > and exporting was to 16-bit PCM, so I was actually doing this all > wrong (no dithering, 32 -> 24 -> 16). > > A better workflow would be to: > A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity > 16bit in, crop, 16bit out > B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, > crop, export 16bit (with dither). > > The important part being to not dither twice, since then you'll be > adding noise to the signal twice! > > I'll be using option A above from now on I think, since it involves > less bit-depth changes. > Living and learning :) -Harry One of FLACs competitors does 32bit float since many years: wavpack. I'm not quite sure why it is much less popular than FLAC. Regards, Philipp -- JID: murks at jit.si From grekimj at acousticrefuge.com Sun Mar 2 13:11:58 2014 From: grekimj at acousticrefuge.com (Grekim Jennings) Date: Sun, 02 Mar 2014 08:11:58 -0500 Subject: [LAU] Dithering...should we dither about it? Message-ID: <53132E1E.70306@acousticrefuge.com> On 03/02/2014 04:38 AM, Harry van Haaren wrote: >/ On Sun, Mar 2, 2014 at 2:43 AM, david />/ >> wrote: />/ > On 03/01/2014 08:51 AM, Harry van Haaren wrote: />/ >> I generally export to 32bit float .flac... so no dithering (or burning />/ >> to CD's :) />/ > Hmm, I thought FLAC only did 24-bit??? />/ I think the FLAC spec says it will handle anything from 4-32 bit-depth: />/ https://xiph.org/flac/faq.html#general__samples />/ That said, Audacity only has FLAC export options of 16 & 24 bit depths. />/ Ardour supports 8, 16 and 24. Still no 32 bit float support (at />/ application level). / 24bit integer is equivalent to 32bit float in terms of resolution, and pretty much identical as long as the float samples are clamped between -1.0f and 1.0f. >/ I should correct my previous statement though: I *thought* I exported />/ 32bit float: but it turns out they're 24bits (from Ardour3), dithering />/ set to None. And cropping the resulting output in Audacity and exporting />/ was to 16-bit PCM, so I was actually doing this all wrong (no dithering, />/ 32 -> 24 -> 16). />/ />/ A better workflow would be to: />/ A) Ardour export 32 bit float -> 16bit (with dither) -> Audacity 16bit />/ in, crop, 16bit out />/ B) Ardour export 32 bit -> 24 bit (no dither) -> Audacity 24bit in, />/ crop, export 16bit (with dither). />/ />/ The important part being to not dither twice, since then you'll be />/ adding noise to the signal twice! / actually, you *must* dither at every truncation step - if you don't, you will lose information _and_ introduce signal-dependent requantisation noise, which can never be removed again. so the best approach is to only reduce the wordlength once, at the very end of the chain, before going to CD. some mastering people (bob katz among them) even go as far as demanding dither at every level control in the chain. however, if you are exporting from JACK's native 32bit float to 24bit int _and_ you make sure that there is no sample larger than full-scale (floats are funny :), then there is no actual loss of information, and no dithering is required. all you do is map the 23 mantissa bits and the sign bit of the float to the 24 bits of the integer. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net I just wanted add a couple thoughts. First, the level of the dither signal that you add depends on the target bit depth. So the dither for 24 bits is generally 8 bits (48 dB) softer than a dither for 16 bits. Therefore, there is is nothing wrong with dithering one or more times to 24 bits. Keep in mind that quantization distortion or dither noise will both be some 24 dB's below the threshold of hearing in a 24 bit file. I wouldn't be particularly worried about either, but dither is much better to have. I think you may run into issues if you use noise-shaped dither repeatedly. Or, using noise shaped dither followed by equalization would not be good in the sense that you could make the dither more audible. There also may be issues with dithering floating point material since the actual level of the musical signal is a bit of a moving target. So, I would really only make sure about dithering one time as the absolute last step in mastering to 16 bits. And if you have any thoughts about futher processing of a 16 bit track, do not used noise-shaped dither. Grekim www.acousticrefuge.com/mixer4.htm -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Sun Mar 2 13:29:40 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 2 Mar 2014 08:29:40 -0500 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: <5312C3CD.2090506@hawaii.rr.com> References: <5312C3CD.2090506@hawaii.rr.com> Message-ID: On Sun, Mar 2, 2014 at 12:38 AM, david wrote: > Yet we have a member of the list who is blind and works with audio ... > > I wasn't saying that it was impossible. I was saying that a huge part of how GUI-driven programs work is based on a feature that is very very very hard to replicate in non-GUI-driven programs. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sun Mar 2 13:59:26 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 2 Mar 2014 13:59:26 +0000 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: <5312F244.5020709@stackingdwarves.net> References: <53129ADC.20806@hawaii.rr.com> <5312F244.5020709@stackingdwarves.net> Message-ID: <20140302135926.GA8469@linuxaudio.org> On Sun, Mar 02, 2014 at 09:56:36AM +0100, J?rn Nettingsmeier wrote: > some mastering people (bob katz among them) even go as far as > demanding dither at every level control in the chain. In theory you'd need to dither every time a signal is rounded to some representable value. That would include intermediate results in filters etc. In practice, as long as you stay in 24 bits or float, most signals have enough noise on them (from analog recording) or are complex enough to be considered self-dithered (no correlation between the rounding error and the signal). I did some test to see the result of dithering years ago. They are still on my website but there's no link to them. Try Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From aiyumi.br at gmail.com Sun Mar 2 15:47:50 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Sun, 2 Mar 2014 12:47:50 -0300 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: 2014-03-02 5:48 GMT-03:00, Ralf Mardorf : > A question to the OP, sorry, your mail was to long to read right now, > perhaps I'll read it later. > > Do you prefer speech synth or braille? > I can only use speech synth via software. I don't have a Braille display nor a hardware voice synth. 2014-03-02 2:36 GMT-03:00, Len Ovens : > As you require the audio reading tools, you need to do one of two > things... use two audio devices one for the screen reader and one for > audio i/o and jack. Or use pulse and jackd(bus) bridged together. Lots of > people don't like pulse for political/personal/performance reasons. I read complaints of it affecting screen reader performance too, among other problems (in fact, most things I found about it were complaints), so I prefer not to use it. > Sorry, there is a third option. You can create an alsa device that is > really just connected to a jack port. This uses less cpu than pulse. You > have to have all this set up before the screen reader starts. Meaning that I won't have any audio feedback until I get that setup right. Maybe the easiest option is really to use a second device... > Not so sure about slackware's stock kernels, but the patch is not really > needed so long as the preempt switch is on. In ubuntu it is the > -lowlatency kernel and uses all the same video drivers as stock. Slackware > may have something similar, but even if not, rolling your own kernel on > the same source tree with no patch but the option changed should work with > the stock video too. Here, it's kernel 3.10.17. The threadirqs and 1000Hz options are on, but the preempt option (out of all things) is not :(. > Aiyumi Moriya wrote: >> * Each note shown as only one event, by matching the note on with it's >> respective note off. This way we don't need to struggle scrolling >> through the events trying to match the ons and offs. > > All the gui based midi editors do this, so it is not an unreasonable > request. Notes have a start time and length rather than a start time and > an end time. > >> * Possibility of inserting/editing/deleting all types of events, >> including text events, lyrics and SysEx messages. > > Just work to add it. But remember most authors do this stuff as a hobby > too. Of course. If I had the knowledge/skills, I would do it. 2014-03-02 3:51 GMT-03:00, Joel Roth : > Aiyumi Moriya wrote: >> While my dream DAW doesn't come true, does anyone know a command line >> way to record MIDI while simultaneously playing an audio file and >> keeping both in sync (even if it involves JACK)? > > Nama[2] is a command-line application that uses Ecasound for > recording and editing audio. Although it is currently > oriented toward audio, Nama can send commands to a midish > process, and has been used with midish and a2jmidi[3] for > combined audio/MIDI recording under JACK. > > A simple hack for starting audio and MIDI in sync was to put > midish and ecasound commands in the same line: > > nama> midish-command r; start # For midish, "r" means "record" > > I expect we'll eventually add in a MidiTrack class that > would handle MIDI recording and playback. For now, you > would have to issue all the midish commands yourself. Oh, that's great to know. I probably wouldn't have guessed. Everyone, thank you all for the tips :). -- ____________________ Blog: http://aiyumi.warpstar.net/ From ralf.mardorf at rocketmail.com Sun Mar 2 16:27:47 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 02 Mar 2014 17:27:47 +0100 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: <1393777667.668.36.camel@archlinux> On Sun, 2014-03-02 at 12:47 -0300, Aiyumi Moriya wrote: > Maybe the easiest option is really to use a second device... I suspect it's the only serious option ;), a blind user hopefully will chime in and correct us/me. For Orca or whatever you are using, you don't need a good audio device. Stay away from pulseaudio :D (just my opinion ;). From xiphmont at gmail.com Sun Mar 2 17:09:45 2014 From: xiphmont at gmail.com (Monty Montgomery) Date: Sun, 2 Mar 2014 12:09:45 -0500 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: <53132E1E.70306@acousticrefuge.com> References: <53132E1E.70306@acousticrefuge.com> Message-ID: > actually, you *must* dither at every truncation step - if you don't, you > will lose information _and_ introduce signal-dependent requantisation > noise, which can never be removed again. > so the best approach is to only reduce the wordlength once, at the very > end of the chain, before going to CD. This is a correct recommendation. > some mastering people (bob katz among them) even go as far as demanding > dither at every level control in the chain. Strictly speaking, this does not save you (and by you, I mean Bob Katz). Further operations after dithering can 'break' the dither and reintroduce distortion. Then you have the worst of both worlds; the distortion _and_ the added noise. Working at 24 bit / 32 float, this is academic and won't hurt you in any audible way. It's too far below the audible floor to care about. Monty From willgodfrey at musically.me.uk Sun Mar 2 23:18:30 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sun, 2 Mar 2014 23:18:30 +0000 Subject: [LAU] Need some programming help. In-Reply-To: <20131122003308.GA1840@rhk.homenet.telecomitalia.it> References: <20131118211444.7c996b70@debian> <20131119115829.GA1772@rhk.homenet.telecomitalia.it> <20131121192231.3e2b5632@debian> <20131122003308.GA1840@rhk.homenet.telecomitalia.it> Message-ID: <20140302231830.3e5fb9c9@debian> On Fri, 22 Nov 2013 01:33:08 +0100 Tito Latini wrote: > On Thu, Nov 21, 2013 at 07:22:31PM +0000, Will Godfrey wrote: > > Thank you both for your suggestions. I tried that bit of code, but as I rather > > expected it behaved perfectly. > > > > The problem seems to be that, for some reason FLTKs 'step' command is being > > ignored. Not only is it failing to round the values but it is also failing to > > truncate them. I even tried replacing the calculated value with a simple > > integer - loads of trailing zeros after the decimal point. > > > > Yoshi is derived from Zyn, and the section of code that sends values to FLTK > > for display is identical, yet Zyn doesn't show this problem. I can only guess > > there is some difference in the way yoshi links to FLTK but can't imagine what > > it is! > > I'm noticing a possible bug generator in a diff between > `ZynAddSubFX-2.4.3' and `yoshimi-1.1.0': > > diff -u ZynAddSubFX-2.4.3/src/UI/ADnoteUI.fl \ > yoshimi-1.1.0/src/UI/ADnoteUI.fl | grep lrintf > + callback {pars->VoicePar[nvoice].PVolume = lrintf(o->value());} > + callback {pars->VoicePar[nvoice].Presonance = lrintf(o->value());} > [...] > > !! | wc -l > 72 > > The method `value' of Fl_Valuator returns a "double" and `lrintf' > requires "float". > > The implementation of `lrintf' depends on the machine (for example, in > glibc-2.15, it uses the instruction CVTSD2SI for x86_64). > > You could try `lrint' or the follow trick: Sorry to be so incredibly slow responding... I'm now in the fortunate? position of being able to devote a lot more time to bug hunting. I tried your suggestion but unfortunately it made no difference. What is even stranger is that if I compile and run the synth on an old version of debian there is no problem. However *exactly* the same image run on the same machine but with debian upgraded shows the fault. A possibly significant difference is the change from FLTK V1.1 to V1.3 but looking on their website I can find nothing that would give a clue. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Sun Mar 2 23:21:30 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sun, 2 Mar 2014 23:21:30 +0000 Subject: [LAU] Little mellow Yoshimi DX improvisation In-Reply-To: <52698B9E.6070702@gmail.com> References: <52698B9E.6070702@gmail.com> Message-ID: <20140302232130.1e472131@debian> On Thu, 24 Oct 2013 23:05:34 +0200 Lorenzo Sutton wrote: > Just a little, mellow improvisation with yoshimi DX rhodes while testing > my E-MU midi-usb dongle with my ancient yamaha keyboard. No editing, > just recorded directly through jack... > > Dedicated with admiration and appreciation to all you sleepless open > source developers out there :-) > > http://lorenzosu.net/temp/yoshimi_dx_impro.ogg > > Good night, > Lorenzo. Simple idea, but very pleasant and relaxing. Sometimes less is more. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From harryhaaren at gmail.com Mon Mar 3 10:04:46 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Mon, 3 Mar 2014 10:04:46 +0000 Subject: [LAU] Dithering...should we dither about it? In-Reply-To: <20140302135926.GA8469@linuxaudio.org> References: <53129ADC.20806@hawaii.rr.com> <5312F244.5020709@stackingdwarves.net> <20140302135926.GA8469@linuxaudio.org> Message-ID: On Sun, Mar 2, 2014 at 1:59 PM, Fons Adriaensen wrote: > Interesting, thanks for sharing. @Paul, apologies, I made an incorrect assumption there. @Thorsten, indeed the trim functions (and add functions) could be used... I'm a bit visual in my "silence adding" workflow.. and generally add some extra compression / mastering in Audacity after export from Ardour: hence the extra import-export. Interesting discussion! Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From silvain at freeshell.de Mon Mar 3 10:36:47 2014 From: silvain at freeshell.de (F. Silvain) Date: Mon, 3 Mar 2014 11:36:47 +0100 (CET) Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: <5312C3CD.2090506@hawaii.rr.com> Message-ID: <1403031135300.25911@%3.3s> Hey hey, (sorry for replying to the thread in general, my mail archive is broken.) I'm not blind yet but on the way. I inherited some software tips and tricks from a friend. Nama is a DAW with good audio support and limited MIDI support. It's based on Ecasound and Midish. It can be remote controlled with a USB footswitch, a piece of gaming equipment. One blind guy on the Nama mailinglist uses it to great effect. Nama can be synchronised to MIDI with a little script I inherited as well. Midish connects easily to ALSA sequencer ports. If you're still interested, I can pass a file of convenience commands that I've been working on. It's not finished yet. There are also commands for connecting Midish to hardware and software synthesizers via ALSA sequencer. JACK can run on any reasonably modern system without a realtime kernel. Use it on your second soundcard - the Fast Track Pro. There are commandline patchbays as well like esjit or jackctl.py and others. Despite their names, most can also handle ALSA sequencer MIDI ports too. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From mista.tapas at gmx.net Mon Mar 3 11:15:28 2014 From: mista.tapas at gmx.net (Florian Paul Schmidt) Date: Mon, 03 Mar 2014 12:15:28 +0100 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: References: Message-ID: <53146450.7090709@gmx.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 02.03.2014 02:16, Aiyumi Moriya wrote: > Hi! > > And last but not least, on the thread in [1], I found Teqqer[18]. > I couldn't test it yet because of the JACK problem (I seriously > need to sort that out), but I took a look at the default config > variables in the Python source code, and it looks like scripting > support and customizable shortcuts are mostly taken care of. From > what I understand from the shortcuts, the interface for > inserting/editing MIDI events is a bit similar to Midiedit (Up/Down > to cycle through events, although note ons and offs are separate) > and it seems to have one view for each track (Left/Right to cycle > through tracks), and it even supports patterns! :D So, minus the > recording part (from what I understood, it wasn't made for that), > the lack of ALSA (without JACK) support, I think it has the > potential to become something like what I'm looking for (plus, I > always wanted to try a tracker, but never found a "screen reader > friendly" one). I'll definitely be watching Teqqer to see where it > goes! Hi, as the author of the (unfinished) teqqer I have to add that recording MIDI is definitely on the agenda for the future. I have to find some time to work on it again and fix the long issues list before I can make a proper release. And I also still ponder how to integrate a modular synth like Carla into it so it becomes an integrated music making package. I do not plan to support recording of audio though, as I don't really know how to represent that in a tracker format. Sure one could use a "generic" audio event type and associated track type. But I'm not sure on how well that would work.. Input appreciated.. Have fun, Flo -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.14 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iQEcBAEBAgAGBQJTFGROAAoJEA5f4Coltk8ZdtgH/0xerQPjxAm+rQD1wQ+FYAOS FoP5ViFeiVDqmEZq+wWn02Q8pDV8/iZLxNuImvtz9YuCSWKaSthkig+pfDgQ5G/0 L/tNQ7ZocS0skVURRwWBuBoj/9mHvHcdWcBMK0XX1aMYmOO9R+JlNu8Sd/Q+8toD MaCS3u4lk5L968n7LVPI0GeN7l+/OlF0JatpHzF7VuqEtUysxeKNeD0MtBfgcOWA /FinZkSaje43yG1tlm8ZLTAxgRbj5VzlJtLal/cfl2dDHByjYq1napEmx74qVeOB oSoP6ShWB7iKOxCxoCLHNwvvP69LKAVzqHUfkLm5z7JkshQcYtj2VrHxSRU8lwc= =MM0L -----END PGP SIGNATURE----- From silvain at freeshell.de Mon Mar 3 11:42:41 2014 From: silvain at freeshell.de (F. Silvain) Date: Mon, 3 Mar 2014 12:42:41 +0100 (CET) Subject: [LAU] Music made with Linux - a couple of tracks Message-ID: <1403031239020.27779@freeshell.de> Hey hey, I took the plunge. Here are my first three tracks, recorded on Linux. http://freeshell.de/~silvain/audio/beatus.ogg http://freeshell.de/~silvain/audio/dragonride.ogg http://freeshell.de/~silvain/audio/letter_to_the_minnesaenger.ogg FWIW I also added a small poem. Maybe I'll set some music to it in future: http://freeshell.de/~silvain/vogonic.html Please give me some feedback, because I want to learn for the future. :) Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From tito.01beta at gmail.com Mon Mar 3 17:54:26 2014 From: tito.01beta at gmail.com (Tito Latini) Date: Mon, 3 Mar 2014 18:54:26 +0100 Subject: [LAU] Need some programming help. In-Reply-To: <20140302231830.3e5fb9c9@debian> References: <20131118211444.7c996b70@debian> <20131119115829.GA1772@rhk.homenet.telecomitalia.it> <20131121192231.3e2b5632@debian> <20131122003308.GA1840@rhk.homenet.telecomitalia.it> <20140302231830.3e5fb9c9@debian> Message-ID: <20140303175426.GA1781@rhk.homenet.telecomitalia.it> On Sun, Mar 02, 2014 at 11:18:30PM +0000, Will Godfrey wrote: > What is even stranger is that if I compile and run the synth on an old > version of debian there is no problem. However *exactly* the same image run on > the same machine but with debian upgraded shows the fault. > > A possibly significant difference is the change from FLTK V1.1 to V1.3 but > looking on their website I can find nothing that would give a clue. Now I'm using gcc-4.8.2 and I can reproduce the problem. Only a rapid glance: with /* From Fl_Valuator.cxx */ void Fl_Valuator::step(double s) { if (s < 0) s = -s; A = rint(s); B = 1; while (fabs(s-A/B) > epsilon && B<=(0x7fffffff/10)) {B *= 10; A = rint(s*B);} } there is a roundoff error but if we use the simplest /* From Fl_Valuator.H */ void step(double a, int b) {A = a; B = b;} the problem disappears --- yoshimi-1.1.0~/src/UI/ADnoteUI.fl 2013-05-08 09:47:45.000000000 +0200 +++ yoshimi-1.1.0/src/UI/ADnoteUI.fl 2014-03-03 14:41:58.054425629 +0100 @@ -89,8 +89,9 @@ } {} Fl_Value_Output detunevalueoutput { callback {o->value(getDetune((pars->VoicePar[nvoice].PDetuneType == 0) ? (pars->GlobalPar.PDetuneType) : (pars->VoicePar[nvoice].PDetuneType), 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} - xywh {265 5 45 20} labelsize 10 align 5 minimum -5000 maximum 5000 step 0.01 textfont 1 textsize 10 - code0 {o->value(getDetune(pars->VoicePar[nvoice].PDetuneType, 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} + xywh {265 5 45 20} labelsize 10 align 5 minimum -5000 maximum 5000 textfont 1 textsize 10 + code0 {o->step(0.01, 1);} + code1 {o->value(getDetune(pars->VoicePar[nvoice].PDetuneType, 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} } [ ... censored for avoiding noise in LAU; the complete patch is in a private msg ] Tito Latini From WillGodfrey at musically.me.uk Mon Mar 3 18:49:50 2014 From: WillGodfrey at musically.me.uk (Will J Godfrey) Date: Mon, 3 Mar 2014 18:49:50 +0000 Subject: [LAU] Need some programming help. In-Reply-To: <20140303175426.GA1781@rhk.homenet.telecomitalia.it> References: <20131118211444.7c996b70@debian> <20131119115829.GA1772@rhk.homenet.telecomitalia.it> <20131121192231.3e2b5632@debian> <20131122003308.GA1840@rhk.homenet.telecomitalia.it> <20140302231830.3e5fb9c9@debian> <20140303175426.GA1781@rhk.homenet.telecomitalia.it> Message-ID: <20140303184950.35748ea0@debian> On Mon, 3 Mar 2014 18:54:26 +0100 Tito Latini wrote: > On Sun, Mar 02, 2014 at 11:18:30PM +0000, Will Godfrey wrote: > > What is even stranger is that if I compile and run the synth on an old > > version of debian there is no problem. However *exactly* the same image run on > > the same machine but with debian upgraded shows the fault. > > > > A possibly significant difference is the change from FLTK V1.1 to V1.3 but > > looking on their website I can find nothing that would give a clue. > > Now I'm using gcc-4.8.2 and I can reproduce the problem. > > Only a rapid glance: with > > /* From Fl_Valuator.cxx */ > void Fl_Valuator::step(double s) { > if (s < 0) s = -s; > A = rint(s); > B = 1; > while (fabs(s-A/B) > epsilon && B<=(0x7fffffff/10)) {B *= 10; A = rint(s*B);} > } > > there is a roundoff error but if we use the simplest > > /* From Fl_Valuator.H */ > void step(double a, int b) {A = a; B = b;} > > the problem disappears > > --- yoshimi-1.1.0~/src/UI/ADnoteUI.fl 2013-05-08 09:47:45.000000000 +0200 > +++ yoshimi-1.1.0/src/UI/ADnoteUI.fl 2014-03-03 14:41:58.054425629 +0100 > @@ -89,8 +89,9 @@ > } {} > Fl_Value_Output detunevalueoutput { > callback {o->value(getDetune((pars->VoicePar[nvoice].PDetuneType == 0) ? (pars->GlobalPar.PDetuneType) : (pars->VoicePar[nvoice].PDetuneType), 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} > - xywh {265 5 45 20} labelsize 10 align 5 minimum -5000 maximum 5000 step 0.01 textfont 1 textsize 10 > - code0 {o->value(getDetune(pars->VoicePar[nvoice].PDetuneType, 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} > + xywh {265 5 45 20} labelsize 10 align 5 minimum -5000 maximum 5000 textfont 1 textsize 10 > + code0 {o->step(0.01, 1);} > + code1 {o->value(getDetune(pars->VoicePar[nvoice].PDetuneType, 0, pars->VoicePar[nvoice].PDetune) * pars->getBandwidthDetuneMultiplier());} > } > > [ ... censored for avoiding noise in LAU; the complete patch is in a private msg ] > > Tito Latini Thanks a lot for this. It has been a nagging source of irritation for years! -- It wasn't me! (Well actually, it probably was) ... the hard part is not dodging what life throws at you, but trying to catch the good bits. From aiyumi.br at gmail.com Mon Mar 3 23:48:54 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Mon, 3 Mar 2014 20:48:54 -0300 Subject: [LAU] Music Editors for Blind Users? In-Reply-To: <1403031135300.25911@%3.3s> References: <5312C3CD.2090506@hawaii.rr.com> <1403031135300.25911@%3.3s> Message-ID: 2014-03-03 7:36 GMT-03:00, F. Silvain : > Nama is a DAW with good audio support and limited MIDI support. It's based > on Ecasound and Midish. It can be remote controlled with a USB footswitch, a > piece of gaming equipment. One blind guy on the Nama mailinglist uses it to > great effect. Nama can be synchronised to MIDI with a little script I > inherited as well. I hadn't tried Nama because Ecasound was already meeting my non-MIDI needs. Before this discussion, I didn't know that Nama was integrated with Midish. Now that I know, I'll give it a try. > Midish connects easily to ALSA sequencer ports. If you're still interested, > I can pass a file of convenience commands that I've been working on. Oh, please :D. > [...] There are commandline > patchbays as well like esjit or jackctl.py and others. Despite their names, > most can also handle ALSA sequencer MIDI ports too. I didn't know of these. I'll check them out. 2014-03-03 8:15 GMT-03:00, Florian Paul Schmidt : > Hi, > > as the author of the (unfinished) teqqer I have to add that recording > MIDI is definitely on the agenda for the future. Really? That's great! > [...] I do not plan to support recording of audio though, as > I don't really know how to represent that in a tracker format. Sure > one could use a "generic" audio event type and associated track type. > But I'm not sure on how well that would work.. Input appreciated.. Actually, what I really wanted was MIDI recording (what I thought of audio was just for playback along with the MIDI tracks), so that'd be good enough for me. That makes Teqqer a strong candidate for becoming my MIDI editor of choice :D. Until then, I'll see what I can do with Nama and Midish. Once again, thank you everyone! -- ____________________ Blog: http://aiyumi.warpstar.net/ From carlo.ratm at gmail.com Tue Mar 4 13:27:04 2014 From: carlo.ratm at gmail.com (Carlo Ascani) Date: Tue, 4 Mar 2014 14:27:04 +0100 Subject: [LAU] Collaborate on the same Ardour project Message-ID: Hi, say two guys would like to collaborate on the same Ardour project. Remotely. Not realtime. What would be the easiest way to share the project? I am thinking about taking the files under git (session's 'interchange' dir *only*). Is it completely insane? Any ideas ? Thank you -- Carlo Ascani | carlorat.me skype: carloratm From paul at linuxaudiosystems.com Tue Mar 4 13:39:25 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 4 Mar 2014 08:39:25 -0500 Subject: [LAU] Collaborate on the same Ardour project In-Reply-To: References: Message-ID: On Tue, Mar 4, 2014 at 8:27 AM, Carlo Ascani wrote: > Hi, > say two guys would like to collaborate on the same Ardour project. > Remotely. Not realtime. > > What would be the easiest way to share the project? > I am thinking about taking the files under git (session's > 'interchange' dir *only*). > > Is it completely insane? > you really want the whole session tree. using git has some downsides, but it isn't insane. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sakrecoer at gmail.com Tue Mar 4 13:41:16 2014 From: sakrecoer at gmail.com (Set Hallstrom) Date: Tue, 04 Mar 2014 14:41:16 +0100 Subject: [LAU] Blofeld Patch editor for linux Message-ID: <5315D7FC.6070805@gmail.com> Hi guys! The amzing mr Ricard has created a sweet patch editor for Waldorf Blofeld. If you also happen to love that box but not so much its interface, this is special for you! https://github.com/polluxsynth/midiedit Have a delicious awoken time, -- Set Hallstrom AKA Sakrecoer http://sakrecoer.com From carlo.ratm at gmail.com Tue Mar 4 14:03:28 2014 From: carlo.ratm at gmail.com (Carlo Ascani) Date: Tue, 4 Mar 2014 15:03:28 +0100 Subject: [LAU] Collaborate on the same Ardour project In-Reply-To: References: Message-ID: 2014-03-04 14:39 GMT+01:00 Paul Davis : > > > you really want the whole session tree. using git has some downsides, but it > isn't insane. > Thank you. Do we need the same minor version to make sure things work? -- Carlo Ascani | carlorat.me skype: carloratm From idragosani at gmail.com Tue Mar 4 14:17:40 2014 From: idragosani at gmail.com (Brett McCoy) Date: Tue, 4 Mar 2014 09:17:40 -0500 Subject: [LAU] Collaborate on the same Ardour project In-Reply-To: References: Message-ID: On Tue, Mar 4, 2014 at 8:27 AM, Carlo Ascani wrote: > say two guys would like to collaborate on the same Ardour project. > Remotely. Not realtime. > > What would be the easiest way to share the project? > I am thinking about taking the files under git (session's > 'interchange' dir *only*). > > Is it completely insane? > > Any ideas ? > > > Thank you git would be a good idea if you don't want to stomp on each other's work. You could also use a shared directory on Dropbox or similar kind of service, but you'd have to coordinate so you aren't overwriting someone else's files with your stuff. -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi From zotz at 100jamz.com Tue Mar 4 15:43:36 2014 From: zotz at 100jamz.com (drew Roberts) Date: Tue, 4 Mar 2014 10:43:36 -0500 Subject: [LAU] Collaborate on the same Ardour project In-Reply-To: References: Message-ID: <201403041043.36380.zotz@100jamz.com> On Tuesday 04 March 2014 08:27:04 Carlo Ascani wrote: > Hi, > say two guys would like to collaborate on the same Ardour project. > Remotely. Not realtime. > > What would be the easiest way to share the project? > I am thinking about taking the files under git (session's > 'interchange' dir *only*). > > Is it completely insane? > > Any ideas ? Here is what we did several years ago: http://packet-in.org/wiki/index.php?title=Collaboration_Process http://packet-in.org/wiki/index.php?title=Our_Tools http://packet-in.org/wiki/index.php?title=Our_Gear Our repository: http://packet-in.org/repo/ I need to expand on the Collaboration_Process page as I seem to recall us having developed scripts to wavpack files for xfer and such. Any other Packet-In folks have a batter memory for this aspect? > > > Thank you drew From egor.sanin at gmail.com Tue Mar 4 19:53:11 2014 From: egor.sanin at gmail.com (Egor Sanin) Date: Tue, 4 Mar 2014 14:53:11 -0500 Subject: [LAU] Processed poetry Message-ID: Hi List, I'd like to share a few pieces that I've completed recently. "Processed poetry" is a series of poetry readings, augmented by electronic processing. The first four pieces are all created with SuperCollider on Arch Linux. http://integer0.users.sourceforge.net/pages/processed-poetry.html If you prefer direct download, please use the following links: https://soundcloud.com/integer0/processed-poetry-young-man-v2/download https://soundcloud.com/integer0/processed-poetry-young-man-v1/download https://soundcloud.com/integer0/processed-poetry-ideas/download https://soundcloud.com/integer0/processed-poetry-verklarte/download Enjoy! From willgodfrey at musically.me.uk Tue Mar 4 22:39:57 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 4 Mar 2014 22:39:57 +0000 Subject: [LAU] Yoshimi Message-ID: <20140304223957.6a39850e@debian> In case anyone is interested. There are some bug fixes, and plans for more feature additions. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From gnome at hawaii.rr.com Wed Mar 5 07:41:33 2014 From: gnome at hawaii.rr.com (david) Date: Tue, 04 Mar 2014 21:41:33 -1000 Subject: [LAU] Yoshimi In-Reply-To: <20140304223957.6a39850e@debian> References: <20140304223957.6a39850e@debian> Message-ID: <5316D52D.3080400@hawaii.rr.com> On 03/04/2014 12:39 PM, Will Godfrey wrote: > In case anyone is interested. There are some bug fixes, and plans for more > feature additions. Hurrah! -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From byronkeys at jvlnet.com Wed Mar 5 18:25:58 2014 From: byronkeys at jvlnet.com (Brian Hagen) Date: Wed, 05 Mar 2014 12:25:58 -0600 Subject: [LAU] internal sound source for Audacity Message-ID: <53176C36.3070506@jvlnet.com> I have an unusually pleasant surprise here: a MIDI wind controller can drive a software synth without even having both of their apps running. I would like to be able to get the resulting audio-signal that is an output from this (normally sent to the PC's speakers) redirected to Audacity's input-selector so that I can record it. Any ideas? Brian From moshwe at gmail.com Wed Mar 5 18:45:50 2014 From: moshwe at gmail.com (Moshe Werner) Date: Wed, 5 Mar 2014 20:45:50 +0200 Subject: [LAU] internal sound source for Audacity In-Reply-To: <53176C36.3070506@jvlnet.com> References: <53176C36.3070506@jvlnet.com> Message-ID: On Wed, Mar 5, 2014 at 8:25 PM, Brian Hagen wrote: > I have an unusually pleasant surprise here: a MIDI wind controller > can drive a software synth without even having both of their apps > running. I would like to be able to get the resulting audio-signal > that is an output from this (normally sent to the PC's speakers) > redirected to Audacity's input-selector so that I can record it. > > Any ideas? > Jack? Or am I missing something? -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Wed Mar 5 21:45:06 2014 From: len at ovenwerks.net (Len Ovens) Date: Wed, 5 Mar 2014 13:45:06 -0800 (PST) Subject: [LAU] internal sound source for Audacity In-Reply-To: <53176C36.3070506@jvlnet.com> References: <53176C36.3070506@jvlnet.com> Message-ID: On Wed, 5 Mar 2014, Brian Hagen wrote: > I have an unusually pleasant surprise here: a MIDI wind controller > can drive a software synth without even having both of their apps > running. I would like to be able to get the resulting audio-signal > that is an output from this (normally sent to the PC's speakers) > redirected to Audacity's input-selector so that I can record it. Yes you can use jack for this. - Audacity auto connects and so you need to manually disconnect and reconnect. - Audacity does not connect until record is started. The best thing to do is hit pause, then record, then make connection changes in jack, then unpause to begin recording. - Audacity is not listed in jack under it's own name but the name of the library it uses to make jack ports. - The name of the port for audacity changes every time you hit record so this can not be automated... maybe with wild cards. For just recording, I would use mhwaveedit which works with jack "properly", that is it makes jack ports as soon as started and they remain as long as the program is running. Once the wav file is recorded there may be some editing tasks that Audacity does better (some of the built in effects). -- Len Ovens www.ovenwerks.net From len at ovenwerks.net Wed Mar 5 21:54:30 2014 From: len at ovenwerks.net (Len Ovens) Date: Wed, 5 Mar 2014 13:54:30 -0800 (PST) Subject: [LAU] internal sound source for Audacity In-Reply-To: References: <53176C36.3070506@jvlnet.com> Message-ID: On Wed, 5 Mar 2014, Len Ovens wrote: > - The name of the port for audacity changes every time you hit record so > this can not be automated... maybe with wild cards. This seems to have been fixed. It is just called "portaudio" now with no number. Len From gnome at hawaii.rr.com Thu Mar 6 07:20:02 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 05 Mar 2014 21:20:02 -1000 Subject: [LAU] internal sound source for Audacity In-Reply-To: References: <53176C36.3070506@jvlnet.com> Message-ID: <531821A2.9060609@hawaii.rr.com> On 03/05/2014 11:45 AM, Len Ovens wrote: > > > On Wed, 5 Mar 2014, Brian Hagen wrote: > >> I have an unusually pleasant surprise here: a MIDI wind controller >> can drive a software synth without even having both of their apps >> running. I would like to be able to get the resulting audio-signal >> that is an output from this (normally sent to the PC's speakers) >> redirected to Audacity's input-selector so that I can record it. > > Yes you can use jack for this. > > - Audacity auto connects and so you need to manually disconnect and > reconnect. > - Audacity does not connect until record is started. The best thing to > do is hit pause, then record, then make connection changes in jack, then > unpause to begin recording. > - Audacity is not listed in jack under it's own name but the name of > the library it uses to make jack ports. > - The name of the port for audacity changes every time you hit record > so this can not be automated... maybe with wild cards. > > For just recording, I would use mhwaveedit which works with jack > "properly", that is it makes jack ports as soon as started and they > remain as long as the program is running. Once the wav file is recorded > there may be some editing tasks that Audacity does better (some of the > built in effects). Or you can use jack_capture to record the WAV file, and process it afterwards in Audacity. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From moshwe at gmail.com Thu Mar 6 08:50:50 2014 From: moshwe at gmail.com (Moshe Werner) Date: Thu, 6 Mar 2014 10:50:50 +0200 Subject: [LAU] internal sound source for Audacity In-Reply-To: <531821A2.9060609@hawaii.rr.com> References: <53176C36.3070506@jvlnet.com> <531821A2.9060609@hawaii.rr.com> Message-ID: > > Um ... I'm not certain what you mean. Normally, Audacity > routes its output signals to the PC's speakers. What I want to do is > to route signals that are normally headed for those speakers > to go to the Audacity input ... I am not an expert in JACK; > I wish I was, but to me it is mainly a behind-the-scenes resource. > :-) If you've got some sort of Jack front end (like qjackctl) you could open the "Connections" window and route audio signals where you want them. Some info: qjackctl https://help.ubuntu.com/community/HowToQjackCtlConnections Hope this helps. Best Moshe -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelz at pobox.com Thu Mar 6 11:48:05 2014 From: joelz at pobox.com (Joel Roth) Date: Thu, 6 Mar 2014 01:48:05 -1000 Subject: [LAU] Music made with Linux - a couple of tracks In-Reply-To: <1403031239020.27779@freeshell.de> References: <1403031239020.27779@freeshell.de> Message-ID: <20140306114805.GA29848@sprite> On Mon, Mar 03, 2014 at 12:42:41PM +0100, F. Silvain wrote: > Hey hey, > I took the plunge. Here are my first three tracks, recorded on Linux. > http://freeshell.de/~silvain/audio/beatus.ogg > http://freeshell.de/~silvain/audio/dragonride.ogg > http://freeshell.de/~silvain/audio/letter_to_the_minnesaenger.ogg > > FWIW I also added a small poem. Maybe I'll set some music to it in future: > http://freeshell.de/~silvain/vogonic.html > > Please give me some feedback, because I want to learn for the future. :) Hey hey! The songs all seem musical, and are pleasing to listen to. Perhaps golden-eared others can give more specific feedback. As for the poetry, I think Vogonic is underutilized art form. > Ta-ta > ---- > Ffanci > * Internet: http://freeshell.de/~silvain > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Joel Roth From rncbc at rncbc.org Thu Mar 6 21:20:42 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Thu, 06 Mar 2014 21:20:42 +0000 Subject: [LAU] [ANN] Vee One Suite 0.4.0 - A proto-beta party! Message-ID: <5318E6AA.4020102@rncbc.org> Howdy, as to no surprise (rly?) the so called Vee One Suite of old-schoolyards gets bullied down to another bump in the head: now turning into a proto-beta party? righto! again as usual, all made available in dual form: - a pure stand-alone JACK client with JACK-session, NSM (Non Session management) and both JACK MIDI and ALSA MIDI input support; - a LV2 instrument plugin. all free and open-source Linux Audio software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. [1] synthv1 - an old-school polyphonic synthesizer synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic synthesizer with stereo fx. LV2 URI: http://synthv1.sourceforge.net/lv2 website: http://synthv1.sourceforge.net downloads: http://sourceforge.net/projects/synthv1/files - source tarball: http://download.sourceforge.net/synthv1/synthv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.i586.rpm http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.x86_84.rpm [2] samplv1 - an old-school polyphonic sampler samplv1 is an(other) old-school all-digital polyphonic sampler synthesizer with stereo fx. LV2 URI: http://samplv1.sourceforge.net/lv2 website: http://samplv1.sourceforge.net downloads: http://sourceforge.net/projects/samplv1/files - source tarball: http://download.sourceforge.net/samplv1/samplv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.i586.rpm http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.x86_84.rpm [3] drumkv1 - an old-school drum-kit sampler drumkv1 is (yet) an(other) old-school all-digital drum-kit sampler synthesizer with stereo fx. LV2 URI: http://drumkv1.sourceforge.net/lv2 website: http://drumkv1.sourceforge.net downloads: http://sourceforge.net/projects/drumkv1/files - source tarball: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.i586.rpm http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.x86_84.rpm see also: http://www.rncbc.org/drupal/node/770 enjoy && have fun! -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From mira.mikes at gmail.com Fri Mar 7 00:48:04 2014 From: mira.mikes at gmail.com (=?ISO-8859-2?Q?Jarom=EDr_Mike=B9?=) Date: Fri, 7 Mar 2014 01:48:04 +0100 Subject: [LAU] ebur128 batch processing Message-ID: Hi, I will got about 100 stereo wav files mixed hopefully in similar way (loudness). I need them process to meet ebur128 specification. True peaks -3dB RMS -23dB Any chance to do it as batch process? Result quality is critical as material will be broadcast on TV. best mira -------------- next part -------------- An HTML attachment was scrubbed... URL: From robin at gareus.org Fri Mar 7 10:09:17 2014 From: robin at gareus.org (Robin Gareus) Date: Fri, 07 Mar 2014 11:09:17 +0100 Subject: [LAU] [ANN] Vee One Suite 0.4.0 - A proto-beta party! In-Reply-To: <5318E6AA.4020102@rncbc.org> References: <5318E6AA.4020102@rncbc.org> Message-ID: <53199ACD.2080800@gareus.org> On 03/06/2014 10:20 PM, Rui Nuno Capela wrote: > Howdy, > > as to no surprise (rly?) the so called Vee One Suite of old-schoolyards > gets bullied down to another bump in the head: now turning into a > proto-beta party? Thanks for all your work and organizing a party on top of it all! What's new in 0.4.0? - neither the email nor your blog mentions that. Is there an easily accessible changelog? robin From rmouneyres at gmail.com Fri Mar 7 10:28:45 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Fri, 7 Mar 2014 11:28:45 +0100 Subject: [LAU] Jack vu-meter to OSC Message-ID: Hello, Does someone know about a jack client app which is able to publish vu-metering/amplitude of audio jack ports signals as an OSC server ? I mean an app like meterbridge but with OSC output instead of graphical output. thanks Rapha?l From robin at gareus.org Fri Mar 7 10:39:32 2014 From: robin at gareus.org (Robin Gareus) Date: Fri, 07 Mar 2014 11:39:32 +0100 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: References: Message-ID: <5319A1E4.1090206@gareus.org> On 03/07/2014 11:28 AM, Rapha?l Mouneyres wrote: > Hello, > > Does someone know about a jack client app which is able to publish > vu-metering/amplitude of audio jack ports signals as an OSC server ? > I mean an app like meterbridge but with OSC output instead of graphical output. Not exactly. It's JSON output (for use with a webserver) and digital-peak (not VU): http://gareus.org/gitweb/?p=jack-peak.git ciao, robin PS. I think Ardour3 can provide meter-info per route via OSC. From rmouneyres at gmail.com Fri Mar 7 10:54:17 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Fri, 7 Mar 2014 11:54:17 +0100 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: <5319A1E4.1090206@gareus.org> References: <5319A1E4.1090206@gareus.org> Message-ID: looks very promising for my use. digital-peak with [0,1] output is what i need to output on OSC. The code is short and clean enough for me to read, I should be able to add the OSC server output with a new -o option. I'll let you know if i'm going for it and achieve something working. thanks robin ! Rapha?l 2014-03-07 11:39 UTC+01:00, Robin Gareus : > On 03/07/2014 11:28 AM, Rapha?l Mouneyres wrote: >> Hello, >> >> Does someone know about a jack client app which is able to publish >> vu-metering/amplitude of audio jack ports signals as an OSC server ? >> I mean an app like meterbridge but with OSC output instead of graphical >> output. > > Not exactly. It's JSON output (for use with a webserver) and > digital-peak (not VU): http://gareus.org/gitweb/?p=jack-peak.git > > ciao, > robin > > PS. I think Ardour3 can provide meter-info per route via OSC. > From hamish.low.net at gmx.com Fri Mar 7 13:12:57 2014 From: hamish.low.net at gmx.com (sub_acoustic) Date: Fri, 7 Mar 2014 05:12:57 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <1393538864605-89649.post@n7.nabble.com> References: <1386015225436-88069.post@n7.nabble.com> <529CF0B8.6050304@youmail.dk> <1386262286208-88128.post@n7.nabble.com> <1393174369381-89557.post@n7.nabble.com> <530A3B73.6010406@youmail.dk> <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> Message-ID: <1394197977787-89728.post@n7.nabble.com> Not there yet unfortunately, I decided to shift to Debian with KXStudio...and one of the first things I did was to try to compile the kernel with the quirk. hopefully it's not a hardware fault, but twice (and about halfway through the kernel rebuild) I lose power to my laptop, despite have a charged battery etc... I've emailed clemens, so hopefully the quirk is added to the kernel to make this easier in future..... I wish I could help to make the R16 quirk duplex, but that kind of thing is still beyond me at this stage cheers -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89728.html Sent from the linux-audio-user mailing list archive at Nabble.com. From jamesstewartmiller at gmail.com Fri Mar 7 13:27:58 2014 From: jamesstewartmiller at gmail.com (millerthegorilla) Date: Fri, 7 Mar 2014 05:27:58 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <1394197977787-89728.post@n7.nabble.com> References: <529CF0B8.6050304@youmail.dk> <1386262286208-88128.post@n7.nabble.com> <1393174369381-89557.post@n7.nabble.com> <530A3B73.6010406@youmail.dk> <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> <1394197977787-89728.post@n7.nabble.com> Message-ID: <1394198878158-89729.post@n7.nabble.com> Did anyone manage to advance the work on the zoom r16/24? Is it possible yet to use it in full duplex mode? Thanks James -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89729.html Sent from the linux-audio-user mailing list archive at Nabble.com. From nettings at stackingdwarves.net Fri Mar 7 13:28:13 2014 From: nettings at stackingdwarves.net (=?windows-1252?Q?J=F6rn_Nettingsmeier?=) Date: Fri, 07 Mar 2014 14:28:13 +0100 Subject: [LAU] ebur128 batch processing In-Reply-To: References: Message-ID: <5319C96D.8090606@stackingdwarves.net> On 03/07/2014 01:48 AM, Jarom?r Mike? wrote: > Hi, > > I will got about 100 stereo wav files mixed hopefully in similar way > (loudness). > I need them process to meet ebur128 specification. > True peaks -3dB > RMS -23dB > > Any chance to do it as batch process? > Result quality is critical as material will be broadcast on TV. hard problem. quickest solution: find somebody with a hardware leveller (j?nger etc.) and pipe it through. problem is that there is no straightforward way from an initial loudness measurement to a level change that predictably hits the -23 dbLUFS mark. can i ask which broadcaster mandates -3dB true peak? that seems quite conservative. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From jason at mancine.net Fri Mar 7 14:04:35 2014 From: jason at mancine.net (jmancine) Date: Fri, 7 Mar 2014 06:04:35 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <1394198878158-89729.post@n7.nabble.com> References: <1386262286208-88128.post@n7.nabble.com> <1393174369381-89557.post@n7.nabble.com> <530A3B73.6010406@youmail.dk> <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> <1394197977787-89728.post@n7.nabble.com> <1394198878158-89729.post@n7.nabble.com> Message-ID: <1394201075048-89731.post@n7.nabble.com> Still no duplex. In fact, no playback. I have narrowed the issue down to a problem with the nitrate being set to 32 instead of 24. This happens despite specifically setting 24 in the .formats section of the quirk. On the recording side it doesn't seem to matter because the device is doing the encoding at 24 and the PC is receiving at 32...not ideal, but it functions because 24 bits fit into 32. But for playback, the PC is sending a 32 bit stream to a device that can only decode 24...and it crashes. If you go back through this thread, I posted the various settings I tried for .formats -- there are likely many that I have not tested. Perhaps there is another way to force 24 bits -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89731.html Sent from the linux-audio-user mailing list archive at Nabble.com. From rncbc at rncbc.org Fri Mar 7 15:56:27 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Fri, 07 Mar 2014 15:56:27 +0000 Subject: [LAU] [ANN] Vee One Suite 0.4.0 - A proto-beta party! In-Reply-To: <53199ACD.2080800@gareus.org> References: <5318E6AA.4020102@rncbc.org> <53199ACD.2080800@gareus.org> Message-ID: <5319EC2B.207@rncbc.org> On 03/07/2014 10:09 AM, Robin Gareus wrote: > On 03/06/2014 10:20 PM, Rui Nuno Capela wrote: >> Howdy, >> >> as to no surprise (rly?) the so called Vee One Suite of old-schoolyards >> gets bullied down to another bump in the head: now turning into a >> proto-beta party? > > Thanks for all your work and organizing a party on top of it all! > > What's new in 0.4.0? - neither the email nor your blog mentions that. > Is there an easily accessible changelog? > news are mainly if not mostly about reverb being added to the stereo fx gang and max.stage time for envelope generators now variable in length--cf. Env.Time knob--was hard-coded/fixed to 5sec (synthv1, samplv1) and to 2sec (drumkv1) before, now the default setting. cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From shanipribadi at gmx.net Fri Mar 7 18:04:12 2014 From: shanipribadi at gmx.net (Shani Pribadi) Date: Sat, 8 Mar 2014 01:04:12 +0700 Subject: [LAU] ebur128 batch processing In-Reply-To: References: Message-ID: <20140307180411.GA23382@shanihplaptop> On Fri, Mar 07, 2014 at 01:48:04AM +0100, Jarom?r Mike? wrote: > Hi, > > I will got about 100 stereo wav files mixed hopefully in similar way > (loudness). > I need them process to meet ebur128 specification. > True peaks -3dB > RMS -23dB > > Any chance to do it as batch process? > Result quality is critical as material will be broadcast on TV. I have been using https://github.com/jiixyj/loudness-scanner to scan audio file and apply replaygain tag according to ebur128, although afaik there is no support for applying the gain to the file itself. You could apply the gain using sox and some scripting (theoretically). There is also http://r128gain.sourceforge.net/#apply that is said to be able to apply the gain in combination with sox, I haven't tried it though. From atte at youmail.dk Fri Mar 7 18:52:37 2014 From: atte at youmail.dk (Atte) Date: Fri, 07 Mar 2014 19:52:37 +0100 Subject: [LAU] lv2 synths in ardour3 Message-ID: <531A1575.2040501@youmail.dk> Hi I'm trying to get started using lv2 synths inside ardour. I have calf monosynth and reasonable sort of working. I use a2jmidid to get input from my usbkeyboard showing up in as jack midi. I then connected the usb keyboard to midi through in alsa tab in qjackctrl and midi through is bridged to jack midi. So I selected a2j as midi in put for the two synts. However they are both playing (makes sense, since they're both connected to the keyboard), but how can I handle them, for instance so that one only plays when I sit on the corresponding track? Regards -- Atte http://atte.dk http://modlys.dk From paul at linuxaudiosystems.com Fri Mar 7 19:01:39 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 7 Mar 2014 14:01:39 -0500 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: <531A1575.2040501@youmail.dk> References: <531A1575.2040501@youmail.dk> Message-ID: On Fri, Mar 7, 2014 at 1:52 PM, Atte wrote: > Hi > > I'm trying to get started using lv2 synths inside ardour. I have calf > monosynth and reasonable sort of working. I use a2jmidid to get input from > my usbkeyboard showing up in as jack midi. I then connected the usb > keyboard to midi through in alsa tab in qjackctrl and midi through is > bridged to jack midi. So I selected a2j as midi in put for the two synts. > However they are both playing (makes sense, since they're both connected to > the keyboard), but how can I handle them, for instance so that one only > plays when I sit on the corresponding track? > in the mixer strip for each track, near the top, is a green MIDI input button. click to disable input on that track. -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Fri Mar 7 19:22:56 2014 From: atte at youmail.dk (Atte) Date: Fri, 07 Mar 2014 20:22:56 +0100 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: References: <531A1575.2040501@youmail.dk> Message-ID: <531A1C90.4010108@youmail.dk> On 03/07/2014 08:01 PM, Paul Davis wrote: > in the mixer strip for each track, near the top, is a green MIDI input > button. click to disable input on that track. Thanks. I was on the right track - pun intended :-) Has it been suggested to implement some kind of link between instrument and track? I mean most of the time I guess most people would like to play/record only one midi instrument at the time, although recording a bunch is certainly possible. It would make the most used scenarios faster for most users... -- Atte http://atte.dk http://modlys.dk From paul at linuxaudiosystems.com Fri Mar 7 20:01:23 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 7 Mar 2014 15:01:23 -0500 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: <531A1C90.4010108@youmail.dk> References: <531A1575.2040501@youmail.dk> <531A1C90.4010108@youmail.dk> Message-ID: On Fri, Mar 7, 2014 at 2:22 PM, Atte wrote: > > Has it been suggested to implement some kind of link between instrument > and track? use 1 track per instrument plugin? -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Fri Mar 7 20:05:39 2014 From: atte at youmail.dk (Atte) Date: Fri, 07 Mar 2014 21:05:39 +0100 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: References: <531A1575.2040501@youmail.dk> Message-ID: <531A2693.50401@youmail.dk> On 03/07/2014 08:01 PM, Paul Davis wrote: > in the mixer strip for each track, near the top, is a green MIDI input > button. click to disable input on that track. Ok. When playback is stopped there's sound, when record is pressed on channel + at the top there's sound (and midi is recorded), but I can't figure out how to "jam along" without recording. Any hints? -- Atte http://atte.dk http://modlys.dk From atte at youmail.dk Fri Mar 7 20:11:39 2014 From: atte at youmail.dk (Atte) Date: Fri, 07 Mar 2014 21:11:39 +0100 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: References: <531A1575.2040501@youmail.dk> <531A1C90.4010108@youmail.dk> Message-ID: <531A27FB.4000804@youmail.dk> On 03/07/2014 09:01 PM, Paul Davis wrote: > On Fri, Mar 7, 2014 at 2:22 PM, Atte > wrote: > Has it been suggested to implement some kind of link between > instrument and track? > > use 1 track per instrument plugin? I'm not sure if you're proposing a solution for my workflow or asking about what I meant, maybe you could be a little bit more elaborate? -- Atte http://atte.dk http://modlys.dk From paul at linuxaudiosystems.com Fri Mar 7 20:29:48 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 7 Mar 2014 15:29:48 -0500 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: <531A27FB.4000804@youmail.dk> References: <531A1575.2040501@youmail.dk> <531A1C90.4010108@youmail.dk> <531A27FB.4000804@youmail.dk> Message-ID: On Fri, Mar 7, 2014 at 3:11 PM, Atte wrote: > On 03/07/2014 09:01 PM, Paul Davis wrote: > >> On Fri, Mar 7, 2014 at 2:22 PM, Atte > > wrote: >> Has it been suggested to implement some kind of link between >> instrument and track? >> >> use 1 track per instrument plugin? >> > > I'm not sure if you're proposing a solution for my workflow or asking > about what I meant, maybe you could be a little bit more elaborate? i'm offering the most obvious "link between instrument and track" that i can think of. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.santamauro at gmail.com Fri Mar 7 20:30:11 2014 From: david.santamauro at gmail.com (David Santamauro) Date: Fri, 07 Mar 2014 15:30:11 -0500 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: References: <531A1575.2040501@youmail.dk> <531A1C90.4010108@youmail.dk> Message-ID: <531A2C53.6060901@gmail.com> On 03/07/2014 03:01 PM, Paul Davis wrote: >> >> On Fri, Mar 7, 2014 at 2:22 PM, Atte > >> Has it been suggested to implement some kind of link between >> instrument and track? >> >> >> > use 1 track per instrument plugin? I'm pretty sure he means what I talked about a bit on ardour irc. Basically, selected track gets midi input-enabled. So instead of having to enable/disable every time you change tracks (which, if midi tracks, are usually different software or outboard instruments), you instead can simply select a track it will automatically be midi input enabled (and turning off all other tracks). It should obviously be a setting, but it would greatly help the midi workflow. David From atte at youmail.dk Fri Mar 7 21:33:06 2014 From: atte at youmail.dk (Atte) Date: Fri, 07 Mar 2014 22:33:06 +0100 Subject: [LAU] lv2 synths in ardour3 In-Reply-To: References: <531A1575.2040501@youmail.dk> <531A1C90.4010108@youmail.dk> <531A27FB.4000804@youmail.dk> <531A34F3.1010801@youmail.dk> Message-ID: <531A3B12.1030209@youmail.dk> On 03/07/2014 10:22 PM, Paul Davis wrote: > as happens so often with this stuff, user descriptions are > confused/confusing. Yeah, developing would be much more fun without users! > you could mean "a MIDI channel is tied to a MIDI track". you could mean > (as David mentioned) "the selected track receives MIDI". or you could > mean something else. I mean the same as David, then... > we will improve this situation in the not too distant future. Ok, great. Not sure if I'm around after march 26, though... -- Atte http://atte.dk http://modlys.dk From looplog at gmail.com Fri Mar 7 21:57:04 2014 From: looplog at gmail.com (michael noble) Date: Sat, 8 Mar 2014 06:57:04 +0900 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: <5319A1E4.1090206@gareus.org> References: <5319A1E4.1090206@gareus.org> Message-ID: On Fri, Mar 7, 2014 at 7:39 PM, Robin Gareus wrote: > PS. I think Ardour3 can provide meter-info per route via OSC. How could one go about verifying this? It's something I'd like to take advantage of if it's possible. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mira.mikes at gmail.com Fri Mar 7 22:00:44 2014 From: mira.mikes at gmail.com (=?ISO-8859-2?Q?Jarom=EDr_Mike=B9?=) Date: Fri, 7 Mar 2014 23:00:44 +0100 Subject: [LAU] ebur128 batch processing In-Reply-To: <5319C96D.8090606@stackingdwarves.net> References: <5319C96D.8090606@stackingdwarves.net> Message-ID: 2014-03-07 14:28 GMT+01:00 J?rn Nettingsmeier : > On 03/07/2014 01:48 AM, Jarom?r Mike? wrote: > >> Hi, >> >> I will got about 100 stereo wav files mixed hopefully in similar way >> (loudness). >> I need them process to meet ebur128 specification. >> True peaks -3dB >> RMS -23dB >> > > can i ask which broadcaster mandates -3dB true peak? that seems quite > conservative. > Czech TV ;) mira -------------- next part -------------- An HTML attachment was scrubbed... URL: From mira.mikes at gmail.com Fri Mar 7 22:19:28 2014 From: mira.mikes at gmail.com (=?ISO-8859-2?Q?Jarom=EDr_Mike=B9?=) Date: Fri, 7 Mar 2014 23:19:28 +0100 Subject: [LAU] ebur128 batch processing In-Reply-To: <20140307180411.GA23382@shanihplaptop> References: <20140307180411.GA23382@shanihplaptop> Message-ID: 2014-03-07 19:04 GMT+01:00 Shani Pribadi : > On Fri, Mar 07, 2014 at 01:48:04AM +0100, Jarom?r Mike? wrote: > > Hi, > > > > I will got about 100 stereo wav files mixed hopefully in similar way > > (loudness). > > I need them process to meet ebur128 specification. > > True peaks -3dB > > RMS -23dB > > > > Any chance to do it as batch process? > > Result quality is critical as material will be broadcast on TV. > > > There is also http://r128gain.sourceforge.net/#apply that is said to be > able to apply the gain in combination with sox, I haven't tried it > though. > I've just tried r128gain ... works fine ...there are "several presets", but not possible set "truepeaks" -3dB. Also not sure how it handle true peaks and RMS (not sure RMS is right term here). e.g. If RMS is corrected and gained to -23dB then True peaks are handled by limiter or just RMS is never raised to True peaks go over -1dB? regards mira -------------- next part -------------- An HTML attachment was scrubbed... URL: From gheskett at wdtv.com Fri Mar 7 22:56:06 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Fri, 7 Mar 2014 17:56:06 -0500 Subject: [LAU] ebur128 batch processing In-Reply-To: References: <5319C96D.8090606@stackingdwarves.net> Message-ID: <201403071756.06350.gheskett@wdtv.com> On Friday 07 March 2014 17:34:15 Jarom?r Mike? did opine: > 2014-03-07 14:28 GMT+01:00 J?rn Nettingsmeier > > > > On 03/07/2014 01:48 AM, Jarom?r Mike? wrote: > >> Hi, > >> > >> I will got about 100 stereo wav files mixed hopefully in similar way > >> (loudness). > >> I need them process to meet ebur128 specification. > >> True peaks -3dB > >> RMS -23dB > > > > can i ask which broadcaster mandates -3dB true peak? that seems quite > > conservative. > > Czech TV ;) > > mira Humm, that peak is 7db below the average vu meter reading here, and we won't even look at a board that will not handle 20-24db peaks, preferable a solid +30dbm. But now, with everything digital, it seems the producers are all living by the old AM motto, the louder the better, and I can quite easily hear digital clipping that is less than 6 db above the average. That makes it pretty darned obvious, particularly when their excrement production gear does a sign inversion in an add/mix stage to go with the peak. How then do they handle the noise floor when they are running rms 27db below a lot of the rest of the planet? That means their broadcast gear has to have a -83db noise floor to hold the mandated noise floor over here of -60db or better. Best I ever saw was -66 and that was with the video modulation removed on a UHF transmitter. We could do that to get thru a Proof of Performance on UHF because the UHF amplifiers in those days were all klystrons, which if the video was on had so much Incidental Carrier Phase Modulation that the true intercarrier measured noise was usually in the high -40's, to very low -50's if the visual wasn't tuned for best efficiency. That was not often done because it had a very noticeable effect on the monthly power bill, which at best was in the $10,000 USD/month range for most stations. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page NOTICE: Will pay 100 USD for an HP-4815A defective but complete probe assembly. From fons at linuxaudio.org Fri Mar 7 23:17:38 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Mar 2014 23:17:38 +0000 Subject: [LAU] ebur128 batch processing In-Reply-To: <201403071756.06350.gheskett@wdtv.com> References: <5319C96D.8090606@stackingdwarves.net> <201403071756.06350.gheskett@wdtv.com> Message-ID: <20140307231738.GA21734@linuxaudio.org> On Fri, Mar 07, 2014 at 05:56:06PM -0500, Gene Heskett wrote: > How then do they handle the noise floor when they are running rms 27db > below a lot of the rest of the planet? I suspect you are mixing up physical voltage levels in dBm and digital levels here (since 4 + 23 = 27). The -23 dB standard loudness reference of ebur128 is relative to digital peak level, not to 0 dBm. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From gheskett at wdtv.com Fri Mar 7 23:35:48 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Fri, 7 Mar 2014 18:35:48 -0500 Subject: [LAU] ebur128 batch processing In-Reply-To: <20140307231738.GA21734@linuxaudio.org> References: <201403071756.06350.gheskett@wdtv.com> <20140307231738.GA21734@linuxaudio.org> Message-ID: <201403071835.48845.gheskett@wdtv.com> On Friday 07 March 2014 18:33:36 Fons Adriaensen did opine: > On Fri, Mar 07, 2014 at 05:56:06PM -0500, Gene Heskett wrote: > > How then do they handle the noise floor when they are running rms 27db > > below a lot of the rest of the planet? > > I suspect you are mixing up physical voltage levels in dBm and > digital levels here (since 4 + 23 = 27). > > The -23 dB standard loudness reference of ebur128 is relative > to digital peak level, not to 0 dBm. > > Ciao, Ahh, gotcha Fons, so they are saying -23db below the digital clip level. That makes perfect sense in that context, thanks. Now if we could just get the production houses to adhere to that. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page NOTICE: Will pay 100 USD for an HP-4815A defective but complete probe assembly. From robin at gareus.org Sat Mar 8 00:15:32 2014 From: robin at gareus.org (Robin Gareus) Date: Sat, 08 Mar 2014 01:15:32 +0100 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: References: <5319A1E4.1090206@gareus.org> Message-ID: <531A6124.3020104@gareus.org> On 03/07/2014 10:57 PM, michael noble wrote: > On Fri, Mar 7, 2014 at 7:39 PM, Robin Gareus wrote: > >> PS. I think Ardour3 can provide meter-info per route via OSC. > > > How could one go about verifying this? It's something I'd like to take > advantage of if it's possible. > read the source? or just try it :) I was wrong. You can subscribe to receive async messages for the gain fader, but not signal-level (in Ardour3.5.357). Adding support to ardour to send meter-level notifications isnt't very hard, though. Find a very simple script to exercise Ardour's OSC interface attached. best, robin -------------- next part -------------- A non-text attachment was scrubbed... Name: osc_test.py Type: text/x-python Size: 2088 bytes Desc: not available URL: From looplog at gmail.com Sat Mar 8 00:52:17 2014 From: looplog at gmail.com (michael noble) Date: Sat, 8 Mar 2014 09:52:17 +0900 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: <531A6124.3020104@gareus.org> References: <5319A1E4.1090206@gareus.org> <531A6124.3020104@gareus.org> Message-ID: On Sat, Mar 8, 2014 at 9:15 AM, Robin Gareus wrote: > read the source? or just try it :) Yeah, it was a bit of a stupid/lazy question I realize. I've played around with Ardour's OSC support with Pd before, but didn't know there was metering capabilities. Alas... -------------- next part -------------- An HTML attachment was scrubbed... URL: From shanipribadi at gmx.net Sat Mar 8 06:59:11 2014 From: shanipribadi at gmx.net (Shani Pribadi) Date: Sat, 8 Mar 2014 13:59:11 +0700 Subject: [LAU] ebur128 batch processing In-Reply-To: References: <20140307180411.GA23382@shanihplaptop> Message-ID: <20140308065911.GA6223@shanihplaptop> On Fri, Mar 07, 2014 at 11:19:28PM +0100, Jarom?r Mike? wrote: > I've just tried r128gain ... works fine ...there are "several presets", > but not possible set "truepeaks" -3dB. > Also not sure how it handle true peaks and RMS (not sure RMS is right term > here). > e.g. If RMS is corrected and gained to -23dB then True peaks are handled > by limiter or just RMS is never raised to True peaks go over -1dB? > I think r128gain is more like a scanner that outputs the statistic to another processor (in this case sox/ffmpeg). The method used here ("Applying the gain") does not include a limiter in the signal chain. The TGDB field means the gain needed so that Loudness is same as reference (-23LUFS). It also outputs TPDB (maximum truepeak). Since you wanted max true peak at -3dB, for cases where TPDB+TGDB < -3dB the TGDB should be safe to be applied directly as is (if you want to be sure, scan one more time after applying gain). For cases where TPDB+TGDB > -3dB you would need limiter to reduce the peaks. How much limiting should be applied cannot be determined solely from TGDB or TPDB, so my suggestion is applying limiter at let's say TPDB-3-TGDB and then run the scan again. If TPDB+TGDB is still over -3dB then run the limiter again with lower threshold. Most materials that have gone through mastering process would likely be in the first case. Does the program material should have Loudness exactly at -23LUFS or the maximum Loudness allowed is at -23LUFS (so program loudness < -23LUFS)? If it's okay for the Loudness to be less than -23LUFS then you could skip the limiter and just reduce the gain. From lmemsm at gmail.com Sat Mar 8 17:07:08 2014 From: lmemsm at gmail.com (LM) Date: Sat, 8 Mar 2014 12:07:08 -0500 Subject: [LAU] Need suggestions for lightweight audio programs Message-ID: I'm trying to find lightweight compiled audio programs that will work well on older computers or computers with limited resources. I'd like to add more music resource recommendations to the Schoolforge web site ( https://www.schoolforge.net/ ) and think it would be useful for students or schools working with older or second-hand computers. As a starting point, I'll list some of the programs I've run across or used that might meet the needs of students or users wanting to develop/create/edit music on older machines. I've tried to concentrate on command line, ncurses, SDL and FLTK based applications since many are lightweight enough to work well on older or low memory machines. If you have recommendations for other programs not listed, would appreciate very much if you would let me know. abcmidi - abc notation to midi and midi to abc conversion: http://abc.sourceforge.net/abcMIDI/ abcm2ps - abc notation to sheet music in PDF format: http://moinejf.free.fr/ TiMidity++ - midi/midi Karaoke player, converts midi to wave using soundfonts/gus patches: http://timidity.sourceforge.net/ Milkytracker - mod file creator/player: http://www.milkytracker.org/ sox - Sound eXchange: http://sox.sourceforge.net/ gramofile - record records and other input, has pop filter: http://www.opensourcepartners.nl/~costar/gramofile/ Wave utilities: http://billposer.org/Software/waveutils.html WaveProject: http://sourceforge.net/projects/waveproject/ Soundtouch library and program to change tempo, pitch and playback rates (library used by audacity): http://www.surina.net/soundtouch/ Xiph.org has several useful command line programs and libraries: https://www.xiph.org/downloads/ fast and light recorder, FLTK front end for sox: http://www.matteolucarelli.net/flrec/index_en.htm FLTK audio editor (I'm working on some patches to get it to compile with FLTK 1.3.2 and later compilers and possibly add some new features.): http://sourceforge.net/projects/apcstudio/ audio examples using SDL-widgets GUI library: http://members.chello.nl/w.boeke/SDL-widgets/ alsamixergui, FLTK front end to mixer: https://packages.debian.org/wheezy/alsamixergui umix, ncurses mixer for Unix and Linux systems: http://umix.sourceforge.net/ From ralf.mardorf at rocketmail.com Sat Mar 8 17:30:26 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Mar 2014 18:30:26 +0100 Subject: [LAU] Need suggestions for lightweight audio programs In-Reply-To: References: Message-ID: <1394299826.587.117.camel@archlinux> On Sat, 2014-03-08 at 12:07 -0500, LM wrote: > I'm trying to find lightweight compiled audio programs that will work > well on older computers or computers with limited resources. Users can get the source code and compile for what ever architecture they like, enabling, disabling what ever they like. Assumed you're asking for binaries, than use a distro with 32-bit ports, with software compiled for very old CPUs. Debian 32-bit does provide support for very old CPUs. "2.1.2.1. CPU Nearly all x86-based (IA-32) processors still in use in personal computers are supported, including all varieties of Intel's "Pentium" series. This also includes 32-bit AMD and VIA (former Cyrix) processors, and processors like the Athlon XP and Intel P4 Xeon. However, Debian GNU/Linux wheezy will not run on 386 or earlier processors. Despite the architecture name "i386", support for actual 80386 processors (and their clones) was dropped with the Sarge (r3.1) release of Debian[2]. (No version of Linux has ever supported the 286 or earlier chips in the series.) All i486 and later processors are still supported[3]." - http://www.debian.org/releases/stable/i386/ch02s01.html.en As WM I currently prefer JWM on a modern machine, I'm using it with Debian 32-bit and Arch 64-bit and it should be lightweighted enough for old machines. http://joewing.net/projects/jwm/ Once a lightweighted WM is installed and assumed the distro's repositories provide software compiled for old CPUs, users should simply test what software is usable and what software is unusable on their machines. From len at ovenwerks.net Sat Mar 8 17:44:05 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 8 Mar 2014 09:44:05 -0800 (PST) Subject: [LAU] Need suggestions for lightweight audio programs In-Reply-To: References: Message-ID: On Sat, 8 Mar 2014, LM wrote: > I'm trying to find lightweight compiled audio programs that will work > well on older computers or computers with limited resources. I'd like There do seem to be some limiting factors. I did try putting a headless audio machine together on an older machine (P 300). The CPU seems to be up to it, but my memory of 196M will not properly run jackd (pulse runs fine) so there are some minimum requirements for some things. It may be best to stay with ALSA for I/O as pulse tends to dropouts. I have been unable to find more memory for this machine... I have lots that fit in the slot, but do not show up. Yet I can use the memory from this machine in MB that the other memory works with. I gave up because jackd (netjack) was part of what I wanted to play with. I think 1/4G ram ends up being minimum for most things. For headless, or even terminal operation, I would use a text session manager (like screen) rather than VTs as it allows using one instance of dbus for all bash instances. -- Len Ovens www.ovenwerks.net From rosea.grammostola at gmail.com Sat Mar 8 22:08:13 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sat, 8 Mar 2014 23:08:13 +0100 Subject: [LAU] REV-plugins / AMS moog Message-ID: The update of the REV-plugins seems to break the nicest instrument patch in AMS, the Moog instrument :( It asks for g2reverb which it can't find. \r -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sun Mar 9 00:01:49 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 9 Mar 2014 00:01:49 +0000 Subject: [LAU] REV-plugins / AMS moog In-Reply-To: References: Message-ID: <20140309000149.GA30844@linuxaudio.org> On Sat, Mar 08, 2014 at 11:08:13PM +0100, rosea grammostola wrote: > The update of the REV-plugins seems to break the nicest instrument patch in > AMS, the Moog instrument :( > > It asks for g2reverb which it can't find. Open the ams file in a text editor, search for a a line that contains g2reverb G2reverb and change this to zita-reverbs G2reverb Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rosea.grammostola at gmail.com Sun Mar 9 00:37:38 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 9 Mar 2014 01:37:38 +0100 Subject: [LAU] REV-plugins / AMS moog In-Reply-To: <20140309000149.GA30844@linuxaudio.org> References: <20140309000149.GA30844@linuxaudio.org> Message-ID: Thanks! The question is whether it is wise to change the name of something like this ... I don't have enough developer knowledge to give the answer. I can only experience the problem as a user. The fix is quite easy, that's the good part, but these things are still annoying. Best regards, \r On Sun, Mar 9, 2014 at 1:01 AM, Fons Adriaensen wrote: > On Sat, Mar 08, 2014 at 11:08:13PM +0100, rosea grammostola wrote: > > The update of the REV-plugins seems to break the nicest instrument patch > in > > AMS, the Moog instrument :( > > > > It asks for g2reverb which it can't find. > > > Open the ams file in a text editor, search for a a line that > contains > > g2reverb G2reverb > > and change this to > > zita-reverbs G2reverb > > Ciao, > > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sun Mar 9 00:58:54 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 9 Mar 2014 00:58:54 +0000 Subject: [LAU] REV-plugins / AMS moog In-Reply-To: References: <20140309000149.GA30844@linuxaudio.org> Message-ID: <20140309005854.GD30844@linuxaudio.org> On Sun, Mar 09, 2014 at 01:37:38AM +0100, rosea grammostola wrote: > Thanks! The question is whether it is wise to change the name of something > like this ... I don't have enough developer knowledge to give the answer. I > can only experience the problem as a user. > > The fix is quite easy, that's the good part, but these things are still > annoying. The problem here is that the original name of the plugin file (g2reverb) was a bad choice. You could also try to create a (symbolic or real) link g2reverb.so -> zita-reverbs.so. This will probably work with AMS, but may upset other apps which rely on the unique number and will find duplicates. There seems to be little maintenance of AMS. The latest version I could find still depends on libclalsadrv while this has been deprecated for some time, and the changes required to use libzita-alsa-pcmi instead are really trivial. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rosea.grammostola at gmail.com Sun Mar 9 09:43:07 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 9 Mar 2014 10:43:07 +0100 Subject: [LAU] REV-plugins / AMS moog In-Reply-To: <20140309005854.GD30844@linuxaudio.org> References: <20140309000149.GA30844@linuxaudio.org> <20140309005854.GD30844@linuxaudio.org> Message-ID: I use this version with nsm support https://github.com/royvegard/ams IIRC both libclalsadrv and libzita-alsa-pcmi could be used for building On Sun, Mar 9, 2014 at 1:58 AM, Fons Adriaensen wrote: > On Sun, Mar 09, 2014 at 01:37:38AM +0100, rosea grammostola wrote: > > Thanks! The question is whether it is wise to change the name of > something > > like this ... I don't have enough developer knowledge to give the > answer. I > > can only experience the problem as a user. > > > > The fix is quite easy, that's the good part, but these things are still > > annoying. > > The problem here is that the original name of the plugin file (g2reverb) > was a bad choice. > > You could also try to create a (symbolic or real) link g2reverb.so -> > zita-reverbs.so. > > This will probably work with AMS, but may upset other apps which rely > on the unique number and will find duplicates. > > There seems to be little maintenance of AMS. The latest version > I could find still depends on libclalsadrv while this has been > deprecated for some time, and the changes required to use > libzita-alsa-pcmi instead are really trivial. > > Ciao, > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Sun Mar 9 09:52:26 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 9 Mar 2014 10:52:26 +0100 Subject: [LAU] REV-plugins / AMS moog In-Reply-To: References: <20140309000149.GA30844@linuxaudio.org> <20140309005854.GD30844@linuxaudio.org> Message-ID: Hm I still think that renaming should only be done when strictly needed. I can't see the need of this change tbh. Large part of the community has the patches with g2reverb, I assume more patches (also in other apps like jack-rack) exists. So every single member the community has to adjust the patches. When you install it on an other computer, you've to find the adjusted patch or adjust again. This missing g2reverb in AMS killed my workflow yesterday already, I could not make the moog patch work Googling 15min IRC asking 15 min Googling 10 min Reinstalling AMS + depends 20 min Ask question on KXstudio forum 10 min Got close to an answer 45 min later Writing email LAU 5 min Waiting for an answer .... Adjusting patch ... At the end I did nothing with AMS, had to go... On Sun, Mar 9, 2014 at 10:43 AM, rosea grammostola < rosea.grammostola at gmail.com> wrote: > I use this version with nsm support > https://github.com/royvegard/ams > > IIRC both libclalsadrv and libzita-alsa-pcmi could be used for building > > > On Sun, Mar 9, 2014 at 1:58 AM, Fons Adriaensen wrote: > >> On Sun, Mar 09, 2014 at 01:37:38AM +0100, rosea grammostola wrote: >> > Thanks! The question is whether it is wise to change the name of >> something >> > like this ... I don't have enough developer knowledge to give the >> answer. I >> > can only experience the problem as a user. >> > >> > The fix is quite easy, that's the good part, but these things are still >> > annoying. >> >> The problem here is that the original name of the plugin file (g2reverb) >> was a bad choice. >> >> You could also try to create a (symbolic or real) link g2reverb.so -> >> zita-reverbs.so. >> >> This will probably work with AMS, but may upset other apps which rely >> on the unique number and will find duplicates. >> >> There seems to be little maintenance of AMS. The latest version >> I could find still depends on libclalsadrv while this has been >> deprecated for some time, and the changes required to use >> libzita-alsa-pcmi instead are really trivial. >> >> Ciao, >> >> -- >> FA >> >> A world of exhaustive, reliable metadata would be an utopia. >> It's also a pipe-dream, founded on self-delusion, nerd hubris >> and hysterically inflated market opportunities. (Cory Doctorow) >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Sun Mar 9 09:54:08 2014 From: atte at youmail.dk (Atte) Date: Sun, 09 Mar 2014 10:54:08 +0100 Subject: [LAU] compressor with visual feedback in non Message-ID: <531C3A40.70607@youmail.dk> Hi I'm playing with non. In ardour I use the SC4 mono compressor alot, partly because I like the visual feedback of the two meters "amplitude" and "gain reduction", which is really helpful when finding that sweet spot where the compressor starts to kick in. However those meters are missing in Is there a way to get these meters in non? Is there another compressor that works with non with this kind of visual feedback? -- Atte http://atte.dk http://modlys.dk From fons at linuxaudio.org Sun Mar 9 13:04:55 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 9 Mar 2014 13:04:55 +0000 Subject: [LAU] plugin updates Message-ID: <20140309130455.GA13010@linuxaudio.org> Some updates to REV-plugins-0.7.1 * Removed G2reverb from the .so, see below. g2reverb-0.7.1 * Contains the G2reverb plugin from earlier REV-plugins. * Plugin file name is now the same as in the original (2003) version, so AMS patches using this should load without error. WAH-plugins-0.1.0 * Some cleanup, unique ID of the auto-wah changed to 1949 to avoid conflict with recent stereo panner plugins. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From harryhaaren at gmail.com Sun Mar 9 15:24:03 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sun, 9 Mar 2014 15:24:03 +0000 Subject: [LAU] OpenAV : ArtyFX 1.1 Message-ID: Hey All, Its my pleasure to announce ArtyFX 1.1, with three new plugins! The plugin are distortion, feedback delay, and 4-band eq. Demo video: http://www.youtube.com/watch?v=aPQOZK-yKy8 ArtyFX page: http://openavproductions.com/artyfx/ OpenAV wishes to thank Steve Harris for authoring Barry's Satan Maximizer: Satma's DSP routine is derived from that work. OpenAV also wishes to thank Fons Adriaensen for writing the 4-band parameteric equalizer, as Kuiza uses his implementation as DSP routine. Contributions to release ArtyFX 1.1 welcomed, see the ArtyFX page for details: http://openavproductions.com/artyfx Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Mar 10 08:34:54 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 10 Mar 2014 09:34:54 +0100 Subject: [LAU] compressor with visual feedback in non In-Reply-To: <531C3A40.70607@youmail.dk> References: <531C3A40.70607@youmail.dk> Message-ID: <531D792E.1030103@autostatic.com> On 03/09/2014 10:54 AM, Atte wrote: > Hi > > I'm playing with non. In ardour I use the SC4 mono compressor alot, > partly because I like the visual feedback of the two meters "amplitude" > and "gain reduction", which is really helpful when finding that sweet > spot where the compressor starts to kick in. However those meters are > missing in Is there a way to get these meters in non? Is there another > compressor that works with non with this kind of visual feedback? > Hello Atte, Non Mixer only supports LADSPA so there is no way to get the meters into Non because those are part of the LV2 port of SC4. Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From tim at quitte.de Mon Mar 10 08:52:27 2014 From: tim at quitte.de (Tim Goetze) Date: Mon, 10 Mar 2014 09:52:27 +0100 (CET) Subject: [LAU] compressor with visual feedback in non In-Reply-To: <531D792E.1030103@autostatic.com> References: <531C3A40.70607@youmail.dk> <531D792E.1030103@autostatic.com> Message-ID: [Jeremy Jongepier] >Non Mixer only supports LADSPA so there is no way to get the meters into >Non because those are part of the LV2 port of SC4. $ analyseplugin /usr/lib/ladspa/sc4_1882.so|grep output|grep control "Amplitude (dB)" output, control, -40 to 12 "Gain reduction (dB)" output, control, -24 to 0 As you can see, those outputs are available on the LADSPA version as well. It would appear the host in question doesn't provide means of presenting these ports to the user. Cheers, Tim From jeremy at autostatic.com Mon Mar 10 09:08:24 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 10 Mar 2014 10:08:24 +0100 Subject: [LAU] compressor with visual feedback in non In-Reply-To: References: <531C3A40.70607@youmail.dk> <531D792E.1030103@autostatic.com> Message-ID: <531D8108.9000907@autostatic.com> On 03/10/2014 09:52 AM, Tim Goetze wrote: > [Jeremy Jongepier] >> Non Mixer only supports LADSPA so there is no way to get the meters into >> Non because those are part of the LV2 port of SC4. > > $ analyseplugin /usr/lib/ladspa/sc4_1882.so|grep output|grep control > "Amplitude (dB)" output, control, -40 to 12 > "Gain reduction (dB)" output, control, -24 to 0 > > As you can see, those outputs are available on the LADSPA version as > well. It would appear the host in question doesn't provide means of > presenting these ports to the user. > > Cheers, Tim > Thanks Tim for clearing that up. I wasn't aware that LADSPA exposed ports this way and that they could be used to visualize the output values. Another thing learned :) Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From jamesstewartmiller at gmail.com Mon Mar 10 10:00:58 2014 From: jamesstewartmiller at gmail.com (millerthegorilla) Date: Mon, 10 Mar 2014 03:00:58 -0700 (PDT) Subject: [LAU] re Zoom R16 In-Reply-To: <1394201075048-89731.post@n7.nabble.com> References: <1393174369381-89557.post@n7.nabble.com> <530A3B73.6010406@youmail.dk> <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> <1394197977787-89728.post@n7.nabble.com> <1394198878158-89729.post@n7.nabble.com> <1394201075048-89731.post@n7.nabble.com> Message-ID: Hmm, I have been compiling my kernel for a couple of days now and no joy. I have the information from lsusb -vv, cat /proc/asound/R16/stream0 / 1, etc but I'm a little confused about the terminology of quirks-table.h. What is the difference between .ifnum and .iface? is the first (ifnum) a reference to the usb bus? When I put the working zoom-quirks.h through, I get capture on interface 1 and playback on interface 2 listed under one stream in cat /proc/asound/R16/stream0 , when I try the zoom-quirks.h that I include below, I get two playback streams, one listed at /proc/asound/R16/stream0 and the other at /proc/asound/R16/stream1 Notice that I have put a bunch of sample formats so that the correct one can be selected. I don't think that this covers the problem though if you read in my excerpt from kern.log, it complains that ep 3 can set freq. here is the zoom-quirks.h quirks table. { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE(0x1686, 0x00dd), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE), .channels = 2, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x03, /*PLAYBACK*/ .ep_attr = USB_ENDPOINT_XFER_ISOC, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 3, .type = QUIRK_MIDI_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE), .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x84, /*CAPTURE*/ .ep_attr = USB_ENDPOINT_XFER_ISOC, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = .1 }, } } }, here is the excerpt from kern.log Mar 10 09:35:17 super-R720 kernel: [ 98.846039] usb 1-1: new high-speed USB device number 4 using ehci-pci Mar 10 09:35:17 super-R720 kernel: [ 98.974688] usb 1-1: config 1 interface 3 altsetting 0 bulk endpoint 0x1 has invalid maxpacket 64 Mar 10 09:35:17 super-R720 kernel: [ 98.974693] usb 1-1: config 1 interface 3 altsetting 0 bulk endpoint 0x82 has invalid maxpacket 64 Mar 10 09:35:17 super-R720 kernel: [ 98.975307] usb 1-1: New USB device found, idVendor=1686, idProduct=00dd Mar 10 09:35:17 super-R720 kernel: [ 98.975310] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Mar 10 09:35:17 super-R720 kernel: [ 98.975312] usb 1-1: Product: R16 Mar 10 09:35:17 super-R720 kernel: [ 98.975314] usb 1-1: Manufacturer: ZOOM Corporation Mar 10 09:35:17 super-R720 kernel: [ 98.975316] usb 1-1: SerialNumber: 0 Mar 10 09:35:22 super-R720 kernel: [ 104.032341] 4:1:1: cannot get freq at ep 0x3 Mar 10 09:35:27 super-R720 kernel: [ 109.033070] usbcore: registered new interface driver snd-usb-audio I'm fairly certain that I've got the wrong interface number but I have found it virtually impossible to find any documentation on the nomenclature used in quirks-table.h. For instance what is the difference between ifnum and iface? Also, I notice that several source code files, like stream.c for instance, have quirks built in to them, so perhaps I could explicitly set the sample rate format for the playback stream when the stream is created in the source code. Otherwise the bitrate descriptors are listed around about here: www.cs.fsu.edu/~baker/devices/lxr/http/source/linux/include/sound/asound.h#L203 I'm guessing that where a bit rate descriptor has the letter S it stands for signed and the one with U stands for unsigned, I don't know if that would make a difference. My brief foray into the source code so far leaves me with the impression that I should be able to create both capture and playback on the same interface, using the alt setting, but how the format works for that is beyond me at the moment, although I plan to spend tomorrow morning reading through the source code!. On Fri, Mar 7, 2014 at 2:04 PM, jmancine [via Linux Audio] < ml-node+s4202n89731h95 at n7.nabble.com> wrote: > Still no duplex. In fact, no playback. > > I have narrowed the issue down to a problem with the nitrate being set to > 32 instead of 24. This happens despite specifically setting 24 in the > .formats section of the quirk. On the recording side it doesn't seem to > matter because the device is doing the encoding at 24 and the PC is > receiving at 32...not ideal, but it functions because 24 bits fit into 32. > But for playback, the PC is sending a 32 bit stream to a device that can > only decode 24...and it crashes. > > If you go back through this thread, I posted the various settings I tried > for .formats -- there are likely many that I have not tested. Perhaps > there is another way to force 24 bits > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89731.html > To unsubscribe from re Zoom R16, click here > . > NAML > -- James Stewart Miller Bsc(hons) Psych. -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89770.html Sent from the linux-audio-user mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From brummer- at web.de Mon Mar 10 10:41:00 2014 From: brummer- at web.de (hermann meyer) Date: Mon, 10 Mar 2014 11:41:00 +0100 Subject: [LAU] compressor with visual feedback in non In-Reply-To: <531D8108.9000907@autostatic.com> References: <531C3A40.70607@youmail.dk> <531D792E.1030103@autostatic.com> <531D8108.9000907@autostatic.com> Message-ID: <531D96BC.5070402@web.de> Am 10.03.2014 10:08, schrieb Jeremy Jongepier: > On 03/10/2014 09:52 AM, Tim Goetze wrote: >> [Jeremy Jongepier] >>> Non Mixer only supports LADSPA so there is no way to get the meters into >>> Non because those are part of the LV2 port of SC4. >> $ analyseplugin /usr/lib/ladspa/sc4_1882.so|grep output|grep control >> "Amplitude (dB)" output, control, -40 to 12 >> "Gain reduction (dB)" output, control, -24 to 0 >> >> As you can see, those outputs are available on the LADSPA version as >> well. It would appear the host in question doesn't provide means of >> presenting these ports to the user. >> >> Cheers, Tim >> > Thanks Tim for clearing that up. I wasn't aware that LADSPA exposed > ports this way and that they could be used to visualize the output > values. Another thing learned :) > > Bye, > > Jeremy > Check out guitarix for example, :-D it presenting LADSPA output control ports as type "Display" to the user. the above named ports be there. greets hermann From jason at mancine.net Mon Mar 10 14:25:52 2014 From: jason at mancine.net (jmancine) Date: Mon, 10 Mar 2014 07:25:52 -0700 (PDT) Subject: [LAU] re Zoom R16 In-Reply-To: References: <530A3B73.6010406@youmail.dk> <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> <1394197977787-89728.post@n7.nabble.com> <1394198878158-89729.post@n7.nabble.com> <1394201075048-89731.post@n7.nabble.com> Message-ID: <1394461552782-89773.post@n7.nabble.com> A few things: First, many of my original posts to the mailing list did not actually end up on the mailing list! You can go back through here to see the full history of how we got to where we are: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-td87487.html Second, we should all be aware that kernels 3.11 and greater have support for the "ANY INTERFACE" and "AUTODETECT" tags. For the R16, this means that this quirk (yes that is the whole thing!) will work for capture only. /* ZOOM R16 in USB 2.0 mode */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | USB_DEVICE_ID_MATCH_INT_CLASS, .idVendor = 0x1686, .bInterfaceClass = USB_CLASS_VENDOR_SPEC, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUTODETECT } }, If you are after capture only, and are on a kernel >3.11, this is your quirk. Third, the problem with playback is (I believe) narrowed down to ALSA not being able to set set the bit rate. Until we are able to figure out how to make ALSA see the R16 as 24-bit integer, everything else is irrelevant. millerthegorilla wrote > Notice that I have put a bunch of sample formats so that the correct one > can be selected. The R16 does not support 8, 16 or 32... it is only 24 (in interface mode). The problem is that none of the 24 bit formats I am aware of WORK. ALSA has some kind of problem with the R16 when it comes to initializig it, and always sets it to 32 bit (or technically 24 in 32) which it does not support. It is 24 bit INTEGER meaning exactly 24 bits. Again, until we can find a .format that will set the R16 to 24 bits, it will not support playback. Fourth, on the subject of ifnums and ifaces: millerthegorilla wrote > Hmm, I have been compiling my kernel for a couple of days now and no joy. > I have the information from lsusb -vv, cat /proc/asound/R16/stream0 / 1, > etc but I'm a little confused about the terminology of quirks-table.h. > What is the difference between .ifnum and .iface? is the first (ifnum) > a > reference to the usb bus? .ifnum is the hardware interface number, it can be derived from the variable "bInterfaceNumber" in the output of lsusb -v Here is the problem.... there is NO interface #4 despite the quirk working as such. When I initially was testing quirks for the R16, I just added that (and #5) as a placeholders on top of the standard audio quirk! I was as surprised as everyone when #4 enabled capture. It is a working hack, but it is accidental. To my knowledge, these are the correct interface numbers and functions: The correct interfaces are: 0 - device (hardware comm) 1 - PLAYBACK 2 - CAPTURE 3 - MIDI If you move your .ifnum=4 section into the .ifnum=2 section (and delete #4) it will work, and will be in the right place for further testing. Similarly, .iface correlates with the iInterface variable that you can also view in the output of lsusb. The correct setting is zero, like this: .iface = 0,' I noticed you also changed .attributes to USB_ENDPOINT_XFER_ISOC... any reason for that? And finally, here is the THEORETICAL quirk that *should* work for playback, capture and MIDI. By all accounts, this *should* work... BUT IT DOES NOT. My best guess is that this is indeed the right quirk, but that the .formats setting of 24 bits is not accepted and ALSA defaults to 32 (or 24 packed in 32). { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE(0x1686, 0x00dd), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, /*PLAYBACK*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 2, .iface = 0, .altsetting = 1, altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x03, .ep_attr = 9, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 2, /*CAPTURE*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x84, .ep_attr = 13, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 3, .type = QUIRK_MIDI_STANDARD_INTERFACE }, { .ifnum = .1 }, } } }, -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89773.html Sent from the linux-audio-user mailing list archive at Nabble.com. From jason at mancine.net Mon Mar 10 14:29:00 2014 From: jason at mancine.net (jmancine) Date: Mon, 10 Mar 2014 07:29:00 -0700 (PDT) Subject: [LAU] re Zoom R16 In-Reply-To: <1394461552782-89773.post@n7.nabble.com> References: <1393184935653-89561.post@n7.nabble.com> <530AEA41.1090909@youmail.dk> <1393278643930-89584.post@n7.nabble.com> <530F1694.7090205@youmail.dk> <1393538864605-89649.post@n7.nabble.com> <1394197977787-89728.post@n7.nabble.com> <1394198878158-89729.post@n7.nabble.com> <1394201075048-89731.post@n7.nabble.com> <1394461552782-89773.post@n7.nabble.com> Message-ID: <1394461740905-89775.post@n7.nabble.com> Sorry, there was a typo in that last message...there is no apostrophe after the .iface setting It should be just: .iface = 0, -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89775.html Sent from the linux-audio-user mailing list archive at Nabble.com. From rmouneyres at gmail.com Mon Mar 10 22:00:32 2014 From: rmouneyres at gmail.com (raf) Date: Mon, 10 Mar 2014 23:00:32 +0100 Subject: [LAU] Jack vu-meter to OSC In-Reply-To: References: <5319A1E4.1090206@gareus.org> Message-ID: Hello again, replying back here to mentioned that instead of adding OSC support to your (lightweight and efficient) program, i can simply redirect stdout output to a "file" and read that back from my perl script. Then as my script is already running my main OSC server, i simply need to parse the read info and store it into my project. I just realize that i hadn't explained exactly what i needed. Now, jack-peak has became really handy ! Rapha?l Le 7 mars 2014 ? 11:54, Rapha?l Mouneyres a ?crit : > looks very promising for my use. digital-peak with [0,1] output is > what i need to output on OSC. > The code is short and clean enough for me to read, I should be able to > add the OSC server output with a new -o option. I'll let you know if > i'm going for it and achieve something working. > > thanks robin ! > Rapha?l > > 2014-03-07 11:39 UTC+01:00, Robin Gareus : >> On 03/07/2014 11:28 AM, Rapha?l Mouneyres wrote: >>> Hello, >>> >>> Does someone know about a jack client app which is able to publish >>> vu-metering/amplitude of audio jack ports signals as an OSC server ? >>> I mean an app like meterbridge but with OSC output instead of graphical >>> output. >> >> Not exactly. It's JSON output (for use with a webserver) and >> digital-peak (not VU): http://gareus.org/gitweb/?p=jack-peak.git >> >> ciao, >> robin >> >> PS. I think Ardour3 can provide meter-info per route via OSC. >> From piem at piem.org Wed Mar 12 19:12:05 2014 From: piem at piem.org (Paul Brossier) Date: Wed, 12 Mar 2014 20:12:05 +0100 Subject: [LAU] [LAA] aubio 0.4.1 Message-ID: <20140312191205.GA17497@coconut.piem.org> Hi all, A new version of aubio, 0.4.1, is out. aubio is a library of functions to perform audio feature extractions such as: - note onset detection - pitch detection - beat tracking - MFCC computation - spectral descriptors This version is mostly focusing on media file input and output. Here is a quick overview of the changes. The most interesting feature in this release concerns aubiocut. Thanks to the sponsoring of Mark Suppes, the python script to slice sound steams was extended to be sample accurate, cut overlapping segments, and work on multiple channels. New source and sink objects have been added to let aubio read and write WAV files, even when built with no external libraries. This should simplify the use of aubio on platforms such as Android or Windows. Existing sources and sinks have been extended to read and write from and to multiple channels. This makes python-aubio one of the fastest and most versatile Python module to read and write media files. This release also comes with a stack of bug fixes and code clean-ups. Note: this version is API and ABI compatible with 0.4.0. Since it only adds new features to the existing interface, your existing source and binary code will keep working without any modifications. To find out more about aubio and this release: Project homepage: http://aubio.org/ Post announcing aubio 0.4.1: http://aubio.org/news/20140312-1953_aubio_0.4.1 ChangeLog for aubio 0.4.1: http://aubio.org/pub/aubio-0.4.1.changelog Source tarball, signature and digests: http://aubio.org/pub/aubio-0.4.1.tar.bz2 http://aubio.org/pub/aubio-0.4.1.tar.bz2.asc http://aubio.org/pub/aubio-0.4.1.tar.bz2.md5 http://aubio.org/pub/aubio-0.4.1.tar.bz2.sha1 API Documentation: http://aubio.org/doc/latest/ Happy hacking! Paul From beatleboy07 at gmail.com Fri Mar 14 01:52:47 2014 From: beatleboy07 at gmail.com (Clifford Dunn) Date: Thu, 13 Mar 2014 18:52:47 -0700 Subject: [LAU] M-Audio 88es Keyboard Message-ID: Hi List, This probably isnt the right place, but I'm kind of lost on where to go from here: I have an M-Audio Keystation 88es Keyboard. I plug it into my laptop running KxStudio through USB, and pretty much works perfectly out of the box. I've had a lot of success running it into LinuxSampler and running that into Ardour. I pretty much only feel limited by the sample library I have there. That and XRuns. I'm still working on reducing those. If I increase my buffer size in Jack, the keyboard isn't quite as responsive (probably obvious there). I figure I'll eventually find the sweet spot... What I do hope someone can help me with is a simple problem. When I plug a foot pedal into the sustain port of the keyboard, it doesn't add any change to the signal. The only other experience I have is using this keyboard with Logic, Garageband, and Main Stage. The sustain worked like a sustain would in these programs. I wonder if someone could point me towards unlocking this functionality. Google searches aren't too helpful with this. Thanks! Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn From alexandre.prokoudine at gmail.com Fri Mar 14 07:52:56 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Fri, 14 Mar 2014 11:52:56 +0400 Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: 14 ????? 2014 ?. 5:53 ???????????? "Clifford Dunn" ???????: > What I do hope someone can help me with is a simple problem. When I > plug a foot pedal into the sustain port of the keyboard, it doesn't > add any change to the signal. The only other experience I have is > using this keyboard with Logic, Garageband, and Main Stage. The > sustain worked like a sustain would in these programs. I wonder if > someone could point me towards unlocking this functionality. Google > searches aren't too helpful with this. Typically the problem appears when you plug the pedal into an already connected MIDI keyboard (because of inversed polarity). Try connecting the pedal first, then plug the keyboard to your PC. Alexandre -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmouneyres at gmail.com Fri Mar 14 08:37:35 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Fri, 14 Mar 2014 09:37:35 +0100 Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: To me, connecting the midi first or after should have no influence, midi is kind of hot plug. On another hand, some keyboards need to have the pedal connected *before* powering on the keyboard, so the polarity of the pedal is correctly recognised. You can check with a midi monitor if the pedals affects sending the "Note off" messages, instead of just listening to the sound produced by the sampler Rapha?l 2014-03-14 8:52 UTC+01:00, Alexandre Prokoudine : > 14 ????? 2014 ?. 5:53 ???????????? "Clifford Dunn" > ???????: > >> What I do hope someone can help me with is a simple problem. When I >> plug a foot pedal into the sustain port of the keyboard, it doesn't >> add any change to the signal. The only other experience I have is >> using this keyboard with Logic, Garageband, and Main Stage. The >> sustain worked like a sustain would in these programs. I wonder if >> someone could point me towards unlocking this functionality. Google >> searches aren't too helpful with this. > > Typically the problem appears when you plug the pedal into an already > connected MIDI keyboard (because of inversed polarity). > > Try connecting the pedal first, then plug the keyboard to your PC. > > Alexandre > From alexandre.prokoudine at gmail.com Fri Mar 14 08:46:04 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Fri, 14 Mar 2014 12:46:04 +0400 Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: On Fri, Mar 14, 2014 at 12:37 PM, Rapha?l Mouneyres wrote: > To me, connecting the midi first or after should have no influence, > midi is kind of hot plug. > > On another hand, some keyboards need to have the pedal connected > *before* powering on the keyboard, so the polarity of the pedal is > correctly recognised. Doesn't it depends on the sustain pedal instead? :) Alexandre From tim at quitte.de Fri Mar 14 08:53:42 2014 From: tim at quitte.de (Tim Goetze) Date: Fri, 14 Mar 2014 09:53:42 +0100 (CET) Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: [Rapha?l Mouneyres] >You can check with a midi monitor if the pedals affects sending the >"Note off" messages, instead of just listening to the sound produced >by the sampler Sustain pedals usually trigger a Control Change message, numbered 64. Cheers, Tim From rmouneyres at gmail.com Fri Mar 14 09:17:50 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Fri, 14 Mar 2014 10:17:50 +0100 Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: >>Sustain pedals usually trigger a Control Change message, numbered 64. well yes, only if you plug them into a port labelled "sustain pedal" ;) A Sustain pedal in itself is only a switch. I'm sure you know that :p 2014-03-14 9:53 UTC+01:00, Tim Goetze : > [Rapha?l Mouneyres] >>You can check with a midi monitor if the pedals affects sending the >>"Note off" messages, instead of just listening to the sound produced >>by the sampler > > Sustain pedals usually trigger a Control Change message, numbered 64. > > Cheers, Tim From tim at quitte.de Fri Mar 14 11:34:42 2014 From: tim at quitte.de (Tim Goetze) Date: Fri, 14 Mar 2014 12:34:42 +0100 (CET) Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: [Rapha?l Mouneyres] >>>Sustain pedals usually trigger a Control Change message, numbered 64. >well yes, only if you plug them into a port labelled "sustain pedal" >;) A Sustain pedal in itself is only a switch. I'm sure you know that >:p I find it hard to imagine a MIDI master keyboard that would keep track of pressed keys instead of simply passing switch actions through as CC msgs, but certainly I haven't seen everything under the sun yet. :) Cheers, Tim >2014-03-14 9:53 UTC+01:00, Tim Goetze : >> [Rapha?l Mouneyres] >>>You can check with a midi monitor if the pedals affects sending the >>>"Note off" messages, instead of just listening to the sound produced >>>by the sampler >> >> Sustain pedals usually trigger a Control Change message, numbered 64. >> >> Cheers, Tim > From rustompmody at gmail.com Fri Mar 14 12:09:08 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 17:39:08 +0530 Subject: [LAU] debugging hiss Message-ID: Of late I am getting some amount of hiss from my speakers. I believe its a software problem because it is absent at startup and audibly starts sometime during the boot process. My system is fairly vanilla h/w -- ordinary Intel desktop with builtin soundcard, ordinary speakers. The only non-vanilla things maybe some things I installed when playing around with a TASCAM Any suggestions where/how to look? From jeremy at autostatic.com Fri Mar 14 12:40:35 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 14 Mar 2014 13:40:35 +0100 Subject: [LAU] M-Audio 88es Keyboard In-Reply-To: References: Message-ID: <5322F8C3.8020802@autostatic.com> On 03/14/2014 02:52 AM, Clifford Dunn wrote: > When I > plug a foot pedal into the sustain port of the keyboard, it doesn't > add any change to the signal. Hello Clifford, What kind of pedal are you using? I've got a Boss FS-5U myself which has a polarity switch. Maybe your pedal has such a switch too and it got switched accidentally? Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From fons at linuxaudio.org Fri Mar 14 13:11:41 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 14 Mar 2014 13:11:41 +0000 Subject: [LAU] debugging hiss In-Reply-To: References: Message-ID: <20140314131141.GA28877@linuxaudio.org> On Fri, Mar 14, 2014 at 05:39:08PM +0530, Rustom Mody wrote: > Of late I am getting some amount of hiss from my speakers. > I believe its a software problem because it is absent at startup and > audibly starts sometime during the boot process. > > My system is fairly vanilla h/w -- ordinary Intel desktop with builtin > soundcard, ordinary speakers. > > The only non-vanilla things maybe some things I installed when playing > around with a TASCAM How on earth can anybody help you if you give so little information. * Do you have active speakers (built-in amplifier) or is there an amplifier in between the sound card and the speakser ? If so, which input are you using ? How are things connected ? * Using ALSA/Jack/Pulse ?? * TASCAM is a brand name. They are hundreds if not thousands of different things with that name on it. So what is 'a TASCAM', and what did you install for playing around ? Apart from that, hiss is usually the result of either - something broken, - or using the wrong inputs, - or a volume control set too high. Another cause could be the mic inputs of your sound card routed to the outputs. Check the mixer. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rustompmody at gmail.com Fri Mar 14 13:46:47 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 19:16:47 +0530 Subject: [LAU] debugging hiss In-Reply-To: <20140314131141.GA28877@linuxaudio.org> References: <20140314131141.GA28877@linuxaudio.org> Message-ID: On Fri, Mar 14, 2014 at 6:41 PM, Fons Adriaensen wrote: > On Fri, Mar 14, 2014 at 05:39:08PM +0530, Rustom Mody wrote: > >> Of late I am getting some amount of hiss from my speakers. >> I believe its a software problem because it is absent at startup and >> audibly starts sometime during the boot process. >> >> My system is fairly vanilla h/w -- ordinary Intel desktop with builtin >> soundcard, ordinary speakers. >> >> The only non-vanilla things maybe some things I installed when playing >> around with a TASCAM > > How on earth can anybody help you if you give so little information. Sorry not sure what else I should have given... lspci ? aplay -l ?? > > * Do you have active speakers (built-in amplifier) or is there an > amplifier in between the sound card and the speakser ? If so, which > input are you using ? How are things connected ? > > * Using ALSA/Jack/Pulse ?? No jack (to start with at least when the hiss sound starts) I am using debian -- which meant that there was no pulse until recently Of late I see occasionally some pulse related messages. eg I get these messages when I start audacity: ---------------------------------- ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Connection refused Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 --------------------------------- > > * TASCAM is a brand name. They are hundreds if not thousands of > different things with that name on it. So what is 'a TASCAM', > and what did you install for playing around ? Yeah I know :-) I guess the TASCAM reference is a red-herring. I tried it more than 2 years ago. There was no (hiss) issue then. Dont have the hardware right now so dont know the number/spec. The hiss has started after some recent debian upgrade -- couple of weeks > > Apart from that, hiss is usually the result of either > > - something broken, > - or using the wrong inputs, > - or a volume control set too high. > > Another cause could be the mic inputs of your sound card > routed to the outputs. Check the mixer. Are you asking for hardware or software checking? Hardware: There is only one connection -- the speaker into the soundcard. No mic, no line. Software: I will need to know how to check. For now, I checked with xfce4-mixer. If I turn down master or front-mic the hiss reduces However then musescore (my main use right now) also becomes too soft Interestingly there was a musescore upgrade just a day or two ago And now I am seeing segfaults with this message: $ Alsa_driver: pcm_drop(play): Input/output error pcmStop failed Segmentation fault Dont know that this has anything to do with the original hiss question. > > Ciao, > > -- > FA Thanks for the tips Rusi From fons at linuxaudio.org Fri Mar 14 14:04:12 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 14 Mar 2014 14:04:12 +0000 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> Message-ID: <20140314140412.GB28877@linuxaudio.org> On Fri, Mar 14, 2014 at 07:16:47PM +0530, Rustom Mody wrote: > No jack (to start with at least when the hiss sound starts) > I am using debian -- which meant that there was no pulse until recently > Of late I see occasionally some pulse related messages. > eg I get these messages when I start audacity: > > ---------------------------------- > ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear > ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe > ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side > ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: > Connection refused > > ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: > Connection refused > > Expression 'stream->playback.pcm' failed in > 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 > Expression 'stream->playback.pcm' failed in > 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 > Expression 'stream->playback.pcm' failed in > 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 > Expression 'stream->playback.pcm' failed in > 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 > --------------------------------- First of all disable Pulse, or get rid of it. Don't know how to do that on Debian. > Hardware: There is only one connection -- the speaker into the soundcard. > No mic, no line. > For now, I checked with xfce4-mixer. > If I turn down master or front-mic the hiss reduces So you are sending the mic signal (which will be lots of hiss when no mic is connected) to the speakers. > However then musescore (my main use right now) also becomes too soft That should not depend on any 'Mic' settings, only on 'PCM' and 'Master'. If it does, that means the upgrade botched your ALSA setup. Try turning down (or switching off) anything 'Mic'. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From lorenzofsutton at gmail.com Fri Mar 14 14:07:44 2014 From: lorenzofsutton at gmail.com (Lorenzo Sutton) Date: Fri, 14 Mar 2014 15:07:44 +0100 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> Message-ID: <53230D30.9040903@gmail.com> On 14/03/2014 14:46, Rustom Mody wrote: [...] > For now, I checked with xfce4-mixer. > If I turn down master or front-mic the hiss reduces Is the machine a laptop? In that case the 'front-mic' could be the inbuilt one? Anyhow, mute all mic-related controls it in the _Playback_ tab of the mixer. You only want the microphone(s) enabled in the Capture section, unless you explicitly want to 'monitor' them. Also check that any 'mic boost' or similar is turned off/down, still in the Playback section. Ciao, Lorenzo. From rustompmody at gmail.com Fri Mar 14 14:21:03 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 19:51:03 +0530 Subject: [LAU] debugging hiss In-Reply-To: <53230D30.9040903@gmail.com> References: <20140314131141.GA28877@linuxaudio.org> <53230D30.9040903@gmail.com> Message-ID: On Fri, Mar 14, 2014 at 7:37 PM, Lorenzo Sutton wrote: > On 14/03/2014 14:46, Rustom Mody wrote: > [...] > >> For now, I checked with xfce4-mixer. >> If I turn down master or front-mic the hiss reduces > > > Is the machine a laptop? In that case the 'front-mic' could be the inbuilt > one? No its a desktop > Anyhow, mute all mic-related controls it in the _Playback_ tab of the mixer. Strange thing is in xfce4-mixer there is only a playback tab no record/capture tab -- weird! > You only want the microphone(s) enabled in the Capture section, unless you > explicitly want to 'monitor' them. > Also check that any 'mic boost' or similar is turned off/down, still in the > Playback section. If I turn these off my playback stops From rustompmody at gmail.com Fri Mar 14 14:23:26 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 19:53:26 +0530 Subject: [LAU] debugging hiss In-Reply-To: <20140314140412.GB28877@linuxaudio.org> References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> Message-ID: On Fri, Mar 14, 2014 at 7:34 PM, Fons Adriaensen wrote: > On Fri, Mar 14, 2014 at 07:16:47PM +0530, Rustom Mody wrote: > >> No jack (to start with at least when the hiss sound starts) >> I am using debian -- which meant that there was no pulse until recently >> Of late I see occasionally some pulse related messages. >> eg I get these messages when I start audacity: >> >> ---------------------------------- >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side >> ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: >> Connection refused >> >> ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: >> Connection refused >> >> Expression 'stream->playback.pcm' failed in >> 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 >> Expression 'stream->playback.pcm' failed in >> 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 >> Expression 'stream->playback.pcm' failed in >> 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 >> Expression 'stream->playback.pcm' failed in >> 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 >> --------------------------------- > > First of all disable Pulse, or get rid of it. Don't know how to > do that on Debian. > >> Hardware: There is only one connection -- the speaker into the soundcard. >> No mic, no line. >> For now, I checked with xfce4-mixer. >> If I turn down master or front-mic the hiss reduces > > So you are sending the mic signal (which will be lots of hiss when no > mic is connected) to the speakers. Yes But how am 'I' sending that signal? I dont have any .asoundrc Is there some other place to look for configs? > >> However then musescore (my main use right now) also becomes too soft > > That should not depend on any 'Mic' settings, only on 'PCM' and > 'Master'. If it does, that means the upgrade botched your ALSA setup. > Try turning down (or switching off) anything 'Mic'. When I turn off Mic my playback shuts off From rustompmody at gmail.com Fri Mar 14 15:22:17 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 20:52:17 +0530 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> <53230D30.9040903@gmail.com> Message-ID: On Fri, Mar 14, 2014 at 7:51 PM, Rustom Mody wrote: > On Fri, Mar 14, 2014 at 7:37 PM, Lorenzo Sutton > wrote: >> On 14/03/2014 14:46, Rustom Mody wrote: >> [...] >> >>> For now, I checked with xfce4-mixer. >>> If I turn down master or front-mic the hiss reduces >> >> >> Is the machine a laptop? In that case the 'front-mic' could be the inbuilt >> one? > > No its a desktop > >> Anyhow, mute all mic-related controls it in the _Playback_ tab of the mixer. > > Strange thing is in xfce4-mixer there is only a playback tab no > record/capture tab -- weird! To remove any variables due to malfunctioning xfce4-mixer, I tried alsamixer Here is the list of sliders that are under playback and under capture Note the misspellings and repeats are because I dont know how to see the name at the bottom in full so I am just showing what I see Capture has Front Mic Boos Line Boost Capture Capture 1 Digital Input Source (but no slider) Input Source 1 (no slider) Rear Mic boost Playback has Master Headphon PCM Front Front Mi Front Mi Surround Center LFE Line Line Boo Beep Channel (no slider) Rear Mic Rear Mic From rustompmody at gmail.com Fri Mar 14 15:42:42 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 21:12:42 +0530 Subject: [LAU] [LAA] aubio 0.4.1 In-Reply-To: <20140312191205.GA17497@coconut.piem.org> References: <20140312191205.GA17497@coconut.piem.org> Message-ID: On Thu, Mar 13, 2014 at 12:42 AM, Paul Brossier wrote: > Hi all, > > A new version of aubio, 0.4.1, is out. > > aubio is a library of functions to perform audio feature extractions > such as: > > - note onset detection > - pitch detection > - beat tracking > - MFCC computation > - spectral descriptors > > This version is mostly focusing on media file input and output. Here is > a quick overview of the changes. > > The most interesting feature in this release concerns aubiocut. Thanks > to the sponsoring of Mark Suppes, the python script to slice sound > steams was extended to be sample accurate, cut overlapping segments, and > work on multiple channels. > > New source and sink objects have been added to let aubio read and write > WAV files, even when built with no external libraries. This should > simplify the use of aubio on platforms such as Android or Windows. > > Existing sources and sinks have been extended to read and write from and > to multiple channels. This makes python-aubio one of the fastest and > most versatile Python module to read and write media files. > > This release also comes with a stack of bug fixes and code clean-ups. > > Note: this version is API and ABI compatible with 0.4.0. Since it only > adds new features to the existing interface, your existing source and > binary code will keep working without any modifications. > > To find out more about aubio and this release: > > Project homepage: > http://aubio.org/ I find this interesting In particular aubionotes claims to get midi (almost??) out of wav But I cant get it to work. That is aubio -i foo.wav certainly seems to be consuming CPU in that the fan goes fast for a few seconds but there is no output From rustompmody at gmail.com Fri Mar 14 17:47:32 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 14 Mar 2014 23:17:32 +0530 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> <53230D30.9040903@gmail.com> Message-ID: On Fri, Mar 14, 2014 at 8:52 PM, Rustom Mody wrote: > To remove any variables due to malfunctioning xfce4-mixer, I tried alsamixer > > Here is the list of sliders that are under playback and under capture > Note the misspellings and repeats are because I dont know how to see > the name at the bottom in full so I am just showing what I see > > Capture has: > Front Mic Boos > Line Boost > Capture > Capture 1 > Digital > Input Source (but no slider) > Input Source 1 (no slider) > Rear Mic boost > > > Playback has > Master > Headphon > PCM > Front > Front Mic > Front Mi > Surround > Center > LFE > Line > Line Boo > Beep > Channel (no slider) > Rear Mic > Rear Mic Figured that the name shows at the top in full so putting the full name Front Mic Boost Line Boost Capture Capture 1 Digital Input Source (but no slider) Input Source 1 (no slider) Rear Mic boost Playback has Master Headphone PCM Front Front Mic Front Mic Boost Surround Center LFE Line Line Boost Beep Channel Mode (no slider)[1] Rear Mic Rear Mic Boost [1] There is no slider However pressing up/down-arrows changes from 2ch 4ch 6ch. 2 and 4 have hiss 6 has much less So for now my problem is reduced though Ive no idea what this is about!! From federicogalland at gmail.com Fri Mar 14 21:22:31 2014 From: federicogalland at gmail.com (F Tux) Date: Fri, 14 Mar 2014 18:22:31 -0300 Subject: [LAU] [ANN] Vee One Suite 0.4.0 - A proto-beta party! In-Reply-To: <5319EC2B.207@rncbc.org> References: <5318E6AA.4020102@rncbc.org> <53199ACD.2080800@gareus.org> <5319EC2B.207@rncbc.org> Message-ID: Hi! Thanks for this new release. I performed a quick test, and on my debian 7 686 system, with the kxstudio repos, samplv1 loads samples, but doesn't show the waveform. This happens on the standalone and the plugin clients... Is there any bug tracking system for your apps? Thank you! Bye! On 3/7/14, Rui Nuno Capela wrote: > On 03/07/2014 10:09 AM, Robin Gareus wrote: >> On 03/06/2014 10:20 PM, Rui Nuno Capela wrote: >>> Howdy, >>> >>> as to no surprise (rly?) the so called Vee One Suite of old-schoolyards >>> gets bullied down to another bump in the head: now turning into a >>> proto-beta party? >> >> Thanks for all your work and organizing a party on top of it all! >> >> What's new in 0.4.0? - neither the email nor your blog mentions that. >> Is there an easily accessible changelog? >> > > news are mainly if not mostly about reverb being added to the stereo fx > gang and max.stage time for envelope generators now variable in > length--cf. Env.Time knob--was hard-coded/fixed to 5sec (synthv1, > samplv1) and to 2sec (drumkv1) before, now the default setting. > > cheers > -- > rncbc aka Rui Nuno Capela > rncbc at rncbc.org > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From ralf.mardorf at rocketmail.com Sat Mar 15 01:43:13 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 15 Mar 2014 02:43:13 +0100 Subject: [LAU] debugging hiss In-Reply-To: <20140314140412.GB28877@linuxaudio.org> References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> Message-ID: <1394847793.850.28.camel@archlinux> On Fri, 2014-03-14 at 14:04 +0000, Fons Adriaensen wrote: > First of all disable Pulse, or get rid of it. Don't know how to > do that on Debian. To get rid of it by removing and not disabling it and assumed there should be a hard dependency to pulseaudio, the easiest way is to build a dummy package using equivs. http://www.debian.org/doc/manuals/apt-howto/ch-helpers.en.html From rustompmody at gmail.com Sat Mar 15 02:42:57 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Sat, 15 Mar 2014 08:12:57 +0530 Subject: [LAU] debugging hiss In-Reply-To: <1394847793.850.28.camel@archlinux> References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> <1394847793.850.28.camel@archlinux> Message-ID: On Sat, Mar 15, 2014 at 7:13 AM, Ralf Mardorf wrote: > On Fri, 2014-03-14 at 14:04 +0000, Fons Adriaensen wrote: >> First of all disable Pulse, or get rid of it. Don't know how to >> do that on Debian. > > To get rid of it by removing and not disabling it and assumed there > should be a hard dependency to pulseaudio, the easiest way is to build a > dummy package using equivs. > > http://www.debian.org/doc/manuals/apt-howto/ch-helpers.en.html Thanks Ralf Actually I am on debian because ubuntu was forcing pulseaudio which was too much of a headache. When debian started doing the same, I decided to opt out of a clearly losing battle :-( I just checked : Trying to remove pulseaudio-utils and pavucontrol does not disturb anything Trying to remove libpulse0 removes everything! In any case the hiss problem is for now 'cured' -- not sure how Some fiddling in the alsamixer which includes -- 6 channels -- turning off all 'mic-ly' stuff in playback -- turning on dont exactly remember My only question now -- apart from "What the hell does a mic in playback mean?!" -- should I file a bug with xfce4-mixer for having only a playback tab? From rustompmody at gmail.com Sat Mar 15 02:45:53 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Sat, 15 Mar 2014 08:15:53 +0530 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> <1394847793.850.28.camel@archlinux> Message-ID: On Sat, Mar 15, 2014 at 8:12 AM, Rustom Mody wrote: > My only question now -- apart from "What the hell does a mic in > playback mean?!" -- should I file a bug with xfce4-mixer for having > only a playback tab? Ok got it: If I select the actual(!!) mics viz the ones that appear in capture in alsamixer in xfce4-mixer's select-controls the capture tab appears From ralf.mardorf at rocketmail.com Sat Mar 15 03:25:37 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 15 Mar 2014 04:25:37 +0100 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> <1394847793.850.28.camel@archlinux> Message-ID: <1394853937.850.61.camel@archlinux> On Sat, 2014-03-15 at 08:12 +0530, Rustom Mody wrote: > Trying to remove libpulse0 removes everything! You need to keep libpulse for many apps, but only pulseaudio is something that could cause trouble. It's ok to replace a pulseaudio package with a pulseaudio dummy package, but you need to keep a libpulse package. From ralf.mardorf at rocketmail.com Sat Mar 15 03:30:28 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 15 Mar 2014 04:30:28 +0100 Subject: [LAU] debugging hiss In-Reply-To: <1394853937.850.61.camel@archlinux> References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> <1394847793.850.28.camel@archlinux> <1394853937.850.61.camel@archlinux> Message-ID: <1394854228.850.64.camel@archlinux> On Sat, 2014-03-15 at 04:25 +0100, Ralf Mardorf wrote: > On Sat, 2014-03-15 at 08:12 +0530, Rustom Mody wrote: > > Trying to remove libpulse0 removes everything! > > You need to keep libpulse for many apps, but only pulseaudio is > something that could cause trouble. It's ok to replace a pulseaudio > package with a pulseaudio dummy package, but you need to keep a > libpulse package. Perhaps you also could replace libpulse by a dummy without getting side effects, I never tested it, there's no need to do it. From simonzwise at gmail.com Sat Mar 15 05:45:29 2014 From: simonzwise at gmail.com (Simon Wise) Date: Sat, 15 Mar 2014 16:45:29 +1100 Subject: [LAU] debugging hiss In-Reply-To: References: <20140314131141.GA28877@linuxaudio.org> <20140314140412.GB28877@linuxaudio.org> <1394847793.850.28.camel@archlinux> Message-ID: <5323E8F9.7090800@gmail.com> On 15/03/14 13:42, Rustom Mody wrote: > On Sat, Mar 15, 2014 at 7:13 AM, Ralf Mardorf > wrote: >> On Fri, 2014-03-14 at 14:04 +0000, Fons Adriaensen wrote: >>> First of all disable Pulse, or get rid of it. Don't know how to >>> do that on Debian. >> >> To get rid of it by removing and not disabling it and assumed there >> should be a hard dependency to pulseaudio, the easiest way is to build a >> dummy package using equivs. >> >> http://www.debian.org/doc/manuals/apt-howto/ch-helpers.en.html > > Thanks Ralf > > Actually I am on debian because ubuntu was forcing pulseaudio which > was too much of a headache. > When debian started doing the same, I decided to opt out of a clearly > losing battle :-( debian forces very little, no need for any desktop at all, or X, let alone pulse and such. But there are many desktop audio apps that are intended for use with pulse, and they depend on pulse. > > I just checked : > Trying to remove pulseaudio-utils and pavucontrol does not disturb anything > Trying to remove libpulse0 removes everything! apps which have a pulse option do of course need the library they have been compiled with, so you need that even if you do not use it. To avoid that you would need to recompile without pulse support, and clearly that isn't appropriate for general distribution. It is possible for apps to be a bit more sophisticated than this but most do not bother, it is easy just to make the library a dependency ... libraries are separate packages for this reason. Simon From piem at piem.org Sat Mar 15 19:02:42 2014 From: piem at piem.org (Paul Brossier) Date: Sat, 15 Mar 2014 20:02:42 +0100 Subject: [LAU] [LAA] aubio 0.4.1 In-Reply-To: References: <20140312191205.GA17497@coconut.piem.org> Message-ID: <20140315190242.GA14763@coconut.piem.org> On Fri, Mar 14, 2014 at 09:12:42PM +0530, Rustom Mody wrote: > I find this interesting > In particular aubionotes claims to get midi (almost??) out of wav > But I cant get it to work. > That is aubio -i foo.wav wow, that's a bug! adding -v, you will get the expected output. thanks for the report, best, piem > certainly seems to be consuming CPU in that the fan goes fast for a > few seconds but there is no output > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From guido-scholz at gmx.net Sat Mar 15 22:21:09 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Sat, 15 Mar 2014 23:21:09 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released Message-ID: <20140315222109.GA20139@traun.gscholz.bayernline.de> Dear all, AlsaModularSynth is a MIDI controlled realtime modular synthesizer and effect processor with support for LADSPA and JACK. After several years of collecting fixes and enhancements the new release provides a long list of changes: ams-2.1.0 (2014-03-15) Fixed Bugs o Linker error "undefined reference to symbol 'dlsym@@GLIBC_2.2.5'" on Fedora 13. o Crash on using looped signal paths o Armel compile error (debian #570848) o Prevent crash if jack handle is NULL (lp #553366) o Fix preferences dialog to show current color setup o Fix broken Yes/No/Cancel response on program exit o Fix memory leak in preferences widget o Fix triggered reset of LFO saw signals, patch provided by Bill Yerazunis o Reorganized commandline options for input and output to be valid for Alsa and JACK New Features o SIGUSR1 handler added to enable LADI session handling on application level 1. o Support for libclalsadrv API version 2.0.0 o Support for libzita-alsa-pcmi as an alternative for libclalsadrv o Improved port selection handling, patch provided by Sebastien Alaiwan o New context menu to disconnect module output ports o Support for JACK session handling o Rewritten preferences dialog o New about dialog o New option for saving window geometry (session handling) o New option hiding recently used files menu (keep secrets) o Add new menu item to open demo patch files directly o Add new menu item to open demo instrument patch files directly o Add keyboard shortcuts for module configuration dialogs o New option for module position grid (snap to grid) o New V8 Sequencer module, provided by Bill Yerazunis o New Analog Memory module, provided by Bill Yerazunis o New Bitgrinder module, provided by Bill Yerazunis o New Hysteresis module, provided by Bill Yerazunis o New VC-Delay module, provided by Bill Yerazunis o Add Pulsetrain Noise type to Noise 2 module, patch provided by Bill Yerazunis o New FFT Vocoder module, provided by Bill Yerazunis o Make control center window position and MIDI settings persistent o Add support for Qt5 (configure option --enable-qt5) General Changes o Separate handling of color scheme directory from patch file directoy. o Improved handling of CXXFLAGS variables o Add check for Ladspa header file. o Obsolete commandline option -l (preset file) removed o New commandline option for program version (--version) Have fun! Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From guido-scholz at gmx.net Sat Mar 15 22:26:06 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Sat, 15 Mar 2014 23:26:06 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140315222109.GA20139@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> Message-ID: <20140315222606.GA20269@traun.gscholz.bayernline.de> Am Sat, 15. Mar 2014 um 23:21:09 +0100 schrieb Guido Scholz: > Dear all, ... and yes, the files are at the usual place: https://sourceforge.net/projects/alsamodular/ Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From fons at linuxaudio.org Sat Mar 15 22:55:58 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 15 Mar 2014 22:55:58 +0000 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140315222109.GA20139@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> Message-ID: <20140315225558.GA12467@linuxaudio.org> On Sat, Mar 15, 2014 at 11:21:09PM +0100, Guido Scholz wrote: > o Support for libclalsadrv API version 2.0.0 > o Support for libzita-alsa-pcmi as an alternative for libclalsadrv Please note that libclalsadrv is deprecated, and libzita-alsa-pcmi should really be the default. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rustompmody at gmail.com Sun Mar 16 02:37:02 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Sun, 16 Mar 2014 08:07:02 +0530 Subject: [LAU] [LAA] aubio 0.4.1 In-Reply-To: <20140315190242.GA14763@coconut.piem.org> References: <20140312191205.GA17497@coconut.piem.org> <20140315190242.GA14763@coconut.piem.org> Message-ID: On Sun, Mar 16, 2014 at 12:32 AM, Paul Brossier wrote: > On Fri, Mar 14, 2014 at 09:12:42PM +0530, Rustom Mody wrote: >> I find this interesting >> In particular aubionotes claims to get midi (almost??) out of wav >> But I cant get it to work. >> That is aubio -i foo.wav > > wow, that's a bug! > > adding -v, you will get the expected output. Any easy/suggested way of converting to (actual) midi? From tim at klingt.org Sun Mar 16 09:32:37 2014 From: tim at klingt.org (Tim Blechmann) Date: Sun, 16 Mar 2014 10:32:37 +0100 Subject: [LAU] jack/oversampling Message-ID: hi all, for a project, i'm working with 192 khz, but unfortunately, none of my audio interfaces (rme multiface and fireface ucx) support 192khz. the ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode only seems to go up to 96khz. so i wonder, is there any way to perform upsampling within jack? tia, tim From grh at mur.at Sun Mar 16 09:47:26 2014 From: grh at mur.at (grh) Date: Sun, 16 Mar 2014 10:47:26 +0100 Subject: [LAU] ebur128 batch processing In-Reply-To: <5319C96D.8090606@stackingdwarves.net> References: <5319C96D.8090606@stackingdwarves.net> Message-ID: <5325732E.2040303@mur.at> Hallo! >> Any chance to do it as batch process? >> Result quality is critical as material will be broadcast on TV. > > hard problem. quickest solution: find somebody with a hardware leveller > (j?nger etc.) and pipe it through. > > problem is that there is no straightforward way from an initial loudness > measurement to a level change that predictably hits the -23 dbLUFS mark. Well, it's not really hard if you have the whole file already. Shameless self promotion: https://auphonic.com > > can i ask which broadcaster mandates -3dB true peak? that seems quite > conservative. For lossy audio codecs it's required to use -3dB True Peak in the official EBU R128 document. For lossless audio -1dBTP should be used ... LG Georg -- auphonic - automatic audio post production http://auphonic.com audio development, machine learning, open source and more -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature URL: From cbannister at slingshot.co.nz Sun Mar 16 11:21:37 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Mon, 17 Mar 2014 00:21:37 +1300 Subject: [LAU] Need suggestions for lightweight audio programs In-Reply-To: References: Message-ID: <20140316112137.GB23182@tal> On Sat, Mar 08, 2014 at 12:07:08PM -0500, LM wrote: > alsamixergui, FLTK front end to mixer: > https://packages.debian.org/wheezy/alsamixergui :) Oooops, I think you mean: http://www.iua.upf.es/~mdeboer/projects/alsamixergui/ Although, it does seem to be down. -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X From fons at linuxaudio.org Sun Mar 16 11:52:37 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 16 Mar 2014 11:52:37 +0000 Subject: [LAU] jack/oversampling In-Reply-To: References: Message-ID: <20140316115237.GA344@linuxaudio.org> On Sun, Mar 16, 2014 at 10:32:37AM +0100, Tim Blechmann wrote: > for a project, i'm working with 192 khz, but unfortunately, none of my > audio interfaces (rme multiface and fireface ucx) support 192khz. the > ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode > only seems to go up to 96khz. so i wonder, is there any way to perform > upsampling within jack? You could run Jack with the dummy backend and use zita-a2j and/or j2a to add your HW interface. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From paul at linuxaudiosystems.com Sun Mar 16 11:55:24 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 16 Mar 2014 07:55:24 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <20140316115237.GA344@linuxaudio.org> References: <20140316115237.GA344@linuxaudio.org> Message-ID: On Sun, Mar 16, 2014 at 7:52 AM, Fons Adriaensen wrote: > On Sun, Mar 16, 2014 at 10:32:37AM +0100, Tim Blechmann wrote: > > > for a project, i'm working with 192 khz, but unfortunately, none of my > > audio interfaces (rme multiface and fireface ucx) support 192khz. the > > ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode > > only seems to go up to 96khz. so i wonder, is there any way to perform > > upsampling within jack? > > You could run Jack with the dummy backend and use zita-a2j and/or j2a > to add your HW interface. > and of course if you use jack1 (0.124.0 or later), these tools are built in to jackd, so that adding the device can be as simple as -A DEVNAME (or it can be a bit more complex if you need to specify different parameters from the server's own device, as you do). -------------- next part -------------- An HTML attachment was scrubbed... URL: From tim at klingt.org Sun Mar 16 12:01:35 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 13:01:35 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <20140316115237.GA344@linuxaudio.org> References: <20140316115237.GA344@linuxaudio.org> Message-ID: >> for a project, i'm working with 192 khz, but unfortunately, none of my >> audio interfaces (rme multiface and fireface ucx) support 192khz. the >> ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode >> only seems to go up to 96khz. so i wonder, is there any way to perform >> upsampling within jack? > > You could run Jack with the dummy backend and use zita-a2j and/or j2a > to add your HW interface. that sounds like a good workaround! thnx, tim From tim at klingt.org Sun Mar 16 12:18:12 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 13:18:12 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <20140316115237.GA344@linuxaudio.org> Message-ID: <53259684.6070003@klingt.org> >>> for a project, i'm working with 192 khz, but unfortunately, none of my >>> audio interfaces (rme multiface and fireface ucx) support 192khz. the >>> ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode >>> only seems to go up to 96khz. so i wonder, is there any way to perform >>> upsampling within jack? >> >> You could run Jack with the dummy backend and use zita-a2j and/or j2a >> to add your HW interface. >> > > and of course if you use jack1 (0.124.0 or later), these tools are built in > to jackd, so that adding the device can be as simple as -A DEVNAME (or it > can be a bit more complex if you need to specify different parameters from > the server's own device, as you do). unfortunately jack1 is not an option for me, as it's approach to pre-fault the stack does not have any checks for the stack boundaries [1] ... this consistently crashed my application tim [1] http://trac.jackaudio.org/ticket/276 From federicogalland at gmail.com Sun Mar 16 14:30:25 2014 From: federicogalland at gmail.com (F Tux) Date: Sun, 16 Mar 2014 11:30:25 -0300 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140315225558.GA12467@linuxaudio.org> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> Message-ID: I'm trying to compile on an up to date gentoo system (qtcore-4.8.5) and it fails to find Qtcore... Any suggestions? Thanks! On 3/15/14, Fons Adriaensen wrote: > On Sat, Mar 15, 2014 at 11:21:09PM +0100, Guido Scholz wrote: > >> o Support for libclalsadrv API version 2.0.0 >> o Support for libzita-alsa-pcmi as an alternative for libclalsadrv > > Please note that libclalsadrv is deprecated, and libzita-alsa-pcmi > should really be the default. > > Ciao, > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From len at ovenwerks.net Sun Mar 16 15:58:27 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 16 Mar 2014 08:58:27 -0700 (PDT) Subject: [LAU] jack/oversampling In-Reply-To: References: Message-ID: On Sun, 16 Mar 2014, Tim Blechmann wrote: > for a project, i'm working with 192 khz, but unfortunately, none of my > audio interfaces (rme multiface and fireface ucx) support 192khz. the > ucx supports 192khz in hardware, but the usb/alsa/class-compliant mode > only seems to go up to 96khz. so i wonder, is there any way to perform > upsampling within jack? Personally, I would mix the project at 48k or 96k and upsample the finished product. I don't think you gain anything by upsampling before processing. Your whole system has to do twice the work for no gain and so you have limits to your effects options before you would at the lower rate. Contracts may make this harder, or live streaming (in this case the conversion can happen on the way out though) or working with some material that is already at the higher rate. If you must have the high sample rate going into jack, then zita-a2j is the way to go. -- Len Ovens www.ovenwerks.net From ralf.mardorf at rocketmail.com Sun Mar 16 16:05:05 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 17:05:05 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: Message-ID: <1394985905.9232.57.camel@archlinux> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: > I would mix the project at 48k or 96k Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best sound quality. From tim at klingt.org Sun Mar 16 16:45:22 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 17:45:22 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1394985905.9232.57.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> Message-ID: >> I would mix the project at 48k or 96k > > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best > sound quality. if that would only be true ... a) any non-linearity introduces harmonics, some non-linearities introduce an infinite amount of harmonics, which will cause foldover distortion. the large the sampling-rate, the lower the foldover. b) delay-lines have a higher precision at higher sampling-rates c) the tuning of digital filters is more precise at higher sampling-rates due to the frequency warping in the blt iir filters may have a higher quantization noise, but that is the reason, why a good filter implementation is done in double-precision. frankly, 48k may be a good enough for distribution, but it is sub-optimal not for production ... and it is horrible for digital synthesis. fwiw, for digital synthesis (non-standard or distortion synthesis) i ended up rendering my compositions at 3mhz ... which was a good compromise between computation time and sound quality. best, tim note on a: if your signal processor introduces the Nth harmonic, you have to upsample your signal by a factor of N. or apply a pre-filter on your signal by nyquist/N. question for the reader: in order to completely prevent foldover distortion, how much do you have to upsample for a tanh waveshaper (a processor that introduces infinite harmonics)? From aiyumi.br at gmail.com Sun Mar 16 16:50:18 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Sun, 16 Mar 2014 13:50:18 -0300 Subject: [LAU] Yamaha Motif XF as USB Soundcard Message-ID: With my previous kernels (2.6.29 and 3.2.29), I tried plugging the Motif XF to my PC via USB, but it wasn't recognized. After my latest system upgrade to Slackware 14.1 and kernel 3.10.17, I decided to try again, to see if something had changed. To my surprise, the MIDI ports were recognized. $ aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 28:0 Virtual Raw MIDI 3-0 VirMIDI 3-0 29:0 Virtual Raw MIDI 3-1 VirMIDI 3-1 30:0 Virtual Raw MIDI 3-2 VirMIDI 3-2 31:0 Virtual Raw MIDI 3-3 VirMIDI 3-3 32:0 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 1 32:1 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 2 32:2 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 3 32:3 YAMAHA MOTIF XF8 YAMAHA MOTIF XF8 MIDI 4 MIDI recording and playback work fine, but now I want to know if it can be made to work as a soundcard. I know that the Motif XF can work as a USB audio interface on Windows and Mac. Here, it even happens to be listed as one in "/proc/asound/cards" $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7910000 irq 42 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf7080000 irq 17 2 [Loopback ]: Loopback - Loopback Loopback 1 3 [VirMIDI ]: VirMIDI - VirMIDI Virtual MIDI Card 1 4 [XF8 ]: USB-Audio - YAMAHA MOTIF XF8 Yamaha YAMAHA MOTIF XF8 at usb-0000:00:1a.0-1.2, full speed but, problem is, it doesn't show on "aplay -l" nor "arecord -l" (nor any other audio app for that matter). $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC662 rev3 Analog [ALC662 rev3 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 2: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Trying to force using it as soundcard (based on info from "/proc/asound/cards") doesn't work either. For example: $ aplay -D plughw:XF8 file.wav aplay: main:722: audio open error: No such file or directory Same result with "plughw:4" instead of "plughw:XF8". I read that problems like this could happen if the USB soundcard was plugged into a USB 3.0 port. Indeed, I have USB 3 ports here, but I tried plugging it into USB 2 ports and the result was the same. More command outputs: $ lsusb Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 003 Device 010: ID 046d:c315 Logitech, Inc. Classic Keyboard 200 Bus 003 Device 029: ID 046d:c077 Logitech, Inc. Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 010: ID 0499:105c Yamaha Corp. # lsusb -v [...] Bus 001 Device 010: ID 0499:105c Yamaha Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0499 Yamaha Corp. idProduct 0x105c bcdDevice 1.00 iManufacturer 1 YAMAHA Corporation iProduct 2 YAMAHA MOTIF XF8 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 99 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 51 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 06 24 02 02 03 00 ** UNRECOGNIZED: 06 24 02 02 04 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 02 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 03 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 04 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Status: 0x0001 Self Powered Any ideas? Anything else I can try? Googling about using the Motif XF on Linux, I found this: http://www.motifator.com/index.php/forum/viewthread/451381/ I also found this thread, but it's about MOX and not Motif XF: http://www.motifator.com/index.php/forum/viewthread/458166/ which lead me to this patch: http://www.mail-archive.com/alsa-user at lists.sourceforge.net/msg28116.html The discussions are a bit old, but it seems that the patch was included upstream and MOX is working. Maybe the problem with the XF is similar? -- ____________________ Blog: http://aiyumi.warpstar.net/ From ralf.mardorf at rocketmail.com Sun Mar 16 17:03:56 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 18:03:56 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> Message-ID: <1394989436.9232.69.camel@archlinux> On Sun, 2014-03-16 at 17:45 +0100, tim wrote: > fwiw, for digital synthesis (non-standard or distortion synthesis) i > ended up rendering my compositions at 3mhz ... which was a good > compromise between computation time and sound quality. And I prefer analog gear over digital gear, but 48 KHz 32 bit float is perfectly ok for production, assumed there aren't software or hardware issues. From tim at klingt.org Sun Mar 16 17:21:18 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 18:21:18 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1394989436.9232.69.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <1394989436.9232.69.camel@archlinux> Message-ID: >> fwiw, for digital synthesis (non-standard or distortion synthesis) i >> ended up rendering my compositions at 3mhz ... which was a good >> compromise between computation time and sound quality. > > And I prefer analog gear over digital gear maybe because you did not like the aliasing artifacts and/or quantization noise? scnr > but 48 KHz 32 bit float is > perfectly ok for production, assumed there aren't software or hardware > issues. > From guido-scholz at gmx.net Sun Mar 16 17:25:32 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Sun, 16 Mar 2014 18:25:32 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> Message-ID: <20140316172531.GA4250@traun.gscholz.bayernline.de> Am Sun, 16. Mar 2014 um 11:30:25 -0300 schrieb F Tux: > I'm trying to compile on an up to date gentoo system (qtcore-4.8.5) > and it fails to find Qtcore... > > Any suggestions? What is the result of a locate libQtCore.so on your system? Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From willgodfrey at musically.me.uk Sun Mar 16 17:39:39 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sun, 16 Mar 2014 17:39:39 +0000 Subject: [LAU] Yoshimi V1.2.0 Message-ID: <20140316173939.37279207@debian> I'm pleased to say this is now available from: http://sourceforge.net/projects/yoshimi/files/1.2/yoshimi-1.2.0.tar.bz2/download Apart from a number of bug fixes we have: Circle and Spike AddSynth waveshapes MIDI bank and program change - with extra configuration in 'settings' Included patch set additions and updates. Next on the roadmap is midilearn and extra control exposure. Discussion would be welcome on: yoshimi-user at lists.sourceforge.net -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From ralf.mardorf at rocketmail.com Sun Mar 16 17:40:18 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 18:40:18 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1394991527.9232.79.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <1394989436.9232.69.camel@archlinux> <1394991527.9232.79.camel@archlinux> Message-ID: <1394991618.9232.81.camel@archlinux> On Sun, 2014-03-16 at 18:21 +0100, tim wrote: > >> fwiw, for digital synthesis (non-standard or distortion synthesis) i > >> ended up rendering my compositions at 3mhz ... which was a good > >> compromise between computation time and sound quality. > > > > And I prefer analog gear over digital gear > > maybe because you did not like the aliasing artifacts and/or > quantization noise? I'm a dino and used to analog gear. When I need to use digital gear I stay with 48 KHz, if I have a choice, I use analog gear. I don't know why I prefer analog gear. From tim at klingt.org Sun Mar 16 17:53:34 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 18:53:34 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1394991618.9232.81.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <1394989436.9232.69.camel@archlinux> <1394991527.9232.79.camel@archlinux> <1394991618.9232.81.camel@archlinux> Message-ID: >>>> fwiw, for digital synthesis (non-standard or distortion synthesis) i >>>> ended up rendering my compositions at 3mhz ... which was a good >>>> compromise between computation time and sound quality. >>> >>> And I prefer analog gear over digital gear >> >> maybe because you did not like the aliasing artifacts and/or >> quantization noise? > > I'm a dino and used to analog gear. When I need to use digital gear I > stay with 48 KHz, if I have a choice, I use analog gear. I don't know > why I prefer analog gear. so, why 48 and not 44.1? From ralf.mardorf at rocketmail.com Sun Mar 16 18:02:34 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 19:02:34 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> <1394989436.9232.69.camel@archlinux> <1394991527.9232.79.camel@archlinux> <1394991618.9232.81.camel@archlinux> Message-ID: <1394992954.9232.87.camel@archlinux> When using 44.1 KHz blood is dripping down my ears. IOW the borderline for me is >= 48 KHz and when ever possible I use analog gear. From tim at klingt.org Sun Mar 16 18:10:07 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 19:10:07 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1394992954.9232.87.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <1394989436.9232.69.camel@archlinux> <1394991527.9232.79.camel@archlinux> <1394991618.9232.81.camel@archlinux> <1394992954.9232.87.camel@archlinux> Message-ID: > When using 44.1 KHz blood is dripping down my ears. IOW the borderline > for me is >= 48 KHz and when ever possible I use analog gear. then don't saturate too much. and please, don't tell anyone that they should reduce their production quality to your standards. it is a shame for From ralf.mardorf at rocketmail.com Sun Mar 16 18:31:11 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 19:31:11 +0100 Subject: [LAU] jack/oversampling Message-ID: <1394994671.9232.95.camel@archlinux> On Sun, 2014-03-16 at 19:10 +0100, tim wrote: > don't tell anyone that they should reduce their production > quality to your standards. 48 KHz isn't my standard, for good reasons it's a common studio standard (as long as live usage latency isn't an issue). If you want analog sound, than use analog gear. If you want to have perfect digital quality 48 KHz / 24 bit is all you need, for production 32 bit float _might_ be better. From fons at linuxaudio.org Sun Mar 16 18:32:53 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 16 Mar 2014 18:32:53 +0000 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> Message-ID: <20140316183253.GA32563@linuxaudio.org> On Sun, Mar 16, 2014 at 05:45:22PM +0100, tim wrote: > a) any non-linearity introduces harmonics, some non-linearities > introduce an infinite amount of harmonics, which will cause foldover > distortion. the large the sampling-rate, the lower the foldover. You should not have any non-linearities, except those introduced on purpose, i.e. distortion plugins and the like. And then it all depends on how these are designed. If done well, they will not add any aliased components. One way to avoid that is using higher sample rates internally, but it's not the only one. > b) delay-lines have a higher precision at higher sampling-rates Fractional delays are possible at any rate, to any precision. The only limit is that you can't have very short ones (as the output would depend on future samples). > c) the tuning of digital filters is more precise at higher > sampling-rates due to the frequency warping in the blt Assuming the filter is _tuned_ correctly (e.g. the centre frequency for a parametric is corrected for warping), there will be a difference in the actual shape of the FR. But there is _no_ reason to assume that the original 'analog' shape is any better (or worse) than the warped one. > iir filters may have a higher quantization noise, but that is the > reason, why a good filter implementation is done in double-precision. No. If a filter requires double precision to avoid problems then you made a bad choice of filter architecture. Lots of plugins, (usually using 'textbook' biquads) fail in this way. It's perfectly possible to create audio filters that work perfectly even in 16-bit fixed point format (with a higher precision multiply). A lot of research went into this in the early years of digital audio - just look up the AES journals from the 1970s. The solution is to understand the problem and use the correct filter architecture, not the brute force method of using doubles blindly. > frankly, 48k may be a good enough for distribution, but it is > sub-optimal not for production ... and it is horrible for digital > synthesis. Only if you use 'primitive' algorithms. Unfortunately there's a lot of those around. In summary, 96 or 192 kHz will allow you to use simpler algorithms. That may be a good reason for higher sampler rates, but it doesn't mean you can't have the same performance at 48 kHz. Another good reason for higher sampling rates is that the antialising filters in the converters can have a much wider transition band (assuming you don't actually use the higher bandwidth), leading to much reduced latency. It's the reason why 'digital snakes' used in PA system usually work at 96 kHz. By starting the transition band at 24 kHz or so they can use very short filters, a fraction of a millisecond for some. The same matter makes all the difference between 44.1 and 48 kHz. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rosea.grammostola at gmail.com Sun Mar 16 18:37:28 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 16 Mar 2014 19:37:28 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140316172531.GA4250@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> <20140316172531.GA4250@traun.gscholz.bayernline.de> Message-ID: Huh no NSM support? It's here https://github.com/royvegard/ams On Sun, Mar 16, 2014 at 6:25 PM, Guido Scholz wrote: > Am Sun, 16. Mar 2014 um 11:30:25 -0300 schrieb F Tux: > > > I'm trying to compile on an up to date gentoo system (qtcore-4.8.5) > > and it fails to find Qtcore... > > > > Any suggestions? > > What is the result of a > > locate libQtCore.so > > on your system? > > Guido > > -- > http://wie-im-flug.net/ > http://www.lug-burghausen.org/ > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gheskett at wdtv.com Sun Mar 16 18:39:42 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Sun, 16 Mar 2014 14:39:42 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <1394985905.9232.57.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> Message-ID: <201403161439.42284.gheskett@wdtv.com> On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: > > I would mix the project at 48k or 96k > > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best > sound quality. > That I think is a personal call Ralf, primarily because at 48 Khz, your anti-aliasing filters had better be very very good brick walls by the time you get above 24Khz in input content just for the aliasing control. And aliasing noise, once introduced, cannot be removed by any known math function that does not have precise knowledge of the phasing (aka group delay) of the original signal. And those very good brick wall filters _will_ have a considerable group delay. IMO doing the sampling at 240K, then doing a weighted sum shift (5 stage shift) to decimate the data down to 48K, should result in dropping the alias caused noise floor by several db. Just bring lots of expensive hardware to do that. > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From ralf.mardorf at rocketmail.com Sun Mar 16 18:47:15 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 19:47:15 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <20140316183253.GA32563@linuxaudio.org> References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> Message-ID: <1394995635.9232.99.camel@archlinux> On Sun, 2014-03-16 at 18:32 +0000, Fons Adriaensen wrote: > The same matter makes all the difference between 44.1 and 48 kHz. Theoretically? Plug in your Mini Moog to your RMR card and record it using 44.1 and 48. If you can't hear a difference, than visit your otolaryngologist. From nettings at stackingdwarves.net Sun Mar 16 18:50:57 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 16 Mar 2014 19:50:57 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> Message-ID: <5325F291.70005@stackingdwarves.net> On 03/16/2014 05:45 PM, tim wrote: > a) any non-linearity introduces harmonics, some non-linearities > introduce an infinite amount of harmonics, which will cause foldover > distortion. the large the sampling-rate, the lower the foldover. ok, so you are trying to do weird synthesis that can produce non-bandlimited output? i can see how you might want to use high sampling rates there, but then again there will always be another processing step that causes yet higher harmonics - addressing that with high sample rates seems like a somewhat blunt approach that is bound to fail eventually. > b) delay-lines have a higher precision at higher sampling-rates that statement is definitely not correct. granted, if you only do delays with sample granularity (which has the big advantage of not requiring any computation), there is some benefit in using higher rates. but you can produce sub-sample delays with arbitrary precision easily. for IIR feedback, i sure see the point, but then the question becomes "why do you need to expose this to the outside world?" - just upsample in your processing application and leave the rest of the jack graph running at a sane rate. > c) the tuning of digital filters is more precise at higher > sampling-rates due to the frequency warping in the blt i don't understand this. can you elaborate? what is "blt"? > note on a: > if your signal processor introduces the Nth harmonic, you have to > upsample your signal by a factor of N. or apply a pre-filter on your > signal by nyquist/N. true. it's a funny and somewhat strange thought exercise for me to try and achieve the highest possible "fidelity" with brutal distortion algorithms - obviously, since i don't work with distortion, i try to keep my signal chains as linear as possible. but i can see how somebody well trained in distortion synthesis would want to eliminate aliasing artefacts, since those would conceivably interfere with systematic exploration of sounds based on prior experience, and make the sonic outcome even more erratic than it already is... but in any case, there is no point in taking the internal higher sampling rates out into the real world, so the zita resampling approach might be your best bet. > question for the reader: in order to completely prevent foldover > distortion, how much do you have to upsample for a tanh waveshaper (a > processor that introduces infinite harmonics)? incidentally, just returned from musikmesse, and i've had my share of DXD/DSD loonies... if you want to go there, there is people who want to sell you 256-times oversampled single-bit delta sigma gear, and they will happily talk megahertz with you. it would be a ton of fun to discuss with them the best way to handle a tanh waveshaper, and what new ultimate fidelity frontiers are required for the distortion synthesis crowd. just make sure you avoid the term distortion, call it "spectral enhancement processing" instead. >;-> best, j?rn -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From ralf.mardorf at rocketmail.com Sun Mar 16 18:57:50 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 19:57:50 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <201403161439.42284.gheskett@wdtv.com> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> Message-ID: <1394996270.9232.107.camel@archlinux> On Sun, 2014-03-16 at 14:39 -0400, Gene Heskett wrote: > On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > > > On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: > > > I would mix the project at 48k or 96k > > > > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best > > sound quality. > > > That I think is a personal call Ralf, primarily because at 48 Khz, your > anti-aliasing filters had better be very very good brick walls by the time > you get above 24Khz in input content just for the aliasing control. And > aliasing noise, once introduced, cannot be removed by any known math > function that does not have precise knowledge of the phasing (aka group > delay) of the original signal. > > And those very good brick wall filters _will_ have a considerable group > delay. IMO doing the sampling at 240K, then doing a weighted sum shift (5 > stage shift) to decimate the data down to 48K, should result in dropping > the alias caused noise floor by several db. Just bring lots of expensive > hardware to do that. Are we talking about reality or golden ears? I disagree with Fons regarding to the 44.1 KHz are as good as 48 KHz (in theory he might be right), but at least as long as there aren't software or hardware issues, _we_ are unable to hear a difference between 48 KHz and > 48 KHz. Analog audio quality is something different Gene. Is you claim that > 48 KHz you get that analog thingy, I'm missing for digital recordings? My RME card can do 192 KHz, I never tested it, but I will compare recordings ASAP, likely after August this year I have got the time to do it. From tim at klingt.org Sun Mar 16 19:16:34 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 20:16:34 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <20140316183253.GA32563@linuxaudio.org> References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> Message-ID: >> a) any non-linearity introduces harmonics, some non-linearities >> introduce an infinite amount of harmonics, which will cause foldover >> distortion. the large the sampling-rate, the lower the foldover. > > You should not have any non-linearities, except those introduced > on purpose, i.e. distortion plugins and the like. And then it > all depends on how these are designed. If done well, they will > not add any aliased components. One way to avoid that is using > higher sample rates internally, but it's not the only one. i'm curious, what are the other ways? >> frankly, 48k may be a good enough for distribution, but it is >> sub-optimal not for production ... and it is horrible for digital >> synthesis. > > Only if you use 'primitive' algorithms. Unfortunately there's > a lot of those around. well, we are living in a world of df2 biquad filters, which tend to blow up when modulating parameters, most delay lines are 1/2/4-point interpolations and non-linearities are applied without any oversampling ... > In summary, 96 or 192 kHz will allow you to use simpler algorithms. or get better sound quality from existing plugins ;) tim From piem at piem.org Sun Mar 16 19:51:07 2014 From: piem at piem.org (Paul Brossier) Date: Sun, 16 Mar 2014 16:51:07 -0300 Subject: [LAU] [LAA] aubio 0.4.1 In-Reply-To: References: <20140312191205.GA17497@coconut.piem.org> <20140315190242.GA14763@coconut.piem.org> Message-ID: <532600AB.40703@piem.org> On 03/15/2014 11:37 PM, Rustom Mody wrote: > On Sun, Mar 16, 2014 at 12:32 AM, Paul Brossier > wrote: >> On Fri, Mar 14, 2014 at 09:12:42PM +0530, Rustom Mody wrote: >>> I find this interesting In particular aubionotes claims to get >>> midi (almost??) out of wav But I cant get it to work. That is >>> aubio -i foo.wav >> >> wow, that's a bug! >> >> adding -v, you will get the expected output. > > Any easy/suggested way of converting to (actual) midi? i'm looking into ways to improve it. With python, the midiutil package should do the trick. in C, i don't know of a simple and portable library to write midi files. http://code.google.com/p/midiutil/ right now, the easiest way would be to use the vamp aubio plugin with sonic visualizer, which should let you save the detected notes as a midi file. http://www.sonicvisualiser.org/ Let me know how it goes! cheers, piem -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 242 bytes Desc: OpenPGP digital signature URL: From gheskett at wdtv.com Sun Mar 16 19:52:19 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Sun, 16 Mar 2014 15:52:19 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <1394996270.9232.107.camel@archlinux> References: <201403161439.42284.gheskett@wdtv.com> <1394996270.9232.107.camel@archlinux> Message-ID: <201403161552.19316.gheskett@wdtv.com> On Sunday 16 March 2014 15:11:20 Ralf Mardorf did opine: > On Sun, 2014-03-16 at 14:39 -0400, Gene Heskett wrote: > > On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > > > On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: > > > > I would mix the project at 48k or 96k > > > > > > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides > > > best sound quality. > > > > That I think is a personal call Ralf, primarily because at 48 Khz, > > your anti-aliasing filters had better be very very good brick walls > > by the time you get above 24Khz in input content just for the > > aliasing control. And aliasing noise, once introduced, cannot be > > removed by any known math function that does not have precise > > knowledge of the phasing (aka group delay) of the original signal. > > > > And those very good brick wall filters _will_ have a considerable > > group delay. IMO doing the sampling at 240K, then doing a weighted > > sum shift (5 stage shift) to decimate the data down to 48K, should > > result in dropping the alias caused noise floor by several db. Just > > bring lots of expensive hardware to do that. > > Are we talking about reality or golden ears? > > I disagree with Fons regarding to the 44.1 KHz are as good as 48 KHz (in > theory he might be right), but at least as long as there aren't software > or hardware issues, _we_ are unable to hear a difference between 48 KHz > and > 48 KHz. > Pretty much true, even for golden ears, PROVIDED the anti-aliasing filtration in front of that digitizer is down at least 70 db by 1/2 the sample frequency. You would be surprised at the gear available that depends on the microphone to do 80% of the filtering, and at the atrociously digitized sound you could get when miking a set of snares or cymbals with an Altec M21b or a PZM, both of which have pretty good response yet at 25Khz. Even some of the better ribbon mikes will cover those frequencies although they would tend to be very "peaky". > Analog audio quality is something different Gene. Is you claim that > 48 > KHz you get that analog thingy, I'm missing for digital recordings? I hesitate to answer this, my German is far worse than your English, and I don't quite get the translation. What I am saying is that when digitizing at 48Khz, any content in the input signal that has components approaching 1/2 the digitization sampler frequency will be aliased. Said another way, a 20Khz input will be digitized as both the 20Khz fundamental, and an 8Khz beat frequency "alias" component, and that this is one of the more distracting artifacts. The only way to stop it is to filter that input to where the stuff above 16Khz has been reduced to undetectable levels. > My RME card can do 192 KHz, I never tested it, but I will compare > recordings ASAP, likely after August this year I have got the time to do > it. To properly decimate to 48Khz that would take a 4 stage weighted summing adder, but given that, the results might be startlingly better (at least for the golden ears) than a straight 48Khz digitization. Listening to that 192Khz recording should be a noticeable improvement compared to a 48Khz recording, just from the reduction of aliasing artifacts alone, given good enough mikes and sound sources suitable for exploring/exploiting the alias that will happen in the real world. These aliasing artifacts you don't hear during the quiet passages of the music, but will be superimposed on the higher frequency portions of the src audio signal, contaminating it as it gets louder. It will convert that cymbal to fingernails on a blackboard, ditto for the brushed snare. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From fons at linuxaudio.org Sun Mar 16 20:26:37 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 16 Mar 2014 20:26:37 +0000 Subject: [LAU] jack/oversampling In-Reply-To: <1394996270.9232.107.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <1394996270.9232.107.camel@archlinux> Message-ID: <20140316202636.GA23003@linuxaudio.org> On Sun, Mar 16, 2014 at 07:57:50PM +0100, Ralf Mardorf wrote: > I disagree with Fons regarding to the 44.1 KHz are as good as 48 KHz (in > theory he might be right) You probably misread what I wrote - I never claimed such a thing. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Sun Mar 16 20:28:19 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 21:28:19 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <201403161552.19316.gheskett@wdtv.com> References: <201403161439.42284.gheskett@wdtv.com> <1394996270.9232.107.camel@archlinux> <201403161552.19316.gheskett@wdtv.com> Message-ID: <1395001699.9232.121.camel@archlinux> Gene :) ok, when I've got the time to do it, I will borrow the Brauner VM-1 tube mic + an ART tube pre amp from a friend, connect it directly to my RME HDSPe AIO and pick Blackbird from the Beatles on my classical guitar and do a recording at 48 KHz and 192 KHz. I don't know what will happen, but I suspect there will be no audible difference. Regards, Ralf From xiphmont at gmail.com Sun Mar 16 20:28:21 2014 From: xiphmont at gmail.com (Monty Montgomery) Date: Sun, 16 Mar 2014 16:28:21 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <20140316183253.GA32563@linuxaudio.org> References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> Message-ID: Every time I'm about to jump into one of these discussions, Fons says pretty much what I was going to say. But there is one additional textbook observation I was going to make: > a) any non-linearity introduces harmonics, some non-linearities > introduce an infinite amount of harmonics, which will cause foldover > distortion. the large the sampling-rate, the lower the foldover. Although this is mostly true, it's difficult to believe it matters in a practical audio application. If it does, in most of the cases that are not deliberately using a highly nonlinear transfer, adding one additional bit of depth will get you the same noise/distortion benefit as doubling the sampling rate. Once you're using 32 bit floats, you're already 160dB down... If you need more, I'd love to know what you're doing, and I'm not being sarcastic! Monty From ralf.mardorf at rocketmail.com Sun Mar 16 20:33:19 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 21:33:19 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <20140316202636.GA23003@linuxaudio.org> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <1394996270.9232.107.camel@archlinux> <20140316202636.GA23003@linuxaudio.org> Message-ID: <1395001999.9232.124.camel@archlinux> On Sun, 2014-03-16 at 20:26 +0000, Fons Adriaensen wrote: > On Sun, Mar 16, 2014 at 07:57:50PM +0100, Ralf Mardorf wrote: > > > I disagree with Fons regarding to the 44.1 KHz are as good as 48 KHz (in > > theory he might be right) > > You probably misread what I wrote - I never claimed such a thing. Many apologies :). From xiphmont at gmail.com Sun Mar 16 20:33:40 2014 From: xiphmont at gmail.com (Monty Montgomery) Date: Sun, 16 Mar 2014 16:33:40 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <201403161439.42284.gheskett@wdtv.com> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> Message-ID: > And those very good brick wall filters _will_ have a considerable group delay. ? The only way they'll have any group delay is if they're hybrids using a minimum phase section to save on gates. The linear phase section is going to do the heavy lifting in any modern design... if there's any AA filter at all. Once all the hash has all been shaped into the >1MHz region, many designs don't bother anymore. Despite the Pono and Meridian press releases, minimum phase/"apodizing" AA filters are not actually desirable. Monty On Sun, Mar 16, 2014 at 2:39 PM, Gene Heskett wrote: > On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > >> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: >> > I would mix the project at 48k or 96k >> >> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best >> sound quality. >> > That I think is a personal call Ralf, primarily because at 48 Khz, your > anti-aliasing filters had better be very very good brick walls by the time > you get above 24Khz in input content just for the aliasing control. And > aliasing noise, once introduced, cannot be removed by any known math > function that does not have precise knowledge of the phasing (aka group > delay) of the original signal. > > And those very good brick wall filters _will_ have a considerable group > delay. IMO doing the sampling at 240K, then doing a weighted sum shift (5 > stage shift) to decimate the data down to 48K, should result in dropping > the alias caused noise floor by several db. Just bring lots of expensive > hardware to do that. > >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > Cheers, Gene > -- > "There are four boxes to be used in defense of liberty: > soap, ballot, jury, and ammo. Please use in that order." > -Ed Howdershelt (Author) > Genes Web page > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From fons at linuxaudio.org Sun Mar 16 20:49:28 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 16 Mar 2014 20:49:28 +0000 Subject: [LAU] jack/oversampling In-Reply-To: <5325F291.70005@stackingdwarves.net> References: <1394985905.9232.57.camel@archlinux> <5325F291.70005@stackingdwarves.net> Message-ID: <20140316204928.GB23003@linuxaudio.org> On Sun, Mar 16, 2014 at 07:50:57PM +0100, J?rn Nettingsmeier wrote: > i don't understand this. can you elaborate? what is "blt"? Bi-Linear Transform, a mathematical trick used to transform the transfer function of an analog filter (in the s-domain) into that of a digital one (in the z-domain). It introduces a 'warping' of the frequency axis. If A(f) is the frequency response of the analog filter, and D(f) that of the digital one, they will be different but there is a function F(f) such that D(F(f)) = A(f). And it's always possible to arrange things such that the 'defining' point of the FR (the -3 dB point, or the center frequency) is correct - by applying the inverse of F(f). For low frequencies F(f) = f, so the two filter are the same. But as f->inf, F(f)->Fs/2. So filters in the upper part of the frequency range will have a different shape of the FR. For example a standard 2nd order parametric will have a symmetric shape when plotted on a log frequency scale, but the equivalent digital one won't if the center frequency is high. But there is no reason to say that one is 'better' than the other, there is no reason why any digital filter used in audio processing should be an exact copy of an analog one. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From lorenzofsutton at gmail.com Sun Mar 16 20:50:30 2014 From: lorenzofsutton at gmail.com (Lorenzo Sutton) Date: Sun, 16 Mar 2014 21:50:30 +0100 Subject: [LAU] Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <201403161439.42284.gheskett@wdtv.com> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> Message-ID: <53260E96.80008@gmail.com> On 16/03/14 19:39, Gene Heskett wrote: > On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > >> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: >>> I would mix the project at 48k or 96k >> >> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best >> sound quality. >> > That I think is a personal call Ralf, primarily because at 48 Khz, your > anti-aliasing filters had better be very very good brick walls by the time > you get above 24Khz in input content Can anyone point out a commercially available microphone used in the audio recording domain which will actually pic frequencies above 20 kHz? Likewise can anyone point out any commercially available speaker used in the audio reproduction domain which will actually reproduce frequencies above 20 kHz? If the audio produced is made for fruition of humans it makes absolutely no sense to try and capture or reproduce anything above 20kHz, and for average individuals 15kHz would probably more than enough. And in case anyone is tempted to state that even if we don't hear them frequencies above 20kHz influence the way we hear or 'perceive' music, please also attach any _scientific_ study/paper/evidence (e.g. large-scale blind tests etc. not anecdotal evidence) to such statement. Lorenzo. From alexandre.prokoudine at gmail.com Sun Mar 16 20:51:31 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 17 Mar 2014 00:51:31 +0400 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140315222109.GA20139@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> Message-ID: Hi Guido, It crashes after failing to load LADSPA plugins. http://pastebin.com/kvJk2f3K Alexandre On Sun, Mar 16, 2014 at 2:21 AM, Guido Scholz wrote: > Dear all, > > AlsaModularSynth is a MIDI controlled realtime modular synthesizer > and effect processor with support for LADSPA and JACK. > > After several years of collecting fixes and enhancements the new > release provides a long list of changes: > > > ams-2.1.0 (2014-03-15) > > > Fixed Bugs > > o Linker error "undefined reference to symbol 'dlsym@@GLIBC_2.2.5'" > on Fedora 13. > o Crash on using looped signal paths > o Armel compile error (debian #570848) > o Prevent crash if jack handle is NULL (lp #553366) > o Fix preferences dialog to show current color setup > o Fix broken Yes/No/Cancel response on program exit > o Fix memory leak in preferences widget > o Fix triggered reset of LFO saw signals, patch provided by Bill > Yerazunis > o Reorganized commandline options for input and output to > be valid for Alsa and JACK > > New Features > > o SIGUSR1 handler added to enable LADI session handling on > application level 1. > o Support for libclalsadrv API version 2.0.0 > o Support for libzita-alsa-pcmi as an alternative for libclalsadrv > o Improved port selection handling, patch provided by Sebastien > Alaiwan > o New context menu to disconnect module output ports > o Support for JACK session handling > o Rewritten preferences dialog > o New about dialog > o New option for saving window geometry (session handling) > o New option hiding recently used files menu (keep secrets) > o Add new menu item to open demo patch files directly > o Add new menu item to open demo instrument patch files directly > o Add keyboard shortcuts for module configuration dialogs > o New option for module position grid (snap to grid) > o New V8 Sequencer module, provided by Bill Yerazunis > o New Analog Memory module, provided by Bill Yerazunis > o New Bitgrinder module, provided by Bill Yerazunis > o New Hysteresis module, provided by Bill Yerazunis > o New VC-Delay module, provided by Bill Yerazunis > o Add Pulsetrain Noise type to Noise 2 module, patch provided by > Bill Yerazunis > o New FFT Vocoder module, provided by Bill Yerazunis > o Make control center window position and MIDI settings persistent > o Add support for Qt5 (configure option --enable-qt5) > > General Changes > > o Separate handling of color scheme directory from patch file > directoy. > o Improved handling of CXXFLAGS variables > o Add check for Ladspa header file. > o Obsolete commandline option -l (preset file) removed > o New commandline option for program version (--version) > > > Have fun! > > Guido > > -- > http://wie-im-flug.net/ > http://www.lug-burghausen.org/ > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From nettings at stackingdwarves.net Sun Mar 16 20:54:02 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 16 Mar 2014 21:54:02 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <20140316204928.GB23003@linuxaudio.org> References: <1394985905.9232.57.camel@archlinux> <5325F291.70005@stackingdwarves.net> <20140316204928.GB23003@linuxaudio.org> Message-ID: <53260F6A.7010403@stackingdwarves.net> On 03/16/2014 09:49 PM, Fons Adriaensen wrote: > On Sun, Mar 16, 2014 at 07:50:57PM +0100, J?rn Nettingsmeier wrote: > >> i don't understand this. can you elaborate? what is "blt"? > > Bi-Linear Transform, a mathematical trick used to transform > the transfer function of an analog filter (in the s-domain) > into that of a digital one (in the z-domain). ah, thanks. i had guessed that it probably wasn't "bacon/lettuce/tomato"... > It introduces a 'warping' of the frequency axis. If A(f) > is the frequency response of the analog filter, and D(f) > that of the digital one, they will be different but there > is a function F(f) such that D(F(f)) = A(f). And it's > always possible to arrange things such that the 'defining' > point of the FR (the -3 dB point, or the center frequency) > is correct - by applying the inverse of F(f). > > For low frequencies F(f) = f, so the two filter are the > same. But as f->inf, F(f)->Fs/2. So filters in the upper > part of the frequency range will have a different shape > of the FR. For example a standard 2nd order parametric > will have a symmetric shape when plotted on a log frequency > scale, but the equivalent digital one won't if the center > frequency is high. But there is no reason to say that one > is 'better' than the other, there is no reason why any > digital filter used in audio processing should be an exact > copy of an analog one. thanks! time to review some things... (/me grabs old hardcopy of dspguide.com) -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From clemens at ladisch.de Sun Mar 16 20:58:44 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Sun, 16 Mar 2014 21:58:44 +0100 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: References: Message-ID: <53261084.4030400@ladisch.de> Aiyumi Moriya wrote: > MIDI recording and playback work fine, but now I want to know if it > can be made to work as a soundcard. I know that the Motif XF can work > as a USB audio interface on Windows and Mac. The lsusb output shows that there is no interface for this. Does this need some kind of configuration on the device? Regards, Clemens From tim at quitte.de Sun Mar 16 21:06:12 2014 From: tim at quitte.de (Tim Goetze) Date: Sun, 16 Mar 2014 22:06:12 +0100 (CET) Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> Message-ID: [tim] >question for the reader: in order to completely prevent foldover >distortion, how much do you have to upsample for a tanh waveshaper (a >processor that introduces infinite harmonics)? To _completely_ prevent aliasing of a harmonic series extending infinitely is obviously impossible when you're limited to a finite oversampling ratio. Very slightly more practically, eventually the generated harmonics will drop below the noise floor. Where that limit lies is of course specific to your particular setup and quality demands, and strongly dependent on the amplitude of the signal to be waveshaped. As the input amplitude rises, the tanh output will approach a square wave, with harmonic amplitudes following the well-known 1/n series. Assuming this to be a desired outcome of the manipulation, a noise floor of just 72 dB results in n = 10**(72/20.) = 3981.0717055349733 and for a noise floor of 90 dB it's 10**(90/20.) = 31622.776601683792 etc. In the end, I think it's safe to assume such oversampling ratios intractable for practical purposes. Cheers, Tim From tim at klingt.org Sun Mar 16 21:09:17 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 22:09:17 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> Message-ID: >> a) any non-linearity introduces harmonics, some non-linearities >> > introduce an infinite amount of harmonics, which will cause foldover >> > distortion. the large the sampling-rate, the lower the foldover. > Although this is mostly true, it's difficult to believe it matters in > a practical audio application. If it does, in most of the cases that > are not deliberately using a highly nonlinear transfer, adding one > additional bit of depth will get you the same noise/distortion benefit > as doubling the sampling rate. Once you're using 32 bit floats, > you're already 160dB down... If you need more, I'd love to know what > you're doing, and I'm not being sarcastic! my main use-cases for high sampling rates are coupled/feedback fm/pm and stochastic synthesis (gendy). though today, i just wanted to mix some 192khz recordings and some people claimed that 48k is enough for everyone. From aiyumi.br at gmail.com Sun Mar 16 21:11:33 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Sun, 16 Mar 2014 18:11:33 -0300 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: <53261084.4030400@ladisch.de> References: <53261084.4030400@ladisch.de> Message-ID: 2014-03-16 17:58 GMT-03:00, Clemens Ladisch : > Aiyumi Moriya wrote: >> MIDI recording and playback work fine, but now I want to know if it >> can be made to work as a soundcard. I know that the Motif XF can work >> as a USB audio interface on Windows and Mac. > > The lsusb output shows that there is no interface for this. > Does this need some kind of configuration on the device? Yes, it does. -- ____________________ Blog: http://aiyumi.warpstar.net/ From ralf.mardorf at rocketmail.com Sun Mar 16 21:25:38 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 22:25:38 +0100 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> Message-ID: <1395005138.9232.128.camel@archlinux> On Sun, 2014-03-16 at 22:09 +0100, tim wrote: > though today, i just wanted to mix some 192khz recordings and some > people claimed that 48k is enough for everyone. What happens to the noise floor, when using such a high sampling rate? The noise floor of my RME card looks disgusting at 192 KHz. From tim at klingt.org Sun Mar 16 21:30:07 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 22:30:07 +0100 Subject: [LAU] Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <53260E96.80008@gmail.com> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <53260E96.80008@gmail.com> Message-ID: >> That I think is a personal call Ralf, primarily because at 48 Khz, your >> anti-aliasing filters had better be very very good brick walls by the time >> you get above 24Khz in input content > > Can anyone point out a commercially available microphone used in the > audio recording domain which will actually pic frequencies above 20 kHz? > > Likewise can anyone point out any commercially available speaker used in > the audio reproduction domain which will actually reproduce frequencies > above 20 kHz? my roughly 20 years old genelec s30 are specified to go to >25khz, mundorf amt tweeters go up to 41khz, adam s4x-h are specified do go up to 50khz. the mundorf amt's can be used for PA speakers, a college of mine once used them to build a line array > If the audio produced is made for fruition of humans it makes absolutely > no sense to try and capture or reproduce anything above 20kHz, and for > average individuals 15kHz would probably more than enough. but of course you need to distinguish between distribution and production, were you can benefit from frequency headroom. > And in case anyone is tempted to state that even if we don't hear them > frequencies above 20kHz influence the way we hear or 'perceive' music, > please also attach any _scientific_ study/paper/evidence (e.g. > large-scale blind tests etc. not anecdotal evidence) to such statement. > > Lorenzo. > From nettings at stackingdwarves.net Sun Mar 16 21:34:31 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 16 Mar 2014 22:34:31 +0100 Subject: [LAU] Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <53260E96.80008@gmail.com> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <53260E96.80008@gmail.com> Message-ID: <532618E7.4050906@stackingdwarves.net> On 03/16/2014 09:50 PM, Lorenzo Sutton wrote: > On 16/03/14 19:39, Gene Heskett wrote: >> On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: >> >>> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: >>>> I would mix the project at 48k or 96k >>> >>> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best >>> sound quality. >>> >> That I think is a personal call Ralf, primarily because at 48 Khz, your >> anti-aliasing filters had better be very very good brick walls by the >> time >> you get above 24Khz in input content > > Can anyone point out a commercially available microphone used in the > audio recording domain which will actually pic frequencies above 20 kHz? i once talked to a bat researcher (no joke) who had a simple mod to a r?de nt5 that would allow it to work reasonably well up to 30k. earthworks make special versions of their excellent microphones which are linear up to 50khz, for those who need it (or think they need it). in fact, most if not all microphones can do this but for additional filters added by the manufacturers to increase robustness in the presence of HF electromagnetic noise. the question is how linear they are up there, and whether the pickup pattern is still useful. all but the smallest capsules can be expected to become highly directional at HF. something like a DPA 4060 or a SH MKE-1 might be good experimental capsules for ultrasonics. > Likewise can anyone point out any commercially available speaker used in > the audio reproduction domain which will actually reproduce frequencies > above 20 kHz? most tweeters are easily capable of 30khz, and there are no major engineering obstacles to 50khz on-axis. the big issue is excessive beaming, both on the recording and playback side of things. > If the audio produced is made for fruition of humans it makes absolutely > no sense to try and capture or reproduce anything above 20kHz, and for > average individuals 15kHz would probably more than enough. i'd tend to agree with that statement, but there are very valid reasons to do it: * not all recordings are meant for humans to hear - if you are measuring something, you might appreciate results outside of human sensation. * not all recordings are meant to be heard in its original frequency range - talk to any bat enthusiast. seriously, what those guys do makes you itch to try 192khz and a microphone that is open "from dc to daylight", as the saying goes. * sometimes, preservation of information is extremely important. for instance, there are valid reasons to digitize old analog tapes at ridiculous rates (say, 384 kHz): doing so lets you record traces of the HF bias, which might help in eliminating wow and flutter artefacts more precisely than tracking the 50 or 60hz power grid hum. or there's a colleague from italy, david monacchi, who records sound scapes in soon-to-be-destroyed natural habitats - why would you limit yourself to 10 octaves if you can get 11, before the bulldozers arrive? (i once heard him lecture on one of his works, and indeed he was using sonograms to identify certain species of animals, many of which are capable of uttering ultrasonics.) there is still ongoing debate about indirect audibility of high frequency content via transients - i'm not too convinced, but i can understand any colleague who would rather record too much today and then downsample, as opposed to finding you won't be able to fully exploit future distribution formats with your legacy material. if i'm not maxing out my equipment in terms of cpu cycles, there is no harm done in erring on the side of caution, if high sample rates don't incur higher costs as they go through the workflow. as a counter-example, a tv production i'm involved in uses 96k initially, but only because the live sound desk is a midas which cannot do less. it is immediately downsampled to 48k before going onto the broadcast network via dante, because nobody wants to put up with the extra data and doubled loadin/transfer times. > And in case anyone is tempted to state that even if we don't hear them > frequencies above 20kHz influence the way we hear or 'perceive' music, > please also attach any _scientific_ study/paper/evidence (e.g. > large-scale blind tests etc. not anecdotal evidence) to such statement. actually, i just had a behringer ada8000 die on me, and i will probably replace it with some 96khz capable kit from directout. i don't expect to find anything interesting, like you said, but i'm going to try it nonetheless. i know that my tweeters will only begin to roll off at 40khz. i'm not advocating high sample rates, but hey, "because i can" has always been a strong incentive :-D the main problem is that some equipment actually gets worse at high sample rates, and putting a 192k sticker on your box is actually more important than getting 48k really right... -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From tim at klingt.org Sun Mar 16 21:37:51 2014 From: tim at klingt.org (tim) Date: Sun, 16 Mar 2014 22:37:51 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1395005138.9232.128.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> <1395005138.9232.128.camel@archlinux> Message-ID: >> though today, i just wanted to mix some 192khz recordings and some >> people claimed that 48k is enough for everyone. > > What happens to the noise floor, when using such a high sampling rate? > The noise floor of my RME card looks disgusting at 192 KHz. you must have realized, that i don't care about the sampling rate for recording&playback, but the headroom in the digial domain ... fwiw, you may want to consult the manual: http://www.manualslib.com/manual/514035/Rme-Fireface-Ucx.html?page=96 From nettings at stackingdwarves.net Sun Mar 16 21:42:37 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 16 Mar 2014 22:42:37 +0100 Subject: [LAU] jack/oversampling In-Reply-To: <1395005138.9232.128.camel@archlinux> References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> <1395005138.9232.128.camel@archlinux> Message-ID: <53261ACD.3020908@stackingdwarves.net> On 03/16/2014 10:25 PM, Ralf Mardorf wrote: > On Sun, 2014-03-16 at 22:09 +0100, tim wrote: >> though today, i just wanted to mix some 192khz recordings and some >> people claimed that 48k is enough for everyone. > > What happens to the noise floor, when using such a high sampling rate? > The noise floor of my RME card looks disgusting at 192 KHz. reading such data requires some interpretation. keep in mind you are now looking at four times the bandwidth (which will contain more energy), and that you may be seeing intentional noise shaping (which would be a real problem for bat fans, unfortunately, but is very beneficial for humans). -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From ralf.mardorf at rocketmail.com Sun Mar 16 21:53:23 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 16 Mar 2014 22:53:23 +0100 Subject: [LAU] OT: Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <532618E7.4050906@stackingdwarves.net> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <53260E96.80008@gmail.com> <532618E7.4050906@stackingdwarves.net> Message-ID: <1395006803.9232.133.camel@archlinux> Infrasound and ultrasonic are not important for music recordings. AFIK bats produce sound up to 200 KHz, so 192 KHz anyway can't do the job. A former girl friend did bat research :), but we never talked about bat audio, just about bat behaviour :), so I only have "Wiki" knowledge about bat's audio abilities and this might be wrong. From lorenzofsutton at gmail.com Sun Mar 16 21:54:27 2014 From: lorenzofsutton at gmail.com (Lorenzo Sutton) Date: Sun, 16 Mar 2014 22:54:27 +0100 Subject: [LAU] Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <532618E7.4050906@stackingdwarves.net> References: <1394985905.9232.57.camel@archlinux> <201403161439.42284.gheskett@wdtv.com> <53260E96.80008@gmail.com> <532618E7.4050906@stackingdwarves.net> Message-ID: <53261D93.1020108@gmail.com> On 16/03/14 22:34, J?rn Nettingsmeier wrote: > On 03/16/2014 09:50 PM, Lorenzo Sutton wrote: >> On 16/03/14 19:39, Gene Heskett wrote: >>> On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: >>> >>>> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: >>>>> I would mix the project at 48k or 96k >>>> >>>> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best >>>> sound quality. >>>> >>> That I think is a personal call Ralf, primarily because at 48 Khz, your >>> anti-aliasing filters had better be very very good brick walls by the >>> time >>> you get above 24Khz in input content >> >> Can anyone point out a commercially available microphone used in the >> audio recording domain which will actually pic frequencies above 20 kHz? > > i once talked to a bat researcher (no joke) who had a simple mod to a > r?de nt5 that would allow it to work reasonably well up to 30k. > > earthworks make special versions of their excellent microphones which > are linear up to 50khz, for those who need it (or think they need it). ok ok... :-) I was obviously being a bit sarcastic in my statements.. [...] >> If the audio produced is made for fruition of humans it makes absolutely >> no sense to try and capture or reproduce anything above 20kHz, and for >> average individuals 15kHz would probably more than enough. > > i'd tend to agree with that statement, but there are very valid reasons > to do it: > * not all recordings are meant for humans to hear - if you are measuring > something, you might appreciate results outside of human sensation. > * not all recordings are meant to be heard in its original frequency > range - talk to any bat enthusiast. seriously, what those guys do makes > you itch to try 192khz and a microphone that is open "from dc to > daylight", as the saying goes. > * sometimes, preservation of information is extremely important. for > instance, there are valid reasons to digitize old analog tapes at > ridiculous rates (say, 384 kHz): doing so lets you record traces of the > HF bias, which might help in eliminating wow and flutter artefacts more > precisely than tracking the 50 or 60hz power grid hum. > or there's a colleague from italy, david monacchi, who records sound > scapes in soon-to-be-destroyed natural habitats - why would you limit > yourself to 10 octaves if you can get 11, before the bulldozers arrive? > (i once heard him lecture on one of his works, and indeed he was using > sonograms to identify certain species of animals, many of which are > capable of uttering ultrasonics.) > > there is still ongoing debate about indirect audibility of high > frequency content via transients - i'm not too convinced, but i can > understand any colleague who would rather record too much today and then > downsample, as opposed to finding you won't be able to fully exploit > future distribution formats with your legacy material. if i'm not maxing > out my equipment in terms of cpu cycles, there is no harm done in erring > on the side of caution, if high sample rates don't incur higher costs as > they go through the workflow. I agree with you actually, especially with all the 'preservation' scenarios... I was just being a bit provocative as to overrating gear specs.. Seriously, in all the cases you mention there is at least a thorough thought behind choice of specs and requirements for recording, and I like that. I was also thinking of all the counter-examples, in the _music_ domain, recorded on technically not-so-hi-quality gear which is just great music :-) Lorenzo. From fons at linuxaudio.org Sun Mar 16 22:36:41 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 16 Mar 2014 22:36:41 +0000 Subject: [LAU] jack/oversampling In-Reply-To: References: <1394985905.9232.57.camel@archlinux> <20140316183253.GA32563@linuxaudio.org> <1395005138.9232.128.camel@archlinux> Message-ID: <20140316223641.GC23003@linuxaudio.org> On Sun, Mar 16, 2014 at 10:37:51PM +0100, tim wrote: > you must have realized, that i don't care about the sampling rate for > recording&playback, but the headroom in the digial domain ... The F-range headroom is convenient, but not necessary. > fwiw, you may want to consult the manual: > http://www.manualslib.com/manual/514035/Rme-Fireface-Ucx.html?page=96 This misses the point that for 24-bit converters the noise floor will be determined by the analog electronics. In other words any noise shaping is futile. It helps only for 16-bit systems. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From lorenzofsutton at gmail.com Sun Mar 16 23:04:11 2014 From: lorenzofsutton at gmail.com (Lorenzo Sutton) Date: Mon, 17 Mar 2014 00:04:11 +0100 Subject: [LAU] Proposal for 'lo-fi' music competition Message-ID: <53262DEB.5080207@gmail.com> In light of the interesting discussion on sample rates I propose a music competition among LAU around production of music pieces with quality considered 'low' by current dominating professional/audiophile standards in the digital domain: Specifics to be discussed, but I would start with the following: 1. Final piece shall be delivered with a maximum sampling rate of 32kHz. Lower sample rates .allowed. 2. Final bit depth of the file shall not exceed 8 bit. - Possible variations/additions to this restriction: - allow dithering? - allow 4/8bit codecs (e.g. ADPCM, ulaw)? - mono/stereo? - simply impose a max duration and max filesize (but keep sr and bit thresholds... and variations..)? - Participants shall strongly focus on exploring and exploiting the imposed limitations in a creative and artistic way Other specs, voting mechanisms, competition arrangements, prizes, ... ??? Ciao, Lorenzo. From gheskett at wdtv.com Mon Mar 17 00:17:02 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Sun, 16 Mar 2014 20:17:02 -0400 Subject: [LAU] jack/oversampling In-Reply-To: <1395001699.9232.121.camel@archlinux> References: <201403161552.19316.gheskett@wdtv.com> <1395001699.9232.121.camel@archlinux> Message-ID: <201403162017.02086.gheskett@wdtv.com> On Sunday 16 March 2014 19:56:32 Ralf Mardorf did opine: > Gene :) > > ok, when I've got the time to do it, I will borrow the Brauner VM-1 tube > mic + an ART tube pre amp from a friend, connect it directly to my RME > HDSPe AIO and pick Blackbird from the Beatles on my classical guitar and > do a recording at 48 KHz and 192 KHz. > > I don't know what will happen, but I suspect there will be no audible > difference. > > Regards, > Ralf > > Even with tons of string squeaks from wound strings and heavily callused fingers, I doubt if that would reach high enough to be a valid test. You need a few crickets singing in the background, (typically 17 Khz) and some heavy action on a brushed snare drum to generate very much in the range of sound that would test it for sure. A "white noise" generator made out of the usual 17 stage xor gated shift register for feedback, clocked at 50Khz or more might be a good test, the white noise will seem to have a odd, often disagreeable, definitely non-harmonious pitch to it. But I haven't seen that schematic in yonks. Take a look at Some of the longer ones are pretty decent. the 17th register addition is one I heard once, sounded pretty white to my ears at the time. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From joel at weedlight.ch Mon Mar 17 00:26:40 2014 From: joel at weedlight.ch (=?ISO-8859-1?Q?Jo=EBl_Kr=E4hemann?=) Date: Mon, 17 Mar 2014 01:26:40 +0100 Subject: [LAU] stable release of Advanced Gtk+ Sequencer planed Message-ID: <1395016000.4816.34.camel@debian> Hello, I want to make my first release of a software based audio production environment. I did a 30 day roadmap therefore. Every feedback will be considered. Please contact if your interested in helping to advance Advanced Gtk+ Sequencer development. https://sourceforge.net/p/ags/wiki/release-0_4_0 Hint: there's currently no midi binding thanks in advance Jo?l From gheskett at wdtv.com Mon Mar 17 00:56:33 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Sun, 16 Mar 2014 20:56:33 -0400 Subject: [LAU] Sampling rates [WAS]: Re: jack/oversampling In-Reply-To: <53260E96.80008@gmail.com> References: <201403161439.42284.gheskett@wdtv.com> <53260E96.80008@gmail.com> Message-ID: <201403162056.33769.gheskett@wdtv.com> On Sunday 16 March 2014 20:19:50 Lorenzo Sutton did opine: > On 16/03/14 19:39, Gene Heskett wrote: > > On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine: > >> On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote: > >>> I would mix the project at 48k or 96k > >> > >> Why 96 KHz? 48 KHz doesn't cause any issues, but already provides > >> best sound quality. > > > > That I think is a personal call Ralf, primarily because at 48 Khz, > > your anti-aliasing filters had better be very very good brick walls > > by the time you get above 24Khz in input content > > Can anyone point out a commercially available microphone used in the > audio recording domain which will actually pic frequencies above 20 kHz? > An Altec M-21b, new in about 1955 or so, probably 40 years out of production now has usable response beyond 20Khz. The PZM, a similar but electret powered condenser mike, new in about 1985 IIRC also goes up into that range. > Likewise can anyone point out any commercially available speaker used in > the audio reproduction domain which will actually reproduce frequencies > above 20 kHz? Is the Altec Lansing 075 ring radiator tweeter still available? I once saw some scope photos of the output of an M-21b, 3' away & on axis of the sound output of the 075 driven by approximately a watt of a square wave at 25Khz. The square wave was still recognizable. In the '50's we would set an 075 in the center pocket of the Cobra Horn of a JBL Hartsfield speaker, a huge corner horn that was very good before we did that, and made it raise the hair on the back of your head real if your eyes were closed and a 30ips recording of the Dukes of Dixieland was playing from 10.5" reels on a Berlant-Concertone tape deck. The only thing that gave it away was the tape hiss, a good 68db down. There may yet be a few of those around, but very few since it sold new in the 1950's for $750. That tweeter added another 100 to the price. The Hartsfield was fairly efficient too, better than the Voice at the time. We could hit 130+db in the front room at Woodburn Sound Service in Iowa City Iowa, with a 10 watt rated Fisher amplifier driving it. And I am talking 130 very painful db's. > If the audio produced is made for fruition of humans it makes absolutely > no sense to try and capture or reproduce anything above 20kHz, and for > average individuals 15kHz would probably more than enough. > > And in case anyone is tempted to state that even if we don't hear them > frequencies above 20kHz influence the way we hear or 'perceive' music, > please also attach any _scientific_ study/paper/evidence (e.g. > large-scale blind tests etc. not anecdotal evidence) to such statement. This is, according to your definition, anecdotal. I spent 2 years Chiefing at a radio station in No. Kalipornia, who had one of those ultrasonic motion detector burglar alarms. It wasn't working so the first thing I did was fix it, dead battery. Then I had to rig it with a disabling switch to shut off the power amp that drove the piezo speakers in it. I couldn't hear it as such, but my ears would feel like I had just walked into a high pressure chamber that yawning wouldn't fix, and in 5 minutes I had a splitting headache. So it wound up with a switch that the owner could turn on when he left for the day. To this day I can walk into a shop with one of those things and tell them if its running right, surprising many a shop owner who thought the installation was a secret even from his employees. FWIW, they aren't worth it because of the false alarms they generate when the wind is blowing against those huge plate glass front windows which can move 1/2" or more in a good spring breeze. Sets them off every time. To the LE people, they are the little boy crying wolf and are ignored. > Lorenzo. > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From paul at linuxaudiosystems.com Mon Mar 17 02:20:36 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 16 Mar 2014 22:20:36 -0400 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <53262DEB.5080207@gmail.com> References: <53262DEB.5080207@gmail.com> Message-ID: On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton wrote: > In light of the interesting discussion on sample rates I propose a music > competition among LAU around production of music pieces with quality > considered 'low' by current dominating professional/audiophile standards in > the digital domain: > > Specifics to be discussed, but I would start with the following: > can we just re-record the beatles or miles davis from vinyl and consider it done? -------------- next part -------------- An HTML attachment was scrubbed... URL: From byronkeys at jvlnet.com Mon Mar 17 05:17:36 2014 From: byronkeys at jvlnet.com (Brian Hagen) Date: Mon, 17 Mar 2014 00:17:36 -0500 Subject: [LAU] legacy equipment for a user who does not "program" Message-ID: <53268570.3010907@jvlnet.com> Hello, Although I was a programmer a long, long, time ago in a company far, far away, I cannot seem to get the hang of complicated Linux packages which require complex actions just to get an "app installed" on a Linux machine. I have an Ensoniq ESQ1 which just seems unable to interact with any WIN/OS-based software, despite what people tout as easy-to-use programs. I have tried to download and install Linux MIDI items, and almost all of them require complex configuration even to get them into the system. All I really want to do at this point is get an application running which will upload and download patches; nothing yet in the way of realtime operations such as recording. A big part of this appears to be the MIDI hardware used as an interface; most of those "apps" expect to find the classic Roland MPU401, and those are scarce as could be these days. The old apps apparently cannot recognize anything based on USB ports. Yes, I can do a make/install sequence, but those resulting error messages usually point to matters beyond my understanding such as GTK+ levels, " missing [.so] files", etc. Those usually stop me in my tracks, and then things don't go any farther. Ideas, anyone? Brian From gnome at hawaii.rr.com Mon Mar 17 05:25:32 2014 From: gnome at hawaii.rr.com (david) Date: Sun, 16 Mar 2014 19:25:32 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: References: <53262DEB.5080207@gmail.com> Message-ID: <5326874C.60800@hawaii.rr.com> On 03/16/2014 04:20 PM, Paul Davis wrote: > > On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton > > wrote: > > In light of the interesting discussion on sample rates I propose a > music competition among LAU around production of music pieces with > quality considered 'low' by current dominating > professional/audiophile standards in the digital domain: > > Specifics to be discussed, but I would start with the following: > > can we just re-record the beatles or miles davis from vinyl and consider > it done? Or maybe you mean how music sounded on old battery-powered AM portable radios? I've heard quite a bit of it in the last few days, it seems to be the sound quality that Amazon Kindle Support provides for their "music on hold" ... Or 1/4" inch tape cassettes? 8-track tapes? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From simonzwise at gmail.com Mon Mar 17 05:39:55 2014 From: simonzwise at gmail.com (Simon Wise) Date: Mon, 17 Mar 2014 16:39:55 +1100 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <53268570.3010907@jvlnet.com> References: <53268570.3010907@jvlnet.com> Message-ID: <53268AAB.3010608@gmail.com> On 17/03/14 16:17, Brian Hagen wrote: > Hello, > Although I was a programmer a long, long, time ago in a > company far, far away, I cannot seem to get the hang of complicated > Linux packages which require complex actions just to get an > "app installed" on a Linux machine. > > I have an Ensoniq ESQ1 which just seems unable to interact > with any WIN/OS-based software, despite what people tout > as easy-to-use programs. I have tried to download and install > Linux MIDI items, and almost all of them require complex configuration > even to get them into the system. All I really want to do at this > point is get an application running which will upload and download > patches; nothing yet in the way of realtime operations such as recording. > A big part of this appears to be the MIDI hardware used as an > interface; most of those "apps" expect to find the classic > Roland MPU401, and those are scarce as could be these days. > The old apps apparently cannot recognize anything based > on USB ports. > > Yes, I can do a make/install sequence, but those resulting > error messages usually point to matters beyond my understanding > such as GTK+ levels, " missing [.so] files", etc. Those usually stop me > in my tracks, and then things don't go any farther. > > Ideas, anyone? Before trying to compile anything it is much easier to start with packages already prepared for your system. Depending on what Linux system you have then there are in most cases a wide range of ready-to-use apps available for installing via a system that downloads the binary already compiled for that system along with the library objects it uses (the .so files). This is by far the easiest way to try out different programs, and a lot easier than downloading code or binaries then trying to make them fit your particular setup. That can come later, if you wish. Which Linux system have you got? Simon From gnome at hawaii.rr.com Mon Mar 17 06:41:31 2014 From: gnome at hawaii.rr.com (david) Date: Sun, 16 Mar 2014 20:41:31 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <53262DEB.5080207@gmail.com> References: <53262DEB.5080207@gmail.com> Message-ID: <5326991B.7010008@hawaii.rr.com> OK, now I've got to track down a way to digitize a mix tape I made for my first girlfriend, back in early 70's. It was a mix tape of my own songs, including one with me singing (badly) and playing bottle neck guitar (passably) on a cheap steel-string acoustic that couldn't stay in tune for more than a minute. Recorded onto 1/4" cassette using a portable tape recorder designed to record voice dictation or interview, not music. About as lo-fi as you could get, even by standards back then! On 03/16/2014 01:04 PM, Lorenzo Sutton wrote: > In light of the interesting discussion on sample rates I propose a music > competition among LAU around production of music pieces with quality > considered 'low' by current dominating professional/audiophile standards > in the digital domain: > > Specifics to be discussed, but I would start with the following: > > 1. Final piece shall be delivered with a maximum sampling rate of 32kHz. > Lower sample rates .allowed. > > 2. Final bit depth of the file shall not exceed 8 bit. > - Possible variations/additions to this restriction: > - allow dithering? > - allow 4/8bit codecs (e.g. ADPCM, ulaw)? > - mono/stereo? > - simply impose a max duration and max filesize (but keep sr > and bit thresholds... and variations..)? > > > - Participants shall strongly focus on exploring and exploiting the > imposed limitations in a creative and artistic way > > Other specs, voting mechanisms, competition arrangements, prizes, ... ??? > > Ciao, > Lorenzo. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gordonjcp at gjcp.net Mon Mar 17 07:28:51 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Mon, 17 Mar 2014 07:28:51 +0000 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <53268570.3010907@jvlnet.com> References: <53268570.3010907@jvlnet.com> Message-ID: <20140317072851.GA13487@gjcp.net> On Mon, Mar 17, 2014 at 12:17:36AM -0500, Brian Hagen wrote: > Hello, > Although I was a programmer a long, long, time ago in a > company far, far away, I cannot seem to get the hang of complicated > Linux packages which require complex actions just to get an > "app installed" on a Linux machine. Errr, what? > I have an Ensoniq ESQ1 which just seems unable to interact > with any WIN/OS-based software, despite what people tout > as easy-to-use programs. I have tried to download and install > Linux MIDI items, and almost all of them require complex configuration > even to get them into the system. All I really want to do at this Errrr, what? Example, please? > point is get an application running which will upload and download > patches; nothing yet in the way of realtime operations such as recording. > A big part of this appears to be the MIDI hardware used as an > interface; most of those "apps" expect to find the classic > Roland MPU401, and those are scarce as could be these days. > The old apps apparently cannot recognize anything based > on USB ports. Which "old apps"? Everything has used ALSA for at least the past 15 years. The app does not care (and indeed cannot even see) what the hardware actually is, just that it's a thing it can stuff MIDI messages into and sometimes MIDI messages come out. > > Yes, I can do a make/install sequence, but those resulting > error messages usually point to matters beyond my understanding > such as GTK+ levels, " missing [.so] files", etc. Those usually stop me > in my tracks, and then things don't go any farther. Stop whining about complexity and give us some details. Which distro are you using? Which specific apps? -- Gordonjcp MM0YEQ From gordonjcp at gjcp.net Mon Mar 17 07:33:04 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Mon, 17 Mar 2014 07:33:04 +0000 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <53268570.3010907@jvlnet.com> References: <53268570.3010907@jvlnet.com> Message-ID: <20140317073304.GB13487@gjcp.net> On Mon, Mar 17, 2014 at 12:17:36AM -0500, Brian Hagen wrote: > Ideas, anyone? > > Brian Incidentally, when my ESQ-1 was working (I need to replace the display and display board processor) I just used amidi to send sysex backwards and forwards. It works perfectly. About the only thing it doesn't really work for is Casio CZ stuff, because of their batshit insane sysex protocol. -- Gordonjcp MM0YEQ From cbannister at slingshot.co.nz Mon Mar 17 08:30:20 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Mon, 17 Mar 2014 21:30:20 +1300 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <5326874C.60800@hawaii.rr.com> References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> Message-ID: <20140317083020.GB9131@tal> On Sun, Mar 16, 2014 at 07:25:32PM -1000, david wrote: > Or maybe you mean how music sounded on old battery-powered AM > portable radios? Isn't stereo AM supposed to be better than stereo FM? P.S. I've never personally heard stereo AM, but the stations compress the bejesus out of their FM transmissions -- bloody horrible! -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X From clemens at ladisch.de Mon Mar 17 08:37:19 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Mon, 17 Mar 2014 09:37:19 +0100 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: References: <53261084.4030400@ladisch.de> Message-ID: <5326B43F.3090101@ladisch.de> Aiyumi Moriya wrote: > 2014-03-16 17:58 GMT-03:00, Clemens Ladisch : >> Aiyumi Moriya wrote: >>> MIDI recording and playback work fine, but now I want to know if it >>> can be made to work as a soundcard. I know that the Motif XF can work >>> as a USB audio interface on Windows and Mac. >> >> The lsusb output shows that there is no interface for this. >> Does this need some kind of configuration on the device? > > Yes, it does. And what is the lsusb output after you have enabled this? Regards, Clemens From rmouneyres at gmail.com Mon Mar 17 08:46:51 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Mon, 17 Mar 2014 09:46:51 +0100 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <20140317083020.GB9131@tal> References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> <20140317083020.GB9131@tal> Message-ID: You can take a look at the passed Speedbass event called Spitbass with a similar concept back in 2002. 8 bit, 8KHz electro music. http://www.speedbass.net/Downloads/sublevel/cid/2/start/0 Checking back there i noticed my (Jerash) contributions have broken links. So i'll make them up again to apply the competition. Rapha?l 2014-03-17 9:30 UTC+01:00, Chris Bannister : > On Sun, Mar 16, 2014 at 07:25:32PM -1000, david wrote: >> Or maybe you mean how music sounded on old battery-powered AM >> portable radios? > > Isn't stereo AM supposed to be better than stereo FM? > > P.S. I've never personally heard stereo AM, but the stations compress > the bejesus out of their FM transmissions -- bloody horrible! > > -- > "If you're not careful, the newspapers will have you hating the people > who are being oppressed, and loving the people who are doing the > oppressing." --- Malcolm X > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From edogawa at aon.at Mon Mar 17 09:00:18 2014 From: edogawa at aon.at (Edgar Aichinger) Date: Mon, 17 Mar 2014 10:00:18 +0100 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <20140317073304.GB13487@gjcp.net> References: <53268570.3010907@jvlnet.com> <20140317073304.GB13487@gjcp.net> Message-ID: <1810910.PLeUGHZ99R@edhp> Am Montag, 17. M?rz 2014, 07:33:04 schrieb Gordon JC Pearce: > On Mon, Mar 17, 2014 at 12:17:36AM -0500, Brian Hagen wrote: > > > > > Ideas, anyone? > > > > Brian > > Incidentally, when my ESQ-1 was working (I need to replace the display and display board processor) I just used amidi to send sysex backwards and forwards. It works perfectly. About the only thing it doesn't really work for is Casio CZ stuff, because of their batshit insane sysex protocol. > > And then there's a simple and clean Qt4 based GUI app for that task, called Simple Sysexxer: http://www.christeck.de/wp/products/simple-sysexxer/ This works well for me, tested with Roland GR-33 and Korg Wavestation. I'm supplying binary packages for several openSUSE versions and Fedora 19, in my OBS home project: http://download.opensuse.org/repositories/home:/edogawa/ Hope that helps, Edgar From pshirkey at boosthardware.com Mon Mar 17 09:26:20 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Mon, 17 Mar 2014 20:26:20 +1100 (EST) Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <53262DEB.5080207@gmail.com> References: <53262DEB.5080207@gmail.com> Message-ID: <57312.86.107.254.57.1395048380.squirrel@boosthardware.com> On Mon, March 17, 2014 10:04 am, Lorenzo Sutton wrote: > In light of the interesting discussion on sample rates I propose a music > competition among LAU around production of music pieces with quality > considered 'low' by current dominating professional/audiophile standards > in the digital domain: > > Specifics to be discussed, but I would start with the following: > > 1. Final piece shall be delivered with a maximum sampling rate of 32kHz. > Lower sample rates .allowed. > > 2. Final bit depth of the file shall not exceed 8 bit. > - Possible variations/additions to this restriction: > - allow dithering? > - allow 4/8bit codecs (e.g. ADPCM, ulaw)? > - mono/stereo? > - simply impose a max duration and max filesize (but keep sr and bit > thresholds... and variations..)? > > > - Participants shall strongly focus on exploring and exploiting the > imposed limitations in a creative and artistic way > > Other specs, voting mechanisms, competition arrangements, prizes, ... ??? > A few ideas for the mix. - (one of) The prize(s) could be one of these: https://www-ssl.intel.com/content/www/us/en/nuc/nuc-board-dn2820fykh.html They go for around $130. Add in HDD and RAM for approx $300 total. I'm sure we could all pitch in to make that possible. $10 each x 30 people. or $1 each x 300 people. There are several thousand subscribers to LAU/LAD these days and all the associated forums/mailing lists and irc channels. - Alternatively we could offer cheaper allwinner or rikomagik boards that have been fully preconfigured with a custom Linux OS that actually works (not android because that is fairly pointless). - We could create a webpage for the competition and some banner ads in various sizes and ask members of the community to host them on their websites linking back to the main competition page. - It might also be possible to get some of the other Linux Audio companies to pitch in with license keys for various proprietary Linux software or other physical hardware. - Maybe some of the Linux magazines will be prepared to offer subscriptions in exchange for promotional logo placement on the promotional webpage/media. - We could hold an awards ceremony/event at a bar/venue and live stream the shortlisted tunes in a town/city where there is a large group of Linux Audio developers. Possibly the LAC would make a good forum for such an event. - We could provide a permanent radio stream of all the entries hosted by one of the community focused websites like linuxaudio.org or linux-audio.com - We could ask major global corporations who make ridiculous sums of money from Linux Audio technology (ex. alsa, pulse audio) to contribute prizes, money, resources... - Perhaps one of the numerous car companies that uses Linux for their multimedia system will be interested in contributing. For example, we could contact Tesla Motors to see if they want to throw anything into the pot. An umbrella, t-shirt, other corporate mechandise that they routinely give away at other events. - We could ask of the people who work on the audio team at Ubuntu to ask his boss if he would like to contribute anything. - We could ask one of the people who work on audio at Intel if they can get a contribution. - We could ask one of the people who work on audio at Samsung if they can get a contribution. - We could ask the Linux Foundation if they have any spare Tizen devices floating around that they would like to contribute. - We could ask Valve if they would like to contribute a new SteamOS machine that runs Linux and uses ALSA for the audio system. Or maybe they have something else they can contribute that is not worth as much to them but will still have value tot he recipient. We could do all of these things if we wanted to be organised about promoting the competition. I have time and resources but in the past when these ideas have been suggested no one else wanted to participate so nothing has come together. Maybe now there are some other people who have the time/motivation too? -- Patrick Shirkey Boost Hardware Ltd From ralf.mardorf at rocketmail.com Mon Mar 17 09:26:13 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 17 Mar 2014 10:26:13 +0100 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <20140317072851.GA13487@gjcp.net> References: <53268570.3010907@jvlnet.com> <20140317072851.GA13487@gjcp.net> Message-ID: <1395048373.9232.147.camel@archlinux> On Mon, 2014-03-17 at 07:28 +0000, Gordon JC Pearce wrote: > Which distro are you using? $ cat /etc/issue ; uname -m From neilcsmith.net at googlemail.com Mon Mar 17 09:43:14 2014 From: neilcsmith.net at googlemail.com (Neil C Smith) Date: Mon, 17 Mar 2014 09:43:14 +0000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <5326991B.7010008@hawaii.rr.com> References: <53262DEB.5080207@gmail.com> <5326991B.7010008@hawaii.rr.com> Message-ID: On 17 March 2014 06:41, david wrote: > OK, now I've got to track down a way to digitize a mix tape I made for my > first girlfriend, back in early 70's. It was a mix tape of my own songs, > including one with me singing (badly) and playing bottle neck guitar > (passably) on a cheap steel-string acoustic that couldn't stay in tune for > more than a minute. Recorded onto 1/4" cassette using a portable tape > recorder designed to record voice dictation or interview, not music. I've heard that love is blind .. it would seem it's deaf too! :-P N -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source intermedia development - www.praxislive.org Digital Prisoners - interactive spaces and projections - www.digitalprisoners.co.uk From lorenzofsutton at gmail.com Mon Mar 17 10:00:17 2014 From: lorenzofsutton at gmail.com (Lorenzo Sutton) Date: Mon, 17 Mar 2014 11:00:17 +0100 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: References: <53262DEB.5080207@gmail.com> Message-ID: <5326C7B1.1070606@gmail.com> On 17/03/2014 03:20, Paul Davis wrote: > > > > On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton > > wrote: > > In light of the interesting discussion on sample rates I propose a > music competition among LAU around production of music pieces with > quality considered 'low' by current dominating > professional/audiophile standards in the digital domain: > > Specifics to be discussed, but I would start with the following: > > > can we just re-record the beatles or miles davis from vinyl and consider > it done? > You got he point - well said :-) From harryhaaren at gmail.com Mon Mar 17 11:18:55 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Mon, 17 Mar 2014 11:18:55 +0000 Subject: [LAU] OpenAV ArtyFX 1.1 Release Message-ID: Hey All, It's my pleasure announce that ArtyFX 1.1 is released! See the ArtyFX page for details on the 3 new plugins: http://openavproductions.com/artyfx/ Source available from github: https://github.com/harryhaaren/openAV-ArtyFX/releases And many thanks to the contributors! -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Mon Mar 17 12:46:38 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Mon, 17 Mar 2014 08:46:38 -0400 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <53268570.3010907@jvlnet.com> References: <53268570.3010907@jvlnet.com> Message-ID: On Mon, Mar 17, 2014 at 1:17 AM, Brian Hagen wrote: > Hello, > Although I was a programmer a long, long, time ago in a > company far, far away, I cannot seem to get the hang of complicated > Linux packages which require complex actions just to get an > "app installed" on a Linux machine. > Then don't. If there is software that you might want to use, and it is not installable on whatever version of Linux you use with less than, oh, 5 mouse button clicks, then take it as a sign that you probably don't want to use it. Installing Linux software from source has more or less vanished as part of the work flow for non-technical users, and even for majority of what technical users do. --p -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Mon Mar 17 13:07:45 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 17 Mar 2014 14:07:45 +0100 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: References: <53268570.3010907@jvlnet.com> Message-ID: <1395061665.9232.187.camel@archlinux> On Mon, 2014-03-17 at 08:46 -0400, Paul Davis wrote: > Installing Linux software from source has more or less vanished as > part of the work flow for non-technical users, and even for majority > of what technical users do. Paul is right, but there anyway is a "halfway house", Arch Linux does provide the Arch User Repository. From jh at brainiac.com Mon Mar 17 13:15:28 2014 From: jh at brainiac.com (Joe Hartley) Date: Mon, 17 Mar 2014 09:15:28 -0400 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <1395061665.9232.187.camel@archlinux> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> Message-ID: <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> On Mon, 17 Mar 2014 14:07:45 +0100 Ralf Mardorf wrote: > On Mon, 2014-03-17 at 08:46 -0400, Paul Davis wrote: > > Installing Linux software from source has more or less vanished as > > part of the work flow for non-technical users, and even for majority > > of what technical users do. > > Paul is right, but there anyway is a "halfway house", Arch Linux does > provide the Arch User Repository. I have used Linux since the Yggdrasil days where you had to compile everything from scratch. Now, even in the AUR compilation is a simple process requiring no technical knowledge on the part of the user to build and install packages, save a couple of commands to be entered in the shell. I don't miss those days at all. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From ralf.mardorf at rocketmail.com Mon Mar 17 13:28:06 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 17 Mar 2014 14:28:06 +0100 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> Message-ID: <1395062886.9232.195.camel@archlinux> On Mon, 2014-03-17 at 09:15 -0400, Joe Hartley wrote: > On Mon, 17 Mar 2014 14:07:45 +0100 > Ralf Mardorf wrote: > > > On Mon, 2014-03-17 at 08:46 -0400, Paul Davis wrote: > > > Installing Linux software from source has more or less vanished as > > > part of the work flow for non-technical users, and even for majority > > > of what technical users do. > > > > Paul is right, but there anyway is a "halfway house", Arch Linux does > > provide the Arch User Repository. > > I have used Linux since the Yggdrasil days where you had to compile > everything from scratch. Now, even in the AUR compilation is a simple > process requiring no technical knowledge on the part of the user to > build and install packages, save a couple of commands to be entered in > the shell. Hopefully it won't cause a flame-war ;). Hardcore: $ yaourt -S "package" When asked to edit the PKGBUILD in my experience, it's usually ok to reply with "no". Using FreeBSD and compiling from a FreeBSD port is much more complicated, OTOH, using yaourt for Arch Linux is disputed ;). From dlphillips at woh.rr.com Mon Mar 17 13:28:51 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 17 Mar 2014 09:28:51 -0400 Subject: [LAU] legacy equipment for a user who does not "program" In-Reply-To: <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> Message-ID: <5326F893.4030100@woh.rr.com> On 03/17/2014 09:15 AM, Joe Hartley wrote: > > I have used Linux since the Yggdrasil days where you had to compile > everything from scratch... > > I don't miss those days at all. > I don't miss them nor do I regret a single second I spent in that learning process. Best, dp From simonzwise at gmail.com Mon Mar 17 14:44:20 2014 From: simonzwise at gmail.com (Simon Wise) Date: Tue, 18 Mar 2014 01:44:20 +1100 Subject: [LAU] android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <5326F893.4030100@woh.rr.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> Message-ID: <53270A44.9050604@gmail.com> On 18/03/14 00:28, Dave Phillips wrote: > > On 03/17/2014 09:15 AM, Joe Hartley wrote: >> >> I have used Linux since the Yggdrasil days where you had to compile >> everything from scratch... >> >> I don't miss those days at all. >> > > I don't miss them nor do I regret a single second I spent in that learning process. Kind of related, but the other end of the GUI v DIY spectrum ... I'm only just now diving into Android ... this year's Samsung Note offering. It is proving a nice device, 4 cores with an additional processor for the pen and a very nice screen. But it's getting very frustrating looking at wizards with three options, none of which suit, then scrounging around a zillion half-baked apps available which may or may not do what I want. A bit like looking through a zillion offerings via apt-get for the first time, but without the quality control, man pages, web pages and community history and support that comes built in with debian. Everything is GUI. Everything is simplified. Very little is possible at any particular step, and the offerings are all context determined so the same path will lead to a different set of choices in slightly changed circumstances. By default, everything non-wizard-ish is hidden, mostly locked away by default, there is almost no documentation. It's all plug and play, or not. The basic things it does built in are done well, the interface is well thought out, flexible and effective, and with a bluetooth keyboard and my old Lifebook pen which is proper sized with an extra button (middle click) all dealt with properly and cleanly it is going to be very nice when I get it sorted. But discover-ability is not there at all. Plus anything out of the standard "be a good consumer" thing may require writing it yourself in Java, or struggling with someone else's undocumented offering which suited their particular needs and device. Any hints appreciated ... an app called juiceSSH has given me a command line locally, and ssh access to my other machines with a clean interface and good keyboard support. And the wacom pen works very well, with very nice built in support for handwriting recognition, maths formulas and such ... so the two main reasons I got it are working. I've got an Xserver installed, XSDL, which looks very promising and a debian chroot seems best but which method is the best? I've tested an app which has gimp and inkscape on an xfce desktop, it runs fine, the device copes easily, the (X display from XSDL seems better though, I will use it instead). It is apparently just a debian chroot so this path will be successful. Haven't yet added an admin account, first I am seeing what is possible without it. But the command line isn't much use, even man is in /sbin it seems. Hints from those who have been here already much appreciated! Simon From simonzwise at gmail.com Mon Mar 17 15:10:37 2014 From: simonzwise at gmail.com (Simon Wise) Date: Tue, 18 Mar 2014 02:10:37 +1100 Subject: [LAU] android for the linux user ... has jackd by default In-Reply-To: <53270A44.9050604@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> Message-ID: <5327106D.2080908@gmail.com> .... plus a very nice surprise ... it already has a jack server installed by default, accessible as a normal user with the usual CL jack-apps ... have not tested it yet, but with luck puredata will connect from a chroot, I've already had good experiences with the arm debian stuff including puredata and jack in Raspberry Pis so maybe it will be quite straightforward? Simon From lau at kudla.org Mon Mar 17 15:12:43 2014 From: lau at kudla.org (Rob) Date: Mon, 17 Mar 2014 11:12:43 -0400 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <53270A44.9050604@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> Message-ID: <532710EB.2040801@kudla.org> On 03/17/2014 10:44 AM, Simon Wise wrote: > Any hints appreciated ... an app called juiceSSH has given me a command > line locally, and ssh access to my other machines with a clean interface > and good keyboard support. I use Connectbot, which is FOSS and offers port forwarding without an in-app purchase, and VNC to control my machines remotely over ssh tunnels. I haven't tried JuiceSSH, but as soon as I see in-app purchases I get concerned about an app. I also have an SSH server installed. I used to use sshdroid, but at some point they got overzealous about detecting ad blockers (any change to your hosts file is treated as an ad blocker and the service will fail to start) so I've switched to SSHelper, which is FOSS. However, my current phone (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if there's an option to disable that I haven't been able to find it, and I can't get inbound connections to work with either. But for mounting stuff over sftp, it works reasonably well on wifi. Haven't tried XSDL yet because VNC (x11vnc on the machine being controlled) is so much faster than forwarding X clients in my experience, but I also haven't tried getting a debian chroot -- my old phone couldn't handle it, I only got my current phone a few months ago and haven't had time to root it yet or put on a different ROM, and I thought all the debian chroot methods required root. > Haven't yet added an admin account, first I am seeing what is possible > without it. But the command line isn't much use, even man is in /sbin it > seems. It is very frustrating that even ping requires root, and that devices we paid for require security exploits to even get root. The excuses even prominent Android bloggers give (can't damage your device if you don't have full control of it) are ridiculous, along the lines of advocating welded-shut car hoods just in case the user gets it in his head to open it up and put windshield washer fluid in the oil pan, though it wouldn't surprise me to hear that argument soon in the age of mandatory GPS tracker legislation. You can install bash without a full debian chroot, but it's still fairly limited without the other GNU software, and of course, you're still limited to doing whatever the phone allows you to. My phone is actually more powerful than the laptop I bought a month after I got it, so I've thought about Ubuntu for Android or something similar, something with a bit of effort put into making it usable without a bluetooth keyboard and mouse (I do have a keyboard case for my phone, but when I use VNC I always have to zoom in to operate menus and the like, and right- or middle-clicking is tedious) but the next thing I do will be to root it and possibly get a better ROM on there, if there is one without too much functionality missing. Android still strikes the best balance between functionality and freedom for me, but it's nowhere near as open as even Ubuntu, and Google has been making more and more pieces of it proprietary of late, letting the original FOSS components languish unchanged in the AOSP repositories. Rob From pshirkey at boosthardware.com Mon Mar 17 15:41:31 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Tue, 18 Mar 2014 02:41:31 +1100 (EST) Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532710EB.2040801@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> Message-ID: <60341.86.107.254.57.1395070891.squirrel@boosthardware.com> On Tue, March 18, 2014 2:12 am, Rob wrote: > On 03/17/2014 10:44 AM, Simon Wise wrote: >> Any hints appreciated ... an app called juiceSSH has given me a command >> line locally, and ssh access to my other machines with a clean interface >> and good keyboard support. > > I use Connectbot, which is FOSS and offers port forwarding without an > in-app purchase, and VNC to control my machines remotely over ssh tunnels. > I haven't tried JuiceSSH, but as soon as I see in-app purchases I get > concerned about an app. > > I also have an SSH server installed. I used to use sshdroid, but at some > point they got overzealous about detecting ad blockers (any change to your > hosts file is treated as an ad blocker and the service will fail to start) > so I've switched to SSHelper, which is FOSS. However, my current phone > (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if > there's > an option to disable that I haven't been able to find it, and I can't get > inbound connections to work with either. But for mounting stuff over sftp, > it works reasonably well on wifi. > > Haven't tried XSDL yet because VNC (x11vnc on the machine being > controlled) > is so much faster than forwarding X clients in my experience, but I also > haven't tried getting a debian chroot -- my old phone couldn't handle it, > I > only got my current phone a few months ago and haven't had time to root it > yet or put on a different ROM, and I thought all the debian chroot methods > required root. > >> Haven't yet added an admin account, first I am seeing what is possible >> without it. But the command line isn't much use, even man is in /sbin it >> seems. > > It is very frustrating that even ping requires root, and that devices we > paid for require security exploits to even get root. The excuses even > prominent Android bloggers give (can't damage your device if you don't > have > full control of it) are ridiculous, along the lines of advocating > welded-shut car hoods just in case the user gets it in his head to open it > up and put windshield washer fluid in the oil pan, though it wouldn't > surprise me to hear that argument soon in the age of mandatory GPS tracker > legislation. > > You can install bash without a full debian chroot, but it's still fairly > limited without the other GNU software, and of course, you're still > limited > to doing whatever the phone allows you to. > > My phone is actually more powerful than the laptop I bought a month after > I > got it, so I've thought about Ubuntu for Android or something similar, > something with a bit of effort put into making it usable without a > bluetooth keyboard and mouse (I do have a keyboard case for my phone, but > when I use VNC I always have to zoom in to operate menus and the like, and > right- or middle-clicking is tedious) but the next thing I do will be to > root it and possibly get a better ROM on there, if there is one without > too > much functionality missing. > > Android still strikes the best balance between functionality and freedom > for me, but it's nowhere near as open as even Ubuntu, and Google has been > making more and more pieces of it proprietary of late, letting the > original > FOSS components languish unchanged in the AOSP repositories. > What you'll find is that after a while the locked down filesystem which usually requires a proprietary windows only application to make updates or complex contortion to interact with in Linux is a real pita. Don't even think about using external hardware that requires a custom driver to be installed. When forced to work with Android devices I prefer to use adb to interact with them. ./adb shell You can also use adb wirelessly if your device supports the wireless flag which is pretty handy. setprop service.adb.tcp.port 5555 & stop adbd & start adbd Depending on your distro you can get access to adb as a package or if that is not possible for you might have to install it via the android sdk. Then you can use your desktop to work at the system level while still retaining access to the touch interface for other things... -- Patrick Shirkey Boost Hardware Ltd From federicogalland at gmail.com Mon Mar 17 15:54:00 2014 From: federicogalland at gmail.com (Federico Galland) Date: Mon, 17 Mar 2014 12:54:00 -0300 Subject: [LAU] android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <53270A44.9050604@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> Message-ID: <20140317125400.d8d0beaa7c1b51f2af677ddc@gmail.com> On Tue, 18 Mar 2014 01:44:20 +1100 Simon Wise wrote: > On 18/03/14 00:28, Dave Phillips wrote: > > > > On 03/17/2014 09:15 AM, Joe Hartley wrote: > >> > >> I have used Linux since the Yggdrasil days where you had to compile > >> everything from scratch... > >> > >> I don't miss those days at all. > >> > > > > I don't miss them nor do I regret a single second I spent in that learning process. > > Kind of related, but the other end of the GUI v DIY spectrum ... > > I'm only just now diving into Android ... this year's Samsung Note offering. It > is proving a nice device, 4 cores with an additional processor for the pen and a > very nice screen. But it's getting very frustrating looking at wizards with > three options, none of which suit, then scrounging around a zillion half-baked > apps available which may or may not do what I want. A bit like looking through a > zillion offerings via apt-get for the first time, but without the quality > control, man pages, web pages and community history and support that comes built > in with debian. > > Everything is GUI. Everything is simplified. Very little is possible at any > particular step, and the offerings are all context determined so the same path > will lead to a different set of choices in slightly changed circumstances. By > default, everything non-wizard-ish is hidden, mostly locked away by default, > there is almost no documentation. It's all plug and play, or not. > > The basic things it does built in are done well, the interface is well thought > out, flexible and effective, and with a bluetooth keyboard and my old Lifebook > pen which is proper sized with an extra button (middle click) all dealt with > properly and cleanly it is going to be very nice when I get it sorted. But > discover-ability is not there at all. Plus anything out of the standard "be a > good consumer" thing may require writing it yourself in Java, or struggling with > someone else's undocumented offering which suited their particular needs and device. > > Any hints appreciated ... an app called juiceSSH has given me a command line > locally, and ssh access to my other machines with a clean interface and good > keyboard support. And the wacom pen works very well, with very nice built in > support for handwriting recognition, maths formulas and such ... so the two main > reasons I got it are working. > > I've got an Xserver installed, XSDL, which looks very promising and a debian > chroot seems best but which method is the best? I've tested an app which has > gimp and inkscape on an xfce desktop, it runs fine, the device copes easily, the > (X display from XSDL seems better though, I will use it instead). It is > apparently just a debian chroot so this path will be successful. > > Haven't yet added an admin account, first I am seeing what is possible without > it. But the command line isn't much use, even man is in /sbin it seems. > > Hints from those who have been here already much appreciated! > > > Simon > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I had a smartphone which was literally eaten by my dog, but before that, I used to download apps from the F-droid repos, which are mostly FOSS. IMHO Android sucks, but the apps in those repos at least don't bother you with ads and nonsense. To be honest, I'm happy I don't have a cellphone anymore... Cheers! From louigi.verona at gmail.com Mon Mar 17 17:29:46 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Mon, 17 Mar 2014 21:29:46 +0400 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <5326C7B1.1070606@gmail.com> References: <53262DEB.5080207@gmail.com> <5326C7B1.1070606@gmail.com> Message-ID: Lo-fi competition is a great idea, imho. On Mon, Mar 17, 2014 at 2:00 PM, Lorenzo Sutton wrote: > On 17/03/2014 03:20, Paul Davis wrote: > >> >> >> >> On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton >> > wrote: >> >> In light of the interesting discussion on sample rates I propose a >> music competition among LAU around production of music pieces with >> quality considered 'low' by current dominating >> professional/audiophile standards in the digital domain: >> >> Specifics to be discussed, but I would start with the following: >> >> >> can we just re-record the beatles or miles davis from vinyl and consider >> it done? >> >> > You got he point - well said :-) > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From guido-scholz at gmx.net Mon Mar 17 19:37:20 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Mon, 17 Mar 2014 20:37:20 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> Message-ID: <20140317193720.GA3858@traun.gscholz.bayernline.de> Am Mon, 17. Mar 2014 um 00:51:31 +0400 schrieb Alexandre Prokoudine: > Hi Guido, Hi Alexandre, > It crashes after failing to load LADSPA plugins. > > http://pastebin.com/kvJk2f3K hm, very unpleasant. Can you send me your patch file? Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From guido-scholz at gmx.net Mon Mar 17 19:47:34 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Mon, 17 Mar 2014 20:47:34 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> <20140316172531.GA4250@traun.gscholz.bayernline.de> Message-ID: <20140317194734.GB3858@traun.gscholz.bayernline.de> Am Sun, 16. Mar 2014 um 19:37:28 +0100 schrieb rosea grammostola: Hi rosea, > Huh no NSM support? yes such things happen usually if no-one sends a patch upstream. Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From alexandre.prokoudine at gmail.com Mon Mar 17 19:50:31 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 17 Mar 2014 23:50:31 +0400 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140317193720.GA3858@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> Message-ID: On Mon, Mar 17, 2014 at 11:37 PM, Guido Scholz wrote: >> It crashes after failing to load LADSPA plugins. >> >> http://pastebin.com/kvJk2f3K > > hm, very unpleasant. Can you send me your patch file? I don't need to :) It's example_ams_demo_scope_spectrum.ams. Granted, I'm probably expected to have those plugins installed, but lack thereof shouldn't crash the app :) Alexandre From guido-scholz at gmx.net Mon Mar 17 21:23:15 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Mon, 17 Mar 2014 22:23:15 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> Message-ID: <20140317212315.GA4090@traun.gscholz.bayernline.de> Am Mon, 17. Mar 2014 um 23:50:31 +0400 schrieb Alexandre Prokoudine: Hi Alexandre, > > hm, very unpleasant. Can you send me your patch file? > > I don't need to :) It's example_ams_demo_scope_spectrum.ams. > > Granted, I'm probably expected to have those plugins installed, but > lack thereof shouldn't crash the app :) I am with you, but the error is not about missing plugins, you would get a dialog box showing an appropriate message. There are strange errors while reading the patch file on your system. The same patch runs fine here (L. Mint 16, Petra, aka Ubuntu saucy) without a crash. Which system are you running? Some of the old example files have errors because there are more data to read than the old ams versions saved. I will update them for the next release. Can you check what happens with the attached file (same patch but saved with the current ams version)? Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- Module 7 1 1019 238 0 0 ColorP 0 0 1 16 250 200 50 180 180 180 ColorP 1 1 1 16 250 200 50 180 180 180 FSlider 1 0 8192 0 0 16384 1 FSlider 1 1 8192 0 0 16384 1 FSlider 1 2 8192 0 0 16384 1 ComboBox 1 0 1 1 ConfigDialog 0 0 0 100 30 Module 11 3 318 33 0 0 ColorP 0 1 3 7 250 200 50 180 180 180 ColorP 1 1 3 7 250 200 50 180 180 180 FSlider 3 0 0 0 0 16384 1 FSlider 3 1 -106718 1 -113176 0 1 FSlider 3 2 0 0 0 16384 1 FSlider 3 3 3887 0 0 16384 1 FSlider 3 4 9742 0 0 16384 1 FSlider 3 5 6298 0 0 16384 1 FSlider 3 6 12354 0 1638 163840 1 ConfigDialog 0 0 0 100 30 Module 2 4 612 130 0 0 ColorP 0 0 4 3 250 200 50 180 180 180 ColorP 2 0 4 10 250 200 50 180 180 180 FSlider 4 0 7042 0 0 163840 1 FSlider 4 1 0 0 0 163840 1 FSlider 4 2 8192 0 0 16384 1 FSlider 4 3 8192 0 0 16384 1 FSlider 4 4 8192 0 0 16384 1 ConfigDialog 0 0 0 100 30 Module 1 5 386 177 0 0 ColorP 1 0 5 7 250 200 50 180 180 180 FSlider 5 0 0 0 0 16384 1 FSlider 5 1 1638 0 0 163840 1 FSlider 5 2 0 0 0 163840 1 FSlider 5 3 8192 0 1638 14745 1 FSlider 5 4 0 0 0 16384 1 FSlider 5 5 0 0 0 102940 1 ISlider 5 0 3 1 ISlider 5 1 1 1 ISlider 5 2 1 1 ComboBox 5 0 0 1 ConfigDialog 0 0 0 100 30 Module 16 7 184 104 0 0 ColorP 0 1 7 12 250 200 50 180 180 180 ColorP 1 1 7 21 250 200 50 180 180 180 ComboBox 7 0 10 1 ConfigDialog 0 0 0 100 30 Module 9 10 498 228 0 0 ColorP 0 2 10 5 250 200 50 180 180 180 ColorP 1 0 10 13 250 200 50 180 180 180 ColorP 2 0 10 18 250 200 50 180 180 180 FSlider 10 0 43297 0 0 163840 1 FSlider 10 1 111687 0 0 163840 1 FSlider 10 2 48709 0 0 163840 1 FSlider 10 3 0 0 0 163840 1 FSlider 10 4 13107 0 163 16384 1 FSlider 10 5 0 0 0 16384 1 ComboBox 10 0 6 1 ConfigDialog 0 0 0 100 30 Module 14 12 14 148 0 0 FSlider 12 0 27228 0 0 163840 1 FSlider 12 1 6572 0 0 16384 1 ConfigDialog 0 0 0 100 30 Module 15 13 278 272 0 0 ColorP 0 0 13 7 250 200 50 180 180 180 FSlider 13 0 8192 0 0 163840 1 FSlider 13 1 8192 0 0 163840 1 ConfigDialog 0 0 0 100 30 Module 6 15 715 206 0 cmt canyon_delay ColorP 0 0 15 4 250 200 50 180 180 180 ColorP 1 0 15 4 250 200 50 180 180 180 FSlider 15 0 9273 0 163 16220 1 FSlider 15 1 12347 0 -16384 16384 1 FSlider 15 2 6788 0 163 16220 1 FSlider 15 3 13661 0 -16384 16384 1 FSlider 15 4 81920000 0 16384 81920000 1 ConfigDialog 0 0 0 100 30 Module 6 16 868 232 0 cmt freeverb3 ColorP 0 0 16 15 250 200 50 180 180 180 ColorP 1 1 16 15 250 200 50 180 180 180 FSlider 16 0 8215 0 0 16384 1 FSlider 16 1 -83988 1 -113176 0 1 FSlider 16 2 -9404 1 -113176 0 1 FSlider 16 3 0 1 -113176 0 1 FSlider 16 4 16384 0 0 16384 1 CheckBox 16 0 0 1 ConfigDialog 0 0 0 100 30 Module 15 18 381 411 0 0 ColorP 0 2 18 19 250 200 50 180 180 180 FSlider 18 0 98116 0 0 163840 1 FSlider 18 1 70418 0 0 163840 1 ConfigDialog 0 0 0 100 30 Module 14 19 240 422 0 0 FSlider 19 0 133325 0 0 163840 1 FSlider 19 1 16384 0 0 16384 1 ConfigDialog 0 0 0 100 30 Module 16 21 71 302 0 0 ColorP 0 2 21 12 250 200 50 180 180 180 ComboBox 21 0 0 1 ConfigDialog 0 0 0 100 30 Module 24 23 217 4 0 0 ColorP 0 0 23 4 250 200 50 180 180 180 FSlider 23 0 418607 0 163840 16384000 1 FSlider 23 1 49195 0 1638 163840 1 FSlider 23 2 -686 0 -16384 16384 1 ComboBox 23 0 1 1 ComboBox 23 1 0 1 ConfigDialog 0 4 24 274 290 Module 25 24 113 4 0 0 ColorP 0 0 24 4 250 200 50 180 180 180 ConfigDialog 0 0 0 100 30 Module 25 25 9 4 0 0 ColorP 0 0 25 4 250 200 50 180 180 180 ConfigDialog 0 0 0 100 30 Comment 0 0 640 360 315 174 #PARA# 0 0 0 This patch has been contributed by Bill Allen. #ARAP# #PARA# 0 0 1 #ARAP# #PARA# 0 0 2 Here, a scope and two spectrum modules have been added. Just right-click on the modules to open the scope and spectrum views. #ARAP# -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From guido-scholz at gmx.net Mon Mar 17 22:19:31 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Mon, 17 Mar 2014 23:19:31 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> Message-ID: <20140317221931.GA4205@traun.gscholz.bayernline.de> Am Mon, 17. Mar 2014 um 23:50:31 +0400 schrieb Alexandre Prokoudine: Hi Alexandre, > Granted, I'm probably expected to have those plugins installed, but > lack thereof shouldn't crash the app :) OK, I found an other patch file crashing here. I will see what I can do (but not today ;). Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From brouits at free.fr Mon Mar 17 22:27:30 2014 From: brouits at free.fr (=?ISO-8859-1?Q?Beno=EEt_Rouits?=) Date: Mon, 17 Mar 2014 23:27:30 +0100 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: References: <53262DEB.5080207@gmail.com> <5326C7B1.1070606@gmail.com> Message-ID: <532776D2.4060909@free.fr> Le 17/03/2014 18:29, Louigi Verona a ?crit : > Lo-fi competition is a great idea, imho. > > > On Mon, Mar 17, 2014 at 2:00 PM, Lorenzo Sutton > > wrote: > > On 17/03/2014 03:20, Paul Davis wrote: > > > > > On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton > > >> wrote: > > In light of the interesting discussion on sample rates I > propose a > music competition among LAU around production of music > pieces with > quality considered 'low' by current dominating > professional/audiophile standards in the digital domain: > > Specifics to be discussed, but I would start with the > following: > > > can we just re-record the beatles or miles davis from vinyl and > consider > it done? > > > You got he point - well said :-) > i have somewhere (if i can find it) some field recording taken with a mono, 8bit, 16KHz rate from a so-called dictaphone, overdubbed with a stereo piano track at 16bit/44.1Khz.. the effect is not so bad. Well, to say, lo-fi is interseting to me. this proposition is a good idea, and sometime big restrictions can lead to cool productions. - Ben From aiyumi.br at gmail.com Tue Mar 18 01:14:41 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Mon, 17 Mar 2014 22:14:41 -0300 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: <5326B43F.3090101@ladisch.de> References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> Message-ID: 2014-03-17 5:37 GMT-03:00, Clemens Ladisch : > Aiyumi Moriya wrote: >> 2014-03-16 17:58 GMT-03:00, Clemens Ladisch : >>> Aiyumi Moriya wrote: >>>> MIDI recording and playback work fine, but now I want to know if it >>>> can be made to work as a soundcard. I know that the Motif XF can work >>>> as a USB audio interface on Windows and Mac. >>> >>> The lsusb output shows that there is no interface for this. >>> Does this need some kind of configuration on the device? >> >> Yes, it does. > > And what is the lsusb output after you have enabled this? It was already enabled. If I disable it, the MIDI ports still appear and the device still shows on "/proc/asound/cards", but MIDI playback and recording don't work. It's an option to tell the device to communicate via MIDI cables or USB (or firewire, which I don't have). If set to USB, it should accept MIDI via USB, and on Win/Mac it can also be used as audio interface. I checked Win/Mac tutorials about using it as a soundcard, and setting this option was the only relevant step mentioned, apart from installing the proprietary driver of course. So... Dead end? I thought it had a chance of working since I saw on that thread that MOX (similar product line) works. I got my hopes up when the MIDI communication worked here (it didn't work before. On my previous system, the ports didn't even show), then I saw my device on "/proc/asound/cards"... But now that I think about it, Virmidi is there too, so being listed there probably doesn't mean anything :(. -- ____________________ Blog: http://aiyumi.warpstar.net/ From simonzwise at gmail.com Tue Mar 18 01:48:10 2014 From: simonzwise at gmail.com (Simon Wise) Date: Tue, 18 Mar 2014 12:48:10 +1100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <60341.86.107.254.57.1395070891.squirrel@boosthardware.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <60341.86.107.254.57.1395070891.squirrel@boosthardware.com> Message-ID: <5327A5DA.3070609@gmail.com> On 18/03/14 02:41, Patrick Shirkey wrote: > > On Tue, March 18, 2014 2:12 am, Rob wrote: >> On 03/17/2014 10:44 AM, Simon Wise wrote: >>> Any hints appreciated ... an app called juiceSSH has given me a command >>> line locally, and ssh access to my other machines with a clean interface >>> and good keyboard support. >> >> I use Connectbot, which is FOSS and offers port forwarding without an >> in-app purchase, and VNC to control my machines remotely over ssh tunnels. >> I haven't tried JuiceSSH, but as soon as I see in-app purchases I get >> concerned about an app. >> >> I also have an SSH server installed. I used to use sshdroid, but at some >> point they got overzealous about detecting ad blockers (any change to your >> hosts file is treated as an ad blocker and the service will fail to start) >> so I've switched to SSHelper, which is FOSS. However, my current phone >> (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if >> there's >> an option to disable that I haven't been able to find it, and I can't get >> inbound connections to work with either. But for mounting stuff over sftp, >> it works reasonably well on wifi. >> >> Haven't tried XSDL yet because VNC (x11vnc on the machine being >> controlled) >> is so much faster than forwarding X clients in my experience, but I also >> haven't tried getting a debian chroot -- my old phone couldn't handle it, >> I >> only got my current phone a few months ago and haven't had time to root it >> yet or put on a different ROM, and I thought all the debian chroot methods >> required root. >> >>> Haven't yet added an admin account, first I am seeing what is possible >>> without it. But the command line isn't much use, even man is in /sbin it >>> seems. >> >> It is very frustrating that even ping requires root, and that devices we >> paid for require security exploits to even get root. The excuses even >> prominent Android bloggers give (can't damage your device if you don't >> have >> full control of it) are ridiculous, along the lines of advocating >> welded-shut car hoods just in case the user gets it in his head to open it >> up and put windshield washer fluid in the oil pan, though it wouldn't >> surprise me to hear that argument soon in the age of mandatory GPS tracker >> legislation. >> >> You can install bash without a full debian chroot, but it's still fairly >> limited without the other GNU software, and of course, you're still >> limited >> to doing whatever the phone allows you to. >> >> My phone is actually more powerful than the laptop I bought a month after >> I >> got it, so I've thought about Ubuntu for Android or something similar, >> something with a bit of effort put into making it usable without a >> bluetooth keyboard and mouse (I do have a keyboard case for my phone, but >> when I use VNC I always have to zoom in to operate menus and the like, and >> right- or middle-clicking is tedious) but the next thing I do will be to >> root it and possibly get a better ROM on there, if there is one without >> too >> much functionality missing. >> >> Android still strikes the best balance between functionality and freedom >> for me, but it's nowhere near as open as even Ubuntu, and Google has been >> making more and more pieces of it proprietary of late, letting the >> original >> FOSS components languish unchanged in the AOSP repositories. >> > > What you'll find is that after a while the locked down filesystem which > usually requires a proprietary windows only application to make updates or > complex contortion to interact with in Linux is a real pita. Don't even > think about using external hardware that requires a custom driver to be > installed. > > When forced to work with Android devices I prefer to use adb to interact > with them. > > ./adb shell > > You can also use adb wirelessly if your device supports the wireless flag > which is pretty handy. > > setprop service.adb.tcp.port 5555& stop adbd& start adbd > > Depending on your distro you can get access to adb as a package or if that > is not possible for you might have to install it via the android sdk. > > Then you can use your desktop to work at the system level while still > retaining access to the touch interface for other things... ok, adb was next on the list to try ... I'm treating the android somewhat as a remote extension of my desktop, and I'm quite happy doing any setup or installations via the desktop, I'll want to keep the whole samsung/android set of drives and interface, that seems to be done quite nicely, but also want to explore the development side for my own potential uses, so adb is a must. Patrick .. you're in Sydney I think? I've just moved back here after doing a degree in Perth ... I'm based in Ultimo, want to catch up sometime? Simon From simonzwise at gmail.com Tue Mar 18 03:32:56 2014 From: simonzwise at gmail.com (Simon Wise) Date: Tue, 18 Mar 2014 14:32:56 +1100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532710EB.2040801@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> Message-ID: <5327BE68.4040302@gmail.com> On 18/03/14 02:12, Rob wrote: > On 03/17/2014 10:44 AM, Simon Wise wrote: >> Any hints appreciated ... an app called juiceSSH has given me a command >> line locally, and ssh access to my other machines with a clean interface >> and good keyboard support. > > I use Connectbot, which is FOSS and offers port forwarding without an > in-app purchase, and VNC to control my machines remotely over ssh tunnels. > I haven't tried JuiceSSH, but as soon as I see in-app purchases I get > concerned about an app. I certainly prefer FLOSS, but obviously (since I just chose to jump into Samsung/Google land) I'm not strict about it, I'll try Connectbot. I'll choose mainly based on how well the keyboard input is handled, and how it behaves as a terminal emulator. I'll probably end up using bash, ssh etc in XSDL and a debian chroot, which would be a FOSS solution. > > I also have an SSH server installed. I used to use sshdroid, but at some > point they got overzealous about detecting ad blockers (any change to your > hosts file is treated as an ad blocker and the service will fail to start) > so I've switched to SSHelper, which is FOSS. However, my current phone > (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if there's > an option to disable that I haven't been able to find it, and I can't get > inbound connections to work with either. But for mounting stuff over sftp, > it works reasonably well on wifi. that's pretty much all I'd want an ssh server to do, this is more a remote interface to other computers, I'm more interested in midi, osc, puredata etc being able to connect via UDP and TCP in an isolated local network, preferably via ethernet cable, as far as linux audio uses are concerned. And testing USB audio, potentially using that. > > Haven't tried XSDL yet because VNC (x11vnc on the machine being controlled) > is so much faster than forwarding X clients in my experience, but I also > haven't tried getting a debian chroot -- my old phone couldn't handle it, I > only got my current phone a few months ago and haven't had time to root it > yet or put on a different ROM, and I thought all the debian chroot methods > required root. A terminal for text and compiling is mostly what I want remotely, a gvim window over ssh -X would be a bit nicer if it wasn't laggy, and I'd much prefer to see the individual windows than a whole desktop. I have been working on Pis a lot, running without any X server locally, forwarding any GUI windows for editing puredata patches etc ... in this case X forwarding is certainly much quicker and smoother than VNC with a server on the Pi. Likewise I would rather interact with apps in a chroot this way. In any case whatever is running the window needs to have a good input driver for the touch/pen and use the android graphics well .. I'd guess the android VNC apps would do that, I believe XSDL is essentially an android-native implementation of the X server with an input driver for the screen, so a lot would depend on how good its implementation is. I'll find out more as I go. The chroots certainly don't need root access, an app can set them up ... I assume that I couldn't set one up manually from the android CL without admin access, but that I could do so using adb from a desktop (while leaving the local system without an admin user). That's what I'll try first, it seems a good way of working. > >> Haven't yet added an admin account, first I am seeing what is possible >> without it. But the command line isn't much use, even man is in /sbin it >> seems. > > It is very frustrating that even ping requires root, and that devices we > paid for require security exploits to even get root. The excuses even > prominent Android bloggers give (can't damage your device if you don't have > full control of it) are ridiculous, along the lines of advocating > welded-shut car hoods just in case the user gets it in his head to open it > up and put windshield washer fluid in the oil pan, though it wouldn't > surprise me to hear that argument soon in the age of mandatory GPS tracker > legislation. > > You can install bash without a full debian chroot, but it's still fairly > limited without the other GNU software, and of course, you're still limited > to doing whatever the phone allows you to. If all it takes is adding a root password and a few files then delivering without a root password seems fine to me, even sensible (I haven't looked into it yet) ... in that case the device is not locked, rather the ordinary user is very limited and admin access is off by default. Plus the device can be treated as an embedded device, and administered remotely using adp. Certainly I wouldn't expect much help from Samsung or Google if I choose to take the system in my own direction, they have made a default setup and they claim it works well for what it is sold as, which seems true so far. This is quite different from the hardware locking that the Windows machines have, especially the ARM ones, where the OS will not run anything unless the file is explicitly signed by Microsoft, and the ARM hardware will never boot anything except that OS (the Intel hardware can be unlocked). > > My phone is actually more powerful than the laptop I bought a month after I > got it, so I've thought about Ubuntu for Android or something similar, > something with a bit of effort put into making it usable without a > bluetooth keyboard and mouse (I do have a keyboard case for my phone, but > when I use VNC I always have to zoom in to operate menus and the like, and > right- or middle-clicking is tedious) but the next thing I do will be to > root it and possibly get a better ROM on there, if there is one without too > much functionality missing. Screenshots of XSDL suggest they have made that effort re touch access to the pointer in X apps, like having a small area around the finger zoomed in and two-finger tap for right click. I like Samsung's android pen interface, especially since I have my old Fujitsu pen with two buttons (i.e. right and middle click), an eraser end and is a proper pen size. Their handwriting input is actually quite nice, and quick, for entering text in the various supported languages. > > Android still strikes the best balance between functionality and freedom > for me, but it's nowhere near as open as even Ubuntu, and Google has been > making more and more pieces of it proprietary of late, letting the original > FOSS components languish unchanged in the AOSP repositories. Re the kernel didn't they move back to adding to the mainline kernel rather than maintaining a whole separate fork? Otherwise the setup seems to be plugin/module based ... they have defined an interface for apps, that is the way offered to build your own stuff, their own interface may well become quite closed source, like Apple's one is ... I treat the hardware+drivers+interface as something I buy as is, and refuse to buy if I am locked out of using significant parts of it without approval from above. I keep noticing they are using unix tools ... for jackd, wireless tools, dhcp and hotspot tools, plus /dev, /sys, /proc seem quite populated (but I haven't tested any of it yet). It seems these are more populated than were earlier versions, given remarks in older threads, so it seems more of the standard *nix drivers are being used, the ones that run with linux, so I'd hope their licenses are being respected. Many are BSD licensed of course, so these can be adapted closed source if Google is prepared to maintain its own fork, but working with a linux kernel certainly pushes Google to be more open, Apple chose a BSD licensed base and is much more free to make its additions closed, and indeed pushes that distinction quite hard. Simon From pshirkey at boosthardware.com Tue Mar 18 06:35:35 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Tue, 18 Mar 2014 17:35:35 +1100 (EST) Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <5327A5DA.3070609@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <60341.86.107.254.57.1395070891.squirrel@boosthardware.com> <5327A5DA.3070609@gmail.com> Message-ID: <62329.86.107.254.57.1395124535.squirrel@boosthardware.com> On Tue, March 18, 2014 12:48 pm, Simon Wise wrote: > On 18/03/14 02:41, Patrick Shirkey wrote: >> >> On Tue, March 18, 2014 2:12 am, Rob wrote: >>> On 03/17/2014 10:44 AM, Simon Wise wrote: >>>> Any hints appreciated ... an app called juiceSSH has given me a >>>> command >>>> line locally, and ssh access to my other machines with a clean >>>> interface >>>> and good keyboard support. >>> >>> I use Connectbot, which is FOSS and offers port forwarding without an >>> in-app purchase, and VNC to control my machines remotely over ssh >>> tunnels. >>> I haven't tried JuiceSSH, but as soon as I see in-app purchases I get >>> concerned about an app. >>> >>> I also have an SSH server installed. I used to use sshdroid, but at >>> some >>> point they got overzealous about detecting ad blockers (any change to >>> your >>> hosts file is treated as an ad blocker and the service will fail to >>> start) >>> so I've switched to SSHelper, which is FOSS. However, my current phone >>> (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if >>> there's >>> an option to disable that I haven't been able to find it, and I can't >>> get >>> inbound connections to work with either. But for mounting stuff over >>> sftp, >>> it works reasonably well on wifi. >>> >>> Haven't tried XSDL yet because VNC (x11vnc on the machine being >>> controlled) >>> is so much faster than forwarding X clients in my experience, but I >>> also >>> haven't tried getting a debian chroot -- my old phone couldn't handle >>> it, >>> I >>> only got my current phone a few months ago and haven't had time to root >>> it >>> yet or put on a different ROM, and I thought all the debian chroot >>> methods >>> required root. >>> >>>> Haven't yet added an admin account, first I am seeing what is possible >>>> without it. But the command line isn't much use, even man is in /sbin >>>> it >>>> seems. >>> >>> It is very frustrating that even ping requires root, and that devices >>> we >>> paid for require security exploits to even get root. The excuses even >>> prominent Android bloggers give (can't damage your device if you don't >>> have >>> full control of it) are ridiculous, along the lines of advocating >>> welded-shut car hoods just in case the user gets it in his head to open >>> it >>> up and put windshield washer fluid in the oil pan, though it wouldn't >>> surprise me to hear that argument soon in the age of mandatory GPS >>> tracker >>> legislation. >>> >>> You can install bash without a full debian chroot, but it's still >>> fairly >>> limited without the other GNU software, and of course, you're still >>> limited >>> to doing whatever the phone allows you to. >>> >>> My phone is actually more powerful than the laptop I bought a month >>> after >>> I >>> got it, so I've thought about Ubuntu for Android or something similar, >>> something with a bit of effort put into making it usable without a >>> bluetooth keyboard and mouse (I do have a keyboard case for my phone, >>> but >>> when I use VNC I always have to zoom in to operate menus and the like, >>> and >>> right- or middle-clicking is tedious) but the next thing I do will be >>> to >>> root it and possibly get a better ROM on there, if there is one without >>> too >>> much functionality missing. >>> >>> Android still strikes the best balance between functionality and >>> freedom >>> for me, but it's nowhere near as open as even Ubuntu, and Google has >>> been >>> making more and more pieces of it proprietary of late, letting the >>> original >>> FOSS components languish unchanged in the AOSP repositories. >>> >> >> What you'll find is that after a while the locked down filesystem which >> usually requires a proprietary windows only application to make updates >> or >> complex contortion to interact with in Linux is a real pita. Don't even >> think about using external hardware that requires a custom driver to be >> installed. >> >> When forced to work with Android devices I prefer to use adb to interact >> with them. >> >> ./adb shell >> >> You can also use adb wirelessly if your device supports the wireless >> flag >> which is pretty handy. >> >> setprop service.adb.tcp.port 5555& stop adbd& start adbd >> >> Depending on your distro you can get access to adb as a package or if >> that >> is not possible for you might have to install it via the android sdk. >> >> Then you can use your desktop to work at the system level while still >> retaining access to the touch interface for other things... > > ok, adb was next on the list to try ... I'm treating the android somewhat > as a > remote extension of my desktop, and I'm quite happy doing any setup or > installations via the desktop, I'll want to keep the whole samsung/android > set > of drives and interface, that seems to be done quite nicely, but also want > to > explore the development side for my own potential uses, so adb is a must. > > Patrick .. you're in Sydney I think? I've just moved back here after doing > a > degree in Perth ... I'm based in Ultimo, want to catch up sometime? > I would love to but I'm not in Sydney at the moment ;-) -- Patrick Shirkey Boost Hardware Ltd From thomas at residuum.org Tue Mar 18 09:45:49 2014 From: thomas at residuum.org (Thomas Mayer) Date: Tue, 18 Mar 2014 10:45:49 +0100 (CET) Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <53262DEB.5080207@gmail.com> References: <53262DEB.5080207@gmail.com> Message-ID: <20140318094549.A2EEA3260C7F@dd28920.kasserver.com> Hi, Lorenzo Sutton schrieb am 17.03.2014 00:04: > In light of the interesting discussion on sample rates I propose a music > competition among LAU around production of music pieces with quality > considered 'low' by current dominating professional/audiophile standards > in the digital domain: > > Specifics to be discussed, but I would start with the following: > > 1. Final piece shall be delivered with a maximum sampling rate of 32kHz. > Lower sample rates .allowed. There is a netlabel called 20kbps, that already does something similar without holding any competition: http://20kbps.sofapause.ch/index2.htm All the best, Thomas From clemens at ladisch.de Tue Mar 18 11:21:24 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Tue, 18 Mar 2014 12:21:24 +0100 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> Message-ID: <53282C34.5010203@ladisch.de> Aiyumi Moriya wrote: > MIDI recording and playback work fine, but now I want to know if it > can be made to work as a soundcard. I know that the Motif XF can work > as a USB audio interface on Windows and Mac. The manual says: | The USB connection between the instrument and the computer | can only be used for transfer of MIDI data. No audio data | can be transferred via USB. Regards, Clemens From lau at kudla.org Tue Mar 18 11:56:39 2014 From: lau at kudla.org (Rob) Date: Tue, 18 Mar 2014 07:56:39 -0400 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <5327BE68.4040302@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <5327BE68.4040302@gmail.com> Message-ID: <53283477.8050006@kudla.org> On 03/17/2014 11:32 PM, Simon Wise wrote: > I'm more interested in midi, osc, puredata > etc being able to connect via UDP and TCP in an isolated local network, > preferably via ethernet cable, as far as linux audio uses are concerned. > And testing USB audio, potentially using that. I hope that you'll share your results with us, as I (at least) am very interested in hearing about it and maybe trying to attempt it. My Android music-making so far has been a few rudimentary trackers and similar programs, and getting something "real" on there would be preferable until the gap has been filled. Rob From aiyumi.br at gmail.com Tue Mar 18 12:10:51 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Tue, 18 Mar 2014 09:10:51 -0300 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: <53282C34.5010203@ladisch.de> References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> <53282C34.5010203@ladisch.de> Message-ID: 2014-03-18 8:21 GMT-03:00, Clemens Ladisch : > The manual says: > | The USB connection between the instrument and the computer > | can only be used for transfer of MIDI data. No audio data > | can be transferred via USB. Oops. I feel really stupid now. I read the manual but totally missed this part. I checked it again and it also says: "When the optional FW16E (FireWire expansion board) is installed, the MOTIF XF can be connected to the computer via an IEEE1394 cable. Note that the audio data as well as MIDI data can be transmitted through an IEEE1394 cable by installing the Yamaha Steinberg FW Driver to the computer." So the audio interface function is Firewire only... I don't know what made me believe that USB also worked. But it also means that everything is working perfectly fine! Do you think Firewire has any chance of working? I had no interest to pursue it until now, and have no idea of the state of Firewire support on Linux. Thank you very much for the help, and sorry for the bother and this dumb mistake. -- ____________________ Blog: http://aiyumi.warpstar.net/ From harryhaaren at gmail.com Tue Mar 18 12:26:15 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 18 Mar 2014 12:26:15 +0000 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> <53282C34.5010203@ladisch.de> Message-ID: On Tue, Mar 18, 2014 at 12:10 PM, Aiyumi Moriya wrote: > Do you think Firewire has any chance of working? I had no interest to > pursue it until now, and have no idea of the state of Firewire support > on Linux. > Linux firewire drivers project: http://www.ffado.org/ The Motif XF isn't listed: http://www.ffado.org/?q=devicesupport%2Flist&filter0=yamaha&filter1=&op2=OR A quick google returns this thread, probably of interest: http://www.motifator.com/index.php/forum/viewthread/442794/#456446 Otherwise, just try it: and see if it works. That's the easy way to find out. This seems a related project: sponsored by Yamaha initially it seems.. http://mlanalsa.sourceforge.net/ I have no experience with this particular type of streaming driver / device, so I can't help much more :) -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From simonzwise at gmail.com Tue Mar 18 14:07:19 2014 From: simonzwise at gmail.com (Simon Wise) Date: Wed, 19 Mar 2014 01:07:19 +1100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <53283477.8050006@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <5327BE68.4040302@gmail.com> <53283477.8050006@kudla.org> Message-ID: <53285317.7010309@gmail.com> On 18/03/14 22:56, Rob wrote: > On 03/17/2014 11:32 PM, Simon Wise wrote: >> I'm more interested in midi, osc, puredata >> etc being able to connect via UDP and TCP in an isolated local network, >> preferably via ethernet cable, as far as linux audio uses are concerned. >> And testing USB audio, potentially using that. > > I hope that you'll share your results with us, as I (at least) am very > interested in hearing about it and maybe trying to attempt it. My Android > music-making so far has been a few rudimentary trackers and similar > programs, and getting something "real" on there would be preferable until > the gap has been filled. Yes, I'll let you know. Messing around with the system a bit this evening, the Debian roots I installed from the Play Store (that is a seriously daft name!) are certainly working properly, and I have explored the way they are invoked (standard chroot stuff, just a bit disguised) but they aren't really suited to what I want. They point the way to putting my own in ... they have not required adding root access. Looks like I will do the changes needed via the development interface, rather than mess with the local permissions at all (well ... changing the odd file ownership and turning down the volume on Samsung's "protective" Knox application might make it all more workable). Connectbot definitely does not do it for me ... the local terminal isn't working properly at all, and anyway the interface is all wrong for a 10 inch device and a standard bluetooth keyboard. I'll stick with juiceSSH until Debian and an X server can replace it, keeping it around for those times I want a terminal without a real keyboard. I also tested SSHelper and that is being blocked by Samsung's rather zealous Knox (the icon should be a gilded chastity belt rather than a shield). But the help page suggests a way to simply change its settings behind its back (so the suggestion was well worth it, thanks). I haven't tried that yet but I hope that it is accessible via adb. I had a closer look at what was going on, using the debian chroot, and should be able to undo a lot of the silly restrictions... it would seem that Samsung is more interested in locking things down than Google/Android are, and I certainly want to keep the Samsung drivers and interface stuff fully intact because it is very useful, and I paid them for it so I'd like to keep it, thanks, and without the "helpful" security measures. Simon From simonzwise at gmail.com Tue Mar 18 14:23:28 2014 From: simonzwise at gmail.com (Simon Wise) Date: Wed, 19 Mar 2014 01:23:28 +1100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532710EB.2040801@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> Message-ID: <532856E0.9040009@gmail.com> On 18/03/14 02:12, Rob wrote: > I also have an SSH server installed. I used to use sshdroid, but at some > point they got overzealous about detecting ad blockers (any change to your > hosts file is treated as an ad blocker and the service will fail to start) > so I've switched to SSHelper, which is FOSS. However, my current phone > (Galaxy S4 on Sprint) seems to only do IPv6 on the cell network, if there's > an option to disable that I haven't been able to find it, and I can't get > inbound connections to work with either. But for mounting stuff over sftp, > it works reasonably well on wifi. if sftp works but not a shell then it is possibly Samsung blocking it with Knox, if your Galaxy has it. I'll let you know if I succeed in turning it down without removing it. Simon From lau at kudla.org Tue Mar 18 15:10:27 2014 From: lau at kudla.org (Rob) Date: Tue, 18 Mar 2014 11:10:27 -0400 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532856E0.9040009@gmail.com> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <532856E0.9040009@gmail.com> Message-ID: <532861E3.4090905@kudla.org> On 03/18/2014 10:23 AM, Simon Wise wrote: >> an option to disable that I haven't been able to find it, and I can't get >> inbound connections to work with either. But for mounting stuff over sftp, >> it works reasonably well on wifi. > if sftp works but not a shell then it is possibly Samsung blocking it with > Knox, if your Galaxy has it. I'll let you know if I succeed in turning it > down without removing it. sftp and shell both work over wifi, but neither work over the cell network. I thought it was due to the ipv6 thing, but today I have an ipv4 address and still can't connect -- and whatismyip.com tells me I have a different address, suggesting that my carrier has me behind a proxy. (That IP address doesn't work, either.) But the stock shell, as you observed, is not that useful, especially without root. So, even on wifi, I pretty much only use it for sftp. I don't seem to have Knox on my phone; I haven't seen the shield, and there's nothing about it in settings. There's a system update that's been pending since I got the phone, and I've been avoiding letting it install precisely because I don't want that. (On XDA's forum for my carrier's version of the phone, there are lots of threads that mention "de-Knoxed" stock ROMs.) Rob From hanswil at notam02.no Tue Mar 18 15:20:30 2014 From: hanswil at notam02.no (Hans Wilmers) Date: Tue, 18 Mar 2014 16:20:30 +0100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532861E3.4090905@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <532856E0.9040009@gmail.com> <532861E3.4090905@kudla.org> Message-ID: <5328643E.3030106@notam02.no> On 03/18/2014 04:10 PM, Rob wrote: > > sftp and shell both work over wifi, but neither work over the cell network. > I thought it was due to the ipv6 thing, but today I have an ipv4 address > and still can't connect -- and whatismyip.com tells me I have a different > address, suggesting that my carrier has me behind a proxy. (That IP address > doesn't work, either.) But the stock shell, as you observed, is not that > useful, especially without root. So, even on wifi, I pretty much only use > it for sftp. > Most providers disallow public access to GPRS equipment for security reasons, unless you really want it. You can ask your provider for an APN that will give you a public IP. / Hans From clemens at ladisch.de Tue Mar 18 15:41:08 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Tue, 18 Mar 2014 16:41:08 +0100 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> <53282C34.5010203@ladisch.de> Message-ID: <53286914.4060102@ladisch.de> Harry van Haaren wrote: > On Tue, Mar 18, 2014 at 12:10 PM, Aiyumi Moriya wrote: > > Do you think Firewire has any chance of working? I had no interest to > pursue it until now, and have no idea of the state of Firewire support > on Linux. > > The Motif XF isn't listed: > http://www.ffado.org/?q=devicesupport%2Flist&filter0=yamaha&filter1=&op2=OR It is unsupported, like all the other mLAN devices. Regards, Clemens From lau at kudla.org Tue Mar 18 15:46:05 2014 From: lau at kudla.org (Rob) Date: Tue, 18 Mar 2014 11:46:05 -0400 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <5328643E.3030106@notam02.no> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <532856E0.9040009@gmail.com> <532861E3.4090905@kudla.org> <5328643E.3030106@notam02.no> Message-ID: <53286A3D.4080806@kudla.org> On 03/18/2014 11:20 AM, Hans Wilmers wrote: >> sftp and shell both work over wifi, but neither work over the cell network. > Most providers disallow public access to GPRS equipment for security > reasons, unless you really want it. You can ask your provider for an APN > that will give you a public IP. Well, that sucks. My last two phones (the original Samsung Galaxy S, and before that, the Palm Pre) and my partner's current phone (Galaxy S2) both had/have public IP addresses, which I used all the time because rsyncing to the phone was much more reliable than any of the SCP options I tried on the phone. Must be because the S4 has LTE that they disabled it. Now I'm gonna have to figure something out with Tasker or maybe write an app of my own to set up a reverse SSH tunnel to my colo server. Rob From ydjeho at gmail.com Tue Mar 18 17:32:41 2014 From: ydjeho at gmail.com (=?ISO-8859-1?Q?Dj=E9ho_Youn?=) Date: Tue, 18 Mar 2014 18:32:41 +0100 Subject: [LAU] 100% (of course) linux live set: Z.Karkowski tribute concert Message-ID: Hello, I want to share a little live set I did a few weeks ago in memory of Zbigniew Karkowski @ Krolikarnia, Warsaw (PL) http://jhyoun.wordpress.com/2014/03/18/z-karkowski-tribute-live-recording-is-up/ Supercollider 3.6 + Ubuntu 13.10 + HID/MIDI controllers. 32bit wav file converted to mp3 using Audacity. enjoy! -- Jae Ho YOUN http://jhyoun.wordpress.com/ http://jaehoyoun.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandre.prokoudine at gmail.com Tue Mar 18 19:28:17 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Tue, 18 Mar 2014 23:28:17 +0400 Subject: [LAU] cursynth 1.4 Message-ID: Hi, Apparently Matt Tytel released initial version of Cursynth, a ncurses-based polysynth. http://www.gnu.org/software/cursynth/ Since there are a few visually impaired users around here, I'm curious: does it work well for you? Alexandre From federicogalland at gmail.com Tue Mar 18 20:45:32 2014 From: federicogalland at gmail.com (Federico Galland) Date: Tue, 18 Mar 2014 17:45:32 -0300 Subject: [LAU] cursynth 1.4 In-Reply-To: References: Message-ID: <20140318174532.b30cb7535d6b746c110d9910@gmail.com> On Tue, 18 Mar 2014 23:28:17 +0400 Alexandre Prokoudine wrote: > Hi, > > Apparently Matt Tytel released initial version of Cursynth, a > ncurses-based polysynth. > > http://www.gnu.org/software/cursynth/ > > Since there are a few visually impaired users around here, I'm > curious: does it work well for you? > > Alexandre > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Congrats, looks awesome! Will report back once I've tested how it sounds! From ydjeho at gmail.com Tue Mar 18 21:43:40 2014 From: ydjeho at gmail.com (=?ISO-8859-1?Q?Dj=E9ho_Youn?=) Date: Tue, 18 Mar 2014 22:43:40 +0100 Subject: [LAU] 100% (of course) linux live set: Z.Karkowski tribute concert In-Reply-To: References: Message-ID: also on: https://archive.org/details/jaehoyoun2014-03-04 On Tue, Mar 18, 2014 at 6:32 PM, Dj?ho Youn wrote: > Hello, > > I want to share a little live set I did a few weeks ago in memory of > Zbigniew Karkowski @ Krolikarnia, Warsaw (PL) > > > http://jhyoun.wordpress.com/2014/03/18/z-karkowski-tribute-live-recording-is-up/ > > Supercollider 3.6 + Ubuntu 13.10 + HID/MIDI controllers. 32bit wav file > converted to mp3 using Audacity. > > enjoy! > > -- > Jae Ho YOUN > > http://jhyoun.wordpress.com/ > http://jaehoyoun.com > > -- Jae Ho YOUN http://jhyoun.wordpress.com/ http://jaehoyoun.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From aiyumi.br at gmail.com Tue Mar 18 21:49:24 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Tue, 18 Mar 2014 18:49:24 -0300 Subject: [LAU] cursynth 1.4 In-Reply-To: References: Message-ID: 2014-03-18 16:28 GMT-03:00, Alexandre Prokoudine : > Since there are a few visually impaired users around here, I'm > curious: does it work well for you? "Finally! A command line synthesizer to play with! " Was what I thought. I did a quick compile without running "make install". But running the program from the build directory, I got: RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (hw:0,0), Device or resource busy. RtApiAlsa::probeDeviceOpen: pcm device (hw:0,0) won't open for output. It's probably because of my screen readers that are using the soundcard, the same problem I have with JACK. But in this case, it doesn't look like we can choose another soundcard to use (no command line options apparently). -- ____________________ Blog: http://aiyumi.warpstar.net/ From aiyumi.br at gmail.com Tue Mar 18 22:11:18 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Tue, 18 Mar 2014 19:11:18 -0300 Subject: [LAU] Yamaha Motif XF as USB Soundcard In-Reply-To: <53286914.4060102@ladisch.de> References: <53261084.4030400@ladisch.de> <5326B43F.3090101@ladisch.de> <53282C34.5010203@ladisch.de> <53286914.4060102@ladisch.de> Message-ID: 2014-03-18 9:10 GMT-03:00, Aiyumi Moriya : > So the audio interface function is Firewire only... I don't know what > made me believe that USB also worked. I think I found my point of confusion. It was MOX. I was too excited when I stumbled upon the MOX thread. I downloaded the MOX manual just to check, and it turns out that, indeed, it can work as audio interface via USB, unlike the Motif XF. Quote: "The USB [TO HOST] terminal is used to connect this instrument to the computer via the USB cable and allows you to transfer MIDI data and audio data between the devices." Since the MOX is a simpler keyboard from the same product line, I thought the MOX and Motif were similar enough, and that tutorials made for one could be applied to the other. That's the cause for this whole problem (and that's why I was sure that the USB option was mentioned on the tutorials I read). 2014-03-18 12:41 GMT-03:00, Clemens Ladisch : > Harry van Haaren wrote: >> On Tue, Mar 18, 2014 at 12:10 PM, Aiyumi Moriya wrote: >> >> Do you think Firewire has any chance of working? I had no interest to >> pursue it until now, and have no idea of the state of Firewire >> support >> on Linux. >> >> The Motif XF isn't listed: >> http://www.ffado.org/?q=devicesupport%2Flist&filter0=yamaha&filter1=&op2=OR > > It is unsupported, like all the other mLAN devices. Ok. Case solved, then. Once again, thank you for the help. -- ____________________ Blog: http://aiyumi.warpstar.net/ From robin at gareus.org Tue Mar 18 23:26:19 2014 From: robin at gareus.org (Robin Gareus) Date: Wed, 19 Mar 2014 00:26:19 +0100 Subject: [LAU] ebur128 batch processing In-Reply-To: References: Message-ID: <5328D61B.6020005@gareus.org> On 03/07/2014 01:48 AM, Jarom?r Mike? wrote: > Hi, > > I will got about 100 stereo wav files mixed hopefully in similar way > (loudness). > I need them process to meet ebur128 specification. > True peaks -3dB > RMS -23dB > > Any chance to do it as batch process? http://freelcs.sourceforge.net/ From simonzwise at gmail.com Wed Mar 19 03:09:33 2014 From: simonzwise at gmail.com (Simon Wise) Date: Wed, 19 Mar 2014 14:09:33 +1100 Subject: [LAU] OT: android for the linux user ...was: legacy equipment for a user who does not "program" In-Reply-To: <532861E3.4090905@kudla.org> References: <53268570.3010907@jvlnet.com> <1395061665.9232.187.camel@archlinux> <20140317091528.bfc352c90e0f2a0e7c2bc5ce@brainiac.com> <5326F893.4030100@woh.rr.com> <53270A44.9050604@gmail.com> <532710EB.2040801@kudla.org> <532856E0.9040009@gmail.com> <532861E3.4090905@kudla.org> Message-ID: <53290A6D.2060209@gmail.com> On 19/03/14 02:10, Rob wrote: > I don't seem to have Knox on my phone; I haven't seen the shield, and > there's nothing about it in settings. There's a system update that's been > pending since I got the phone, and I've been avoiding letting it install > precisely because I don't want that. (On XDA's forum for my carrier's > version of the phone, there are lots of threads that mention "de-Knoxed" > stock ROMs.) Sounds right. For the pen and interface I want the current Samsung stuff. I want to avoid any bulk replacement of the system, I'd rather learn it and adjust it myself, that's the reason I wanted an Android system (an ARM one) to use everyday ... paired with a good desktop machine. That was the way I got to know linux properly, after first using it only on projects but using OSX daily. Then Apple dropped PPC and I wanted an pen, so went to a Fujitsu Lifebook and Debian, which lasted me 6 years. But the last 4 years of that was doing a computing and maths degree, which certainly accelerated the learning curve. I guess full-time study was my version of a mid-life crisis after 25 years doing theatre projects freelance! Simon From dj_kaza at hotmail.com Wed Mar 19 03:11:07 2014 From: dj_kaza at hotmail.com (Kaza Kore) Date: Wed, 19 Mar 2014 03:11:07 +0000 Subject: [LAU] Sabrent or Vantec USB audio interface Message-ID: Hi all. I was wondering if anybody has any light they can shine on compatability of either of these two external, USB audio interfaces: Sabrent USB-SND8 http://www.sabrent.com/category/audio/USB-SND8/ Vantec USB External 7.1 Channel Audio Adapter http://www.vantecusa.com/en/product/view_detail/439 I am currently travelling in India with nothing but the onboard sound of my laptop and would like to get something with 4 channel (2 stereo pair) outputs, at least for use with Mixxx. These both are available at a reasonable price through Amazon India... It's been hard to find much online, and what I have seen has been slightly contradictory (plus it can be painful trying to search and read posts when the internet is quite as interminable as it can be here!) Any experiences, or even any better searching ability and summarisation of the findings, would be very much appreciated. Regards, Dale. -------------- next part -------------- An HTML attachment was scrubbed... URL: From marc at hacklava.net Wed Mar 19 06:51:27 2014 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Wed, 19 Mar 2014 02:51:27 -0400 Subject: [LAU] Sabrent or Vantec USB audio interface In-Reply-To: References: Message-ID: <20140319025127.2c3237a1@telecino> Hello Dale. I have a Sabrant USB-SND8. It works out of the box on Linux. The only "problem" I have with this interface: there could be a strong click while switching on the interface when it's connected to power amps; so make sure your amps have good pop filters, or mute them before powering the interface. I can't comment much on its sound quality, but I use it to listen to multichannel audio at normal domestic levels, and it's fine. Just don't expect professional quality. At the price it's worth a try. -- Marc Wed, 19 Mar 2014 03:11:07 +0000, Kaza Kore wrote : > Hi all. > > I was wondering if anybody has any light they can shine on > compatability of either of these two external, USB audio interfaces: > > > Sabrent USB-SND8 http://www.sabrent.com/category/audio/USB-SND8/ > > Vantec USB External 7.1 Channel Audio Adapter > http://www.vantecusa.com/en/product/view_detail/439 > > I am currently travelling in India with nothing but the onboard sound > of my laptop and would like to get something with 4 channel (2 stereo > pair) outputs, at least for use with Mixxx. These both are available > at a reasonable price through Amazon India... > > It's been hard to find much online, and what I have seen has been > slightly contradictory (plus it can be painful trying to search and > read posts when the internet is quite as interminable as it can be > here!) Any experiences, or even any better searching ability and > summarisation of the findings, would be very much appreciated. > > Regards, Dale. > From clemens at ladisch.de Wed Mar 19 07:53:07 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Wed, 19 Mar 2014 08:53:07 +0100 Subject: [LAU] Sabrent or Vantec USB audio interface In-Reply-To: References: Message-ID: <53294CE3.8030204@ladisch.de> Kaza Kore wrote: > Vantec USB External 7.1 Channel Audio Adapter http://www.vantecusa.com/en/product/view_detail/439 That thing needs a Windows driver and does not work with Macs, so it won't work with Linux either. Regards, Clemens From dj_kaza at hotmail.com Wed Mar 19 08:04:19 2014 From: dj_kaza at hotmail.com (Kaza Kore) Date: Wed, 19 Mar 2014 08:04:19 +0000 Subject: [LAU] Sabrent or Vantec USB audio interface In-Reply-To: <53294CE3.8030204@ladisch.de> References: , <53294CE3.8030204@ladisch.de> Message-ID: Thanks guys. Think I'll try and get a Sabrant when I'm next think I'll be still long enough to get one delivered and see how I get on with it. :) Dale. > Date: Wed, 19 Mar 2014 08:53:07 +0100 > From: clemens at ladisch.de > To: linux-audio-user at lists.linuxaudio.org > Subject: Re: [LAU] Sabrent or Vantec USB audio interface > > Kaza Kore wrote: > > Vantec USB External 7.1 Channel Audio Adapter http://www.vantecusa.com/en/product/view_detail/439 > > That thing needs a Windows driver and does not work with Macs, so it > won't work with Linux either. > > > Regards, > Clemens > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Wed Mar 19 10:41:40 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Wed, 19 Mar 2014 11:41:40 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140317194734.GB3858@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> <20140316172531.GA4250@traun.gscholz.bayernline.de> <20140317194734.GB3858@traun.gscholz.bayernline.de> Message-ID: <53297464.3020904@gmail.com> On 03/17/2014 08:47 PM, Guido Scholz wrote: > Am Sun, 16. Mar 2014 um 19:37:28 +0100 schrieb rosea grammostola: > > Hi rosea, > >> Huh no NSM support? > yes such things happen usually if no-one sends a patch upstream. > > Hi Guido, Would be nice if you could add it to AMS nevertheless https://github.com/royvegard/ams It would help simple users as me a lot :) Thanks in advance, \r From federicogalland at gmail.com Wed Mar 19 10:43:20 2014 From: federicogalland at gmail.com (Federico Galland) Date: Wed, 19 Mar 2014 07:43:20 -0300 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <53297464.3020904@gmail.com> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140315225558.GA12467@linuxaudio.org> <20140316172531.GA4250@traun.gscholz.bayernline.de> <20140317194734.GB3858@traun.gscholz.bayernline.de> <53297464.3020904@gmail.com> Message-ID: <20140319074320.6ee7f88468a499866277cc72@gmail.com> I second the request. Also if there was a way to save the poliphony and sample rate settings, that would be awesome! Thanks! On Wed, 19 Mar 2014 11:41:40 +0100 "rosea.grammostola" wrote: > On 03/17/2014 08:47 PM, Guido Scholz wrote: > > Am Sun, 16. Mar 2014 um 19:37:28 +0100 schrieb rosea grammostola: > > > > Hi rosea, > > > >> Huh no NSM support? > > yes such things happen usually if no-one sends a patch upstream. > > > > > Hi Guido, > > Would be nice if you could add it to AMS nevertheless > > https://github.com/royvegard/ams > > It would help simple users as me a lot :) > > Thanks in advance, > \r > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Federico Galland From perodog at gmx.net Wed Mar 19 12:18:49 2014 From: perodog at gmx.net (Dragan Noveski) Date: Wed, 19 Mar 2014 13:18:49 +0100 Subject: [LAU] compiling cursynth 1.4 on linux Message-ID: <53298B29.8040402@gmx.net> hallo, alexandre prokoudine just 'announced' the cursynth at the LAU list yesterday. first of all, thanks for your effort on developing this nice looking jackified synth. trying to compile, i get an error about missing 'soundcard.h'. i am using this configure line: ./configure --prefix=/usr --with-jack --without-oss --without-pulse and also i tried this one: ./configure --prefix=/usr --with-jack --with-oss=no --with-pulse=no but both times it looks to me that oss and pulse are considered to be compiled. are this switches non functional? my system: nowhiskey at murija7:~/Desktop/src/cursynth-1.4$ uname -a Linux murija7 3.13-5.slh.4-aptosid-686 #1 SMP PREEMPT Tue Mar 4 21:17:55 UTC 2014 i686 GNU/Linux i tried to install everything what i found to have to do with soundcard.h but it does not work. here the error: nowhiskey at murija7:~/Desktop/src/cursynth-1.4$ make make all-recursive make[1]: Entering directory `/home/nowhiskey/Desktop/src/cursynth-1.4' Making all in cJSON make[2]: Entering directory `/home/nowhiskey/Desktop/src/cursynth-1.4/cJSON' CC cJSON.o AR libcJSON.a make[2]: Leaving directory `/home/nowhiskey/Desktop/src/cursynth-1.4/cJSON' Making all in rtaudio make[2]: Entering directory `/home/nowhiskey/Desktop/src/cursynth-1.4/rtaudio' CXX RtAudio.o RtAudio.cpp:6888:23: fatal error: soundcard.h: Datei oder Verzeichnis nicht gefunden #include "soundcard.h" ^ compilation terminated. make[2]: *** [RtAudio.o] Fehler 1 make[2]: Leaving directory `/home/nowhiskey/Desktop/src/cursynth-1.4/rtaudio' make[1]: *** [all-recursive] Fehler 1 make[1]: Leaving directory `/home/nowhiskey/Desktop/src/cursynth-1.4' make: *** [all] Fehler 2 nowhiskey at murija7:~/Desktop/src/cursynth-1.4$ if you have any suggestion what the problem could be, please email me. cheers, doc From silvain at freeshell.de Wed Mar 19 18:43:43 2014 From: silvain at freeshell.de (F. Silvain) Date: Wed, 19 Mar 2014 19:43:43 +0100 (CET) Subject: [LAU] compiling cursynth 1.4 on linux In-Reply-To: <53298B29.8040402@gmx.net> References: <53298B29.8040402@gmx.net> Message-ID: <1403191938420.14622@freeshell.de> Dragan Noveski, Mar 19 2014: > trying to compile, i get an error about missing 'soundcard.h'. I compiled with: ./configure --with-jack --with-alsa and it translated and linked correctly. Pulse and OSS are installed on my system though. Yet subjecting the executable to ldd only shows libasound and libjack for audio interfaces. HTH Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From gnome at hawaii.rr.com Wed Mar 19 18:44:56 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 19 Mar 2014 08:44:56 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <532776D2.4060909@free.fr> References: <53262DEB.5080207@gmail.com> <5326C7B1.1070606@gmail.com> <532776D2.4060909@free.fr> Message-ID: <5329E5A8.9040002@hawaii.rr.com> On 03/17/2014 12:27 PM, Beno?t Rouits wrote: > Le 17/03/2014 18:29, Louigi Verona a ?crit : >> Lo-fi competition is a great idea, imho. >> >> >> On Mon, Mar 17, 2014 at 2:00 PM, Lorenzo Sutton >> > wrote: >> >> On 17/03/2014 03:20, Paul Davis wrote: >> >> >> >> >> On Sun, Mar 16, 2014 at 7:04 PM, Lorenzo Sutton >> >> > >> wrote: >> >> In light of the interesting discussion on sample rates I >> propose a >> music competition among LAU around production of music >> pieces with >> quality considered 'low' by current dominating >> professional/audiophile standards in the digital domain: >> >> Specifics to be discussed, but I would start with the >> following: >> >> >> can we just re-record the beatles or miles davis from vinyl and >> consider >> it done? >> >> >> You got he point - well said :-) > > i have somewhere (if i can find it) some field recording taken with a > mono, 8bit, 16KHz rate from a so-called dictaphone, overdubbed with a > stereo piano track at 16bit/44.1Khz.. the effect is not so bad. > Well, to say, lo-fi is interseting to me. this proposition is a good > idea, and sometime big restrictions can lead to cool productions. > - Ben I could try recording things with my Palm Tungsten T3 PDA, it has a built-in mic. Getting recordings out adds yet another conversion step, since the only way I can get a recording out of it is to plug the headphone out jack into my sound card and record it as the PDA plays it ... First piece of music I ever heard that had been digitized was a bit of a Van Halen song that had been digitized using a Commodore 64's direct input connection (an 8-bit connection direct to the CPU). The CPU ran at about 1MHz and spent about 25% of its cycles dedicated to video stuff, so I have no idea what the effective sample rate was. The singing and words were recognizable. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Wed Mar 19 18:46:36 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 19 Mar 2014 08:46:36 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: References: <53262DEB.5080207@gmail.com> <5326991B.7010008@hawaii.rr.com> Message-ID: <5329E60C.5030707@hawaii.rr.com> On 03/16/2014 11:43 PM, Neil C Smith wrote: > On 17 March 2014 06:41, david wrote: >> OK, now I've got to track down a way to digitize a mix tape I made for my >> first girlfriend, back in early 70's. It was a mix tape of my own songs, >> including one with me singing (badly) and playing bottle neck guitar >> (passably) on a cheap steel-string acoustic that couldn't stay in tune for >> more than a minute. Recorded onto 1/4" cassette using a portable tape >> recorder designed to record voice dictation or interview, not music. > > I've heard that love is blind .. it would seem it's deaf too! :-P > > N She wasn't deaf. Could explain why I still have the tape: she broke up with me before I could give it to her. ;-) -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Wed Mar 19 18:58:39 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 19 Mar 2014 08:58:39 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <20140317083020.GB9131@tal> References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> <20140317083020.GB9131@tal> Message-ID: <5329E8DF.8070601@hawaii.rr.com> On 03/16/2014 10:30 PM, Chris Bannister wrote: > On Sun, Mar 16, 2014 at 07:25:32PM -1000, david wrote: >> Or maybe you mean how music sounded on old battery-powered AM >> portable radios? > > Isn't stereo AM supposed to be better than stereo FM? > > P.S. I've never personally heard stereo AM, but the stations compress > the bejesus out of their FM transmissions -- bloody horrible! I couldn't tell. I don't think I've ever heard stereo AM. They probably compress the bejesus out of AM stereo transmissions, too, because they have to overcome the same problem regardless of frequency range (variable signal strength), right? When I was listening to AM radio on old portable battery-powered AM radios, there was no AM stereo. FM was the only broadcast option for stereo at the time. I quit listening to AM radio in 1970 when the local Top 40 station's listeners voted a very young Michael Jackson "male" vocalist of the year. But regardless of the signal type, the radio circuitry that converted it to audio generally wasn't even close to the best, and they certainly didn't put a pile of money into the single, typically small, speaker. Then add in the possible effects of aging on components. Someone on the list must have an old portable AM radio still hanging around in a box in the attic. I might have one, but it's off somewhere on the mainland if it even still exists. Well, my mother probably still has the old AC/Delco "battery" AM radio my dad's former employer gave away ages ago. What an eBay-collectible item that would be now! -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From perodog at gmx.net Wed Mar 19 19:24:15 2014 From: perodog at gmx.net (Dragan Noveski) Date: Wed, 19 Mar 2014 20:24:15 +0100 Subject: [LAU] compiling cursynth 1.4 on linux In-Reply-To: <1403191938420.14622@freeshell.de> References: <53298B29.8040402@gmx.net> <1403191938420.14622@freeshell.de> Message-ID: <5329EEDF.6020206@gmx.net> On 19.03.2014 19:43, F. Silvain wrote: > Dragan Noveski, Mar 19 2014: >> trying to compile, i get an error about missing 'soundcard.h'. > I compiled with: > ./configure --with-jack --with-alsa > and it translated and linked correctly. Pulse and OSS are installed on > my system though. > Yet subjecting the executable to ldd only shows libasound and libjack > for audio interfaces. > HTH > > Ta-ta > ---- > Ffanci > * Internet: http://freeshell.de/~silvain > all right, thanks for the help. that way it compiles here too. no oss and pulse installed here. but now - does it work with jack for you? here, with alsa the synth makes nice noise, but when started with jack, noise is coming out of the speakers immediately, as soon the cursynth is started. cheers, doc From willgodfrey at musically.me.uk Wed Mar 19 19:26:55 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Wed, 19 Mar 2014 19:26:55 +0000 Subject: [LAU] Music made with Linux - a couple of tracks In-Reply-To: <1403031239020.27779@freeshell.de> References: <1403031239020.27779@freeshell.de> Message-ID: <20140319192655.2be9eeb3@debian> On Mon, 3 Mar 2014 12:42:41 +0100 (CET) "F. Silvain" wrote: > Hey hey, > I took the plunge. Here are my first three tracks, recorded on Linux. > http://freeshell.de/~silvain/audio/beatus.ogg > http://freeshell.de/~silvain/audio/dragonride.ogg > http://freeshell.de/~silvain/audio/letter_to_the_minnesaenger.ogg > > FWIW I also added a small poem. Maybe I'll set some music to it in future: > http://freeshell.de/~silvain/vogonic.html > > Please give me some feedback, because I want to learn for the future. :) > > Ta-ta Finally had time to listen to these. The first half of Beatus seemed very hesitant and there were some rather suspect bass notes (might be due to the nature of the sound you used), but afther the halfway mark the whole thing seemed to pick up considerably. Dragonride seemed much more polished and well rounded. A very enjoyable track. Letter to the Minnesaenger was an interesting exercise, and I liked the multiple arp and tune threads but something like this, heavily quantised - as I suspect it was - tends to sound rather mechanical. Fine if that's your intention but my personal preference is for music that 'breathes' more, if you know what IO mean. Anyway thanks for sharing :) -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From federicogalland at gmail.com Wed Mar 19 20:04:54 2014 From: federicogalland at gmail.com (Federico Galland) Date: Wed, 19 Mar 2014 17:04:54 -0300 Subject: [LAU] compiling cursynth 1.4 on linux In-Reply-To: <5329EEDF.6020206@gmx.net> References: <53298B29.8040402@gmx.net> <1403191938420.14622@freeshell.de> <5329EEDF.6020206@gmx.net> Message-ID: <20140319170454.d18cb6f175f3aed531e4f1ac@gmail.com> > here, with alsa the synth makes nice noise, but when started with jack, > noise is coming out of the speakers immediately, as soon the cursynth is > started. Same thing here. Only stops to crash. From silvain at freeshell.de Wed Mar 19 20:17:32 2014 From: silvain at freeshell.de (F. Silvain) Date: Wed, 19 Mar 2014 21:17:32 +0100 (CET) Subject: [LAU] Music made with Linux - a couple of tracks In-Reply-To: <20140319192655.2be9eeb3@debian> References: <1403031239020.27779@freeshell.de> <20140319192655.2be9eeb3@debian> Message-ID: <1403192105300.15865@freeshell.de> Will Godfrey, Mar 19 2014: ... > The first half of Beatus seemed very hesitant and there were some rather > suspect bass notes (might be due to the nature of the sound you used), but > afther the halfway mark the whole thing seemed to pick up considerably. Firstly thank you for listening and writing, Will. I've listened to eatus again and the hesitation is intended. the bass notes seem OK, yet the sound is rather particular. > Dragonride seemed much more polished and well rounded. A very enjoyable track. Thank you very much. > Letter to the Minnesaenger was an interesting exercise, and I liked the > multiple arp and tune threads but something like this, heavily quantised - as > I suspect it was - tends to sound rather mechanical. There was no quantisation applied whatsoever. This indicates that I need to balance better between perfection and liveliness. ... Once more: thank you for your insight and constructive criticism. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From simonzwise at gmail.com Thu Mar 20 05:58:26 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 16:58:26 +1100 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> <20140317083020.GB9131@tal> Message-ID: <532A8382.90404@gmail.com> On 17/03/14 19:46, Rapha?l Mouneyres wrote: >> Isn't stereo AM supposed to be better than stereo FM? >> >> P.S. I've never personally heard stereo AM, but the stations compress >> the bejesus out of their FM transmissions -- bloody horrible! Depends on the taste of the audience and station. FM here started with serious music stations, with the audience mostly using good receivers in a good sound system. And was very much better quality than AM at the time, not lo-fi sound at all unless played on a tiny little portable ... but since they were not compressing you lost most of the music in that case. Simon From simonzwise at gmail.com Thu Mar 20 06:10:53 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 17:10:53 +1100 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <5329E8DF.8070601@hawaii.rr.com> References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> <20140317083020.GB9131@tal> <5329E8DF.8070601@hawaii.rr.com> Message-ID: <532A866D.5060905@gmail.com> On 20/03/14 05:58, david wrote: > I couldn't tell. I don't think I've ever heard stereo AM. They probably compress > the bejesus out of AM stereo transmissions, too, because they have to overcome > the same problem regardless of frequency range (variable signal strength), right? no .. signal strength is a more significant problem with AM, the audio signal is based on the amplitude envelope of the carrier signal, while in FM the frequency of the carrier signal is varied according to the audio, and the receiver follows that. That is of course a simplification, but they are quite different. Simon From pedro.lopez.cabanillas at gmail.com Thu Mar 20 06:16:24 2014 From: pedro.lopez.cabanillas at gmail.com (Pedro Lopez-Cabanillas) Date: Thu, 20 Mar 2014 07:16:24 +0100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk Message-ID: <29688721.q73XtD24xL@boccanegra.localdomain> Hi, I don't know who is "j_e_f_f_g" and how to reach him by mail, so hopefully he is subscribed to this list. j_e_f_f_g: you have been editing every page in http://wiki.linuxaudio.org/apps for some time. The revision below is wrong. vmpk doesn't "display notes on a staff". Including the program in the "Score recognition" category is misleading. Perhaps your revision comes from watching the demo video on YouTube, without realizing that it involves another program (musescore) displaying and playing a score, while connected to vmpk which highlights the piano keys when it receives MIDI events. You may replace musescore by another MIDI player. Here is a similar demo involving kmidimon instead: http://youtu.be/3TGNSYKjEtg That is one use case for vmpk, but for most people the main feature would be creating MIDI events. Regards, Pedro ---------- Forwarded Message ---------- Subject: [Linux-Sound] page changed: apps:all:vmpk Date: Thursday 20 March 2014, 01:30:41 From: apps at linuxaudio.org Hello! The page apps:all:vmpk in the Linux-Sound wiki changed. Here are the changes: -------------------------------------------------------- @@ -3,8 +3,8 @@ VMPK is a virtual MIDI piano keyboard for Linux, Windows and OSX. Based on Qt4 and RtMIDI, the program is a MIDI event generator using the computer's alphanumeric keyboard and the mouse. It may be used also to display received MIDI notes. \\ - {{tag>ALSA_SEQ MIDI_Software Jack_MIDI virtual_midi}} + {{tag>ALSA_SEQ MIDI_Software Jack_MIDI virtual_midi score_recognition_software}} ~~META:title=Virtual MIDI Piano Keyboard~~ - ~~META:desc=Virtual MIDI piano keyboard. Uses JackMidi~~ + ~~META:desc=Qt4 app that displays notes on a staff, and graphical piano, as you play. Uses JackMidi~~ ~~META:logo=:wiki:user:lad:images:vmpk.png~~ ~~META:link=http://vmpk.sourceforge.net/~~ ~~META:screenshot=http://vmpk.sourceforge.net/images/vmpk-0.3.0-linux.png~~ ~~META:banner=~~ -------------------------------------------------------- Date : 2014/03/20 01:30 User : j_e_f_f_g Edit Summary: Old Revision: http://wiki.linuxaudio.org/apps/all/vmpk?rev=1393320469 New Revision: http://wiki.linuxaudio.org/apps/all/vmpk To cancel the page notifications, log into the wiki at http://wiki.linuxaudio.org/ then visit http://wiki.linuxaudio.org/apps/all/vmpk and unsubscribe page and/or namespace changes. -- This mail was generated by DokuWiki at http://wiki.linuxaudio.org/ ----------------------------------------- From jeremy at autostatic.com Thu Mar 20 08:16:43 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 20 Mar 2014 09:16:43 +0100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <29688721.q73XtD24xL@boccanegra.localdomain> References: <29688721.q73XtD24xL@boccanegra.localdomain> Message-ID: <532AA3EB.1050907@autostatic.com> On 03/20/2014 07:16 AM, Pedro Lopez-Cabanillas wrote: > Hi, > > I don't know who is "j_e_f_f_g" and how to reach him by mail, so hopefully he > is subscribed to this list. > > j_e_f_f_g: you have been editing every page in http://wiki.linuxaudio.org/apps > for some time. The revision below is wrong. vmpk doesn't "display notes on a > staff". Including the program in the "Score recognition" category is > misleading. > > Perhaps your revision comes from watching the demo video on YouTube, without > realizing that it involves another program (musescore) displaying and playing > a score, while connected to vmpk which highlights the piano keys when it > receives MIDI events. You may replace musescore by another MIDI player. Here > is a similar demo involving kmidimon instead: http://youtu.be/3TGNSYKjEtg > > That is one use case for vmpk, but for most people the main feature would be > creating MIDI events. > > Regards, > Pedro Hello Pedro, The LAU wiki is a community thing so feel free to change the article or revert to an older revision. I could take a look at it too or maybe one of the other active Wiki editors (who can be reached on IRC #opensourcemusicians). Fwiw, you might be able to reach j_e_f_f_g via the mail address that features on his site: http://home.roadrunner.com/~jgglatt/ Afaik he is not subscribed to this list. Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From pedro.lopez.cabanillas at gmail.com Thu Mar 20 08:38:08 2014 From: pedro.lopez.cabanillas at gmail.com (Pedro Lopez-Cabanillas) Date: Thu, 20 Mar 2014 09:38:08 +0100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AA3EB.1050907@autostatic.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> Message-ID: On Thu, Mar 20, 2014 at 9:16 AM, Jeremy Jongepier wrote: > The LAU wiki is a community thing so feel free to change the article or > revert to an older revision. I could take a look at it too or maybe one > of the other active Wiki editors (who can be reached on IRC > #opensourcemusicians). Fwiw, you might be able to reach j_e_f_f_g via > the mail address that features on his site: > http://home.roadrunner.com/~jgglatt/ Afaik he is not subscribed to this > list. > > Bye, > > Jeremy > Doesn't look like a community thing, from this perspective. He is the only one editing this wiki: http://wiki.linuxaudio.org/feeds?do=recent How do you know that j_e_f_f_g is Jeff Glatt? Regards, Pedro -------------- next part -------------- An HTML attachment was scrubbed... URL: From meissner.fritz at gmail.com Thu Mar 20 09:05:03 2014 From: meissner.fritz at gmail.com (Fritz Meissner) Date: Thu, 20 Mar 2014 11:05:03 +0200 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> Message-ID: That's a name that brings back memories. I still use his midi assembler/disassembler occasionally and I learned most of what I know about midi from his midi fanatic's brainwashing page - on geocities if I remember correctly. Fritz On 20 Mar 2014 10:38, "Pedro Lopez-Cabanillas" < pedro.lopez.cabanillas at gmail.com> wrote: > > On Thu, Mar 20, 2014 at 9:16 AM, Jeremy Jongepier wrote: > >> The LAU wiki is a community thing so feel free to change the article or >> revert to an older revision. I could take a look at it too or maybe one >> of the other active Wiki editors (who can be reached on IRC >> #opensourcemusicians). Fwiw, you might be able to reach j_e_f_f_g via >> the mail address that features on his site: >> http://home.roadrunner.com/~jgglatt/ Afaik he is not subscribed to this >> list. >> >> Bye, >> >> Jeremy >> > > Doesn't look like a community thing, from this perspective. He is the only > one editing this wiki: > http://wiki.linuxaudio.org/feeds?do=recent > > How do you know that j_e_f_f_g is Jeff Glatt? > > Regards, > Pedro > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From simonzwise at gmail.com Thu Mar 20 09:19:08 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 20:19:08 +1100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AA3EB.1050907@autostatic.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> Message-ID: <532AB28C.9010400@gmail.com> On 20/03/14 19:16, Jeremy Jongepier wrote: > > The LAU wiki is a community thing so feel free to change the article or > revert to an older revision. I could take a look at it too or maybe one > of the other active Wiki editors (who can be reached on IRC > #opensourcemusicians). Fwiw, you might be able to reach j_e_f_f_g via > the mail address that features on his site: > http://home.roadrunner.com/~jgglatt/ Afaik he is not subscribed to this > list. I hope most of it is more accurate than the example, since there has been a very busy week of editing and a lot of pages replaced, or at least deleted ... the only activity this year, so indeed nothing much else is happening. Simon From simonzwise at gmail.com Thu Mar 20 09:35:31 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 20:35:31 +1100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AB28C.9010400@gmail.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> <532AB28C.9010400@gmail.com> Message-ID: <532AB663.7070503@gmail.com> ... first random sample of deleted pages ... http://wiki.linuxaudio.org/apps/all/lilypond_guide_for_beginners?rev=1272211347 the page has a link to a recently updated Lillypond guide, http://eugenecormier.com/?p=147 and nothing else shows up in the current wiki if I search for lillypond guide. But maybe it is moved somewhere? Simon From simonzwise at gmail.com Thu Mar 20 09:38:09 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 20:38:09 +1100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AB663.7070503@gmail.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> <532AB28C.9010400@gmail.com> <532AB663.7070503@gmail.com> Message-ID: <532AB701.4050900@gmail.com> On 20/03/14 20:35, Simon Wise wrote: > ... first random sample of deleted pages ... > > http://wiki.linuxaudio.org/apps/all/lilypond_guide_for_beginners?rev=1272211347 > > the page has a link to a recently updated Lillypond guide, > > http://eugenecormier.com/?p=147 > > and nothing else shows up in the current wiki if I search for lillypond guide. > > But maybe it is moved somewhere? sorry ... an oldish lillypond guide ... is it no longer accurate? has lillypond changed that much? Simon From jeremy at autostatic.com Thu Mar 20 09:41:51 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 20 Mar 2014 10:41:51 +0100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> Message-ID: <532AB7DF.9020203@autostatic.com> On 03/20/2014 09:38 AM, Pedro Lopez-Cabanillas wrote: > Doesn't look like a community thing, from this perspective. He is the only > one editing this wiki: > http://wiki.linuxaudio.org/feeds?do=recent > Thanks for the heads up, I'll start keeping an eye on it. Currently the linuxaudio.org is being worked on to make it a more consistent whole. This started with the LinuxMusicians wiki being merged with the linuxaudio.org one and is an ongoing effort. > How do you know that j_e_f_f_g is Jeff Glatt? > He posted threads on the LinuxMusicians forum about software he wrote that could be downloaded from the very same site (http://home.roadrunner.com/~jgglatt/progs/linux.htm) Bye, Jeremy > Regards, > Pedro -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From simonzwise at gmail.com Thu Mar 20 10:26:41 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 20 Mar 2014 21:26:41 +1100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AB7DF.9020203@autostatic.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AA3EB.1050907@autostatic.com> <532AB7DF.9020203@autostatic.com> Message-ID: <532AC261.8070401@gmail.com> On 20/03/14 20:41, Jeremy Jongepier wrote: > On 03/20/2014 09:38 AM, Pedro Lopez-Cabanillas wrote: >> Doesn't look like a community thing, from this perspective. He is the only >> one editing this wiki: >> http://wiki.linuxaudio.org/feeds?do=recent >> > > Thanks for the heads up, I'll start keeping an eye on it. Currently the > linuxaudio.org is being worked on to make it a more consistent whole. > This started with the LinuxMusicians wiki being merged with the > linuxaudio.org one and is an ongoing effort. more browsing ... certainly quite a few dead links removed (following further on the lillypond one it finally did end in a dead link) ... and certainly quite a few real pages added, so real editing is being done, a lot of it. Simon From nachoen79 at hotmail.com Thu Mar 20 16:58:19 2014 From: nachoen79 at hotmail.com (Nacho -) Date: Thu, 20 Mar 2014 17:58:19 +0100 Subject: [LAU] jack on mac-mini and debian Message-ID: Hi there. Can you help me with this doubt, please? I have got this problem, but on my mac-mini: http://jackaudio.org/macbook_distortion I'm working with a debian wheezy distro and can get only clean sound forcing the 16 bits on jack 2. No problem working via alsa but without jack. Instead I have tried some live dvd audio oriented distros on the same computer and then all works perfect, also with 32 bits. Any idea about what I must change? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: From pedro.lopez.cabanillas at gmail.com Thu Mar 20 18:40:35 2014 From: pedro.lopez.cabanillas at gmail.com (Pedro Lopez-Cabanillas) Date: Thu, 20 Mar 2014 19:40:35 +0100 Subject: [LAU] Fwd: [Linux-Sound] page changed: apps:all:vmpk In-Reply-To: <532AB7DF.9020203@autostatic.com> References: <29688721.q73XtD24xL@boccanegra.localdomain> <532AB7DF.9020203@autostatic.com> Message-ID: <3516822.jFfQ974hWs@boccanegra.localdomain> On Thursday 20 March 2014 10:41:51 Jeremy Jongepier wrote: > On 03/20/2014 09:38 AM, Pedro Lopez-Cabanillas wrote: > mail address that features on his site: > http://home.roadrunner.com/~jgglatt/ The error that the other server returned was: 550 Invalid mailbox: jjglatt at adelphia.net > > How do you know that j_e_f_f_g is Jeff Glatt? > > He posted threads on the LinuxMusicians forum about software he wrote > that could be downloaded from the very same site > (http://home.roadrunner.com/~jgglatt/progs/linux.htm) It is an empty page. Regards, Pedro From rustompmody at gmail.com Fri Mar 21 05:40:22 2014 From: rustompmody at gmail.com (Rustom Mody) Date: Fri, 21 Mar 2014 11:10:22 +0530 Subject: [LAU] Jeff Glatt (was page changed: apps:all:vmpk) Message-ID: On Fri, Mar 21, 2014 at 12:10 AM, Pedro Lopez-Cabanillas wrote: > On Thursday 20 March 2014 10:41:51 Jeremy Jongepier wrote: >> On 03/20/2014 09:38 AM, Pedro Lopez-Cabanillas wrote: > >> mail address that features on his site: >> http://home.roadrunner.com/~jgglatt/ > > The error that the other server returned was: > 550 Invalid mailbox: jjglatt at adelphia.net > >> > How do you know that j_e_f_f_g is Jeff Glatt? >> >> He posted threads on the LinuxMusicians forum about software he wrote >> that could be downloaded from the very same site >> (http://home.roadrunner.com/~jgglatt/progs/linux.htm) Thanks to this thread I looked up Jeff Glatt. Found this at top -- Heh!! ------------------ Lesser life forms communicate in more primitive, less artistic manners such as barks, chirps, roars, or any of the many human verbal languages in use throughout the globe. But, gods communicate using MIDI messages. Even a picture, whose value is often equated to a thousand words, cannot match the scope of emotional and intellectual power unleashed by a stream of MIDI messages. But, try to tell that to the visual-and-print-fixated, tone-deaf heathen who infest this planet... -------------------- Reading through I find a glimmer of hope that I may move on from perennial noob status So thanks! Rusi -- http://www.the-magus.in http://blog.languager.org From rncbc at rncbc.org Fri Mar 21 17:18:49 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Fri, 21 Mar 2014 17:18:49 +0000 Subject: [LAU] [ANN] Qtractor 0.6.0 - The Byte Bald Beta release! Message-ID: <532C7479.9040504@rncbc.org> Once again, it's springtime. Once again, the planet revolved and stepped forward against its mother star. Hence the three bees release... it won't be that far fetched if you find this naming a deliberate pun riddle indeed--the wise will know already while the clueless will get it sudden in a couple of days ;) Well, no big surprises nor earthshaking features are being here pitched then, just some small and otherwise humble improvements and fixes and what not. Nevertheless and not just for the record that is, it marks the day of a long overdue and severely procrastinated beta phase release. Whatever. It's dang official now: one step closer to omega, no doubt ;) Contrary to, maybe, newer generations, I always strive to say the least. As to say about literal face-value meaning, as written and read on timeless textbooks, it's all left in between the lines or, this part is the one I like most, left over as an exercise--in some kind of healthy masochism, perhaps :). No wonder, Qtractor 0.6.0 (byte bald beta) is now released! Release highlights: * Plugin automation high resolution option (NEW) * Plugin 'About' page (NEW) * Native DE dialogs option (NEW) * Follow play-head slack time (NEW) * MIDI RPN/NRPN 14-bit controllers input (FIX) Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor Downloads: - source tarball: http://downloads.sourceforge.net/qtractor/qtractor-0.6.0.tar.gz - source package (openSUSE 13.1): http://downloads.sourceforge.net/qtractor/qtractor-0.6.0-10.rncbc.suse131.src.rpm - binary packages (openSUSE 13.1): http://downloads.sourceforge.net/qtractor/qtractor-0.6.0-10.rncbc.suse131.i586.rpm http://downloads.sourceforge.net/qtractor/qtractor-0.6.0-10.rncbc.suse131.x86_64.rpm - quick start guide & user manual (outdated): http://downloads.sourceforge.net/qtractor/qtractor-0.5.x-user-manual.pdf - help wanted (on Qtractor Wiki ;)) http://sourceforge.net/p/qtractor/wiki Weblog (upstream support): http://www.rncbc.org License: Qtractor is free, open-source software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. Change-log: - New user option added: on whether to save plugins automation values with higher resolution as possible, using 14-bit NRPN: cf. View/Options.../Plugins/Experimental/High resolution plugin automation (default=off). - Generic native plugin dialogs now shows an additional "About" last page where authorship credits are due. - A new user preference option is now in place for whether to use desktop environment's own native file requester/browser dialogs (View/Options.../Display/Dialogs/Use native dialogs). - A bit of slack have been introduced to put "Follow Playhead" (aka. auto-scroll view mode) on hold, while doing in-flight selection edit moves. - Fixed some user interface related annoyances while on the MIDI Controllers mappings (ie. View/Controllers...). - Fixed port origin on MIDI RPN/NRPN 14-bit controllers input. - A discretionary plug-in unique identifier have been devised for when more than one from the same type are inserted on a bus or track chain, avoiding destructive clashing of automation data. - Horizontal scrolling shift+mouse-wheel direction now reversed. - LV2 Dyn(amic)-manifest support is now optional (default=off); cf. View/Options.../Plugins/Experimental/LV2 Dynamic Manifest support). - The following options, although decieved on View/Options... as global configuration options, were always and still are proper session instance properties: (JACK) Transport mode, MMC mode, MMC device, MIDI SPP and MIDI Clock modes, are now shown there reflecting the current open session state. - A couple of run-time circumventions have been hacked in, both strictly related to when NSM session management is in charge: 1) the new session template feature is disabled (was aborting initial NSM new client additions); 2) the native (as from the desktop environment eg. KDE) file browser/requester dialogs are disabled (were taking too long to list the current directory on first time invocation). - Update current automation/curve nodes selection while changing horizontal (time axis) zoom levels. - One liner's attempt to make it consistent behaviour on resizing and moving multiple selected notes or events while on the MIDI clip editor (aka. piano-roll; after a ticket request from Daniel MacDonald aka. danboid, thanks). - Introducing tiny quarter-note/crotchet/seminima/beat icon on all snap-to-beat selection items get a new icon :). - Corrected some audio buffering boundary conditions that were causing dead-loops/freezes while merging some audio clips. - Session auto-save period was chronically reduced to one third of its user setting; non critical but fixed now. See also: http://www.rncbc.org/drupal/node/772 See you all on LAC2014 at ZKM-Karlsruhe! and have a happy new springtime, cheers! Enjoy && yet again, have (lots of) fun. -- rncbc aka Rui Nuno Capela From list at nilsgey.de Fri Mar 21 18:17:18 2014 From: list at nilsgey.de (Nils) Date: Fri, 21 Mar 2014 19:17:18 +0100 Subject: [LAU] Your Twitter accounts, I want them! :) Message-ID: <532C822E.7010505@nilsgey.de> Hello users (and developers), do you have a Twitter account remotely connected to Linux Audio, music or programming in general? I would like to follow you, maybe write a blog article recommending Twitter accounts from the LA-community. Nils http://nilsgey.de https://twitter.com/NilsGey From harryhaaren at gmail.com Fri Mar 21 19:01:49 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 21 Mar 2014 19:01:49 +0000 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: OpenAV Productions posts come from @harryhaaren Cheers, -Harry On 21 Mar 2014 19:17, "Nils" wrote: > Hello users (and developers), > > do you have a Twitter account remotely connected to Linux Audio, music or > programming in general? > > I would like to follow you, maybe write a blog article recommending > Twitter accounts from the LA-community. > > Nils > > http://nilsgey.de > https://twitter.com/NilsGey > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From studiochanning at yahoo.com Fri Mar 21 19:42:10 2014 From: studiochanning at yahoo.com (Studio Channing) Date: Fri, 21 Mar 2014 13:42:10 -0600 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: <532C9612.3090703@yahoo.com> doesn't directly answer your question but everybody should be aware of diaspora, and that people are making public posts about linux audio, e.g.: https://diasp.eu/tags/linuxaudio would be great to see more people on there, unlike twitter it is noncommercial and running on FOSS http://en.wikipedia.org/wiki/Diaspora_%28social_network%29 find open pods at http://podupti.me/ On 03/21/2014 12:17 PM, Nils wrote: > Hello users (and developers), > > do you have a Twitter account remotely connected to Linux Audio, music > or programming in general? > > I would like to follow you, maybe write a blog article recommending > Twitter accounts from the LA-community. > > Nils > > http://nilsgey.de > https://twitter.com/NilsGey > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From gnome at hawaii.rr.com Sat Mar 22 05:41:39 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 21 Mar 2014 19:41:39 -1000 Subject: [LAU] Proposal for 'lo-fi' music competition In-Reply-To: <532A866D.5060905@gmail.com> References: <53262DEB.5080207@gmail.com> <5326874C.60800@hawaii.rr.com> <20140317083020.GB9131@tal> <5329E8DF.8070601@hawaii.rr.com> <532A866D.5060905@gmail.com> Message-ID: <532D2293.5030102@hawaii.rr.com> On 03/19/2014 08:10 PM, Simon Wise wrote: > On 20/03/14 05:58, david wrote: > >> I couldn't tell. I don't think I've ever heard stereo AM. They >> probably compress >> the bejesus out of AM stereo transmissions, too, because they have to >> overcome >> the same problem regardless of frequency range (variable signal >> strength), right? > > no .. signal strength is a more significant problem with AM, the audio > signal is based on the amplitude envelope of the carrier signal, while > in FM the frequency of the carrier signal is varied according to the > audio, and the receiver follows that. That is of course a > simplification, but they are quite different. Cool, thanks for clarifying that. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From blablack at gmail.com Sat Mar 22 16:17:26 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sat, 22 Mar 2014 16:17:26 +0000 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released Message-ID: > AlsaModularSynth is a MIDI controlled realtime modular synthesizer > and effect processor with support for LADSPA and JACK. > > After several years of collecting fixes and enhancements the new > release provides a long list of changes: > > ams-2.1.0 (2014-03-15) That's great! Is it ok to port the selected below into ams-lv2? http://github.com/blablack/ams-lv2 > Fixed Bugs > > o Fix triggered reset of LFO saw signals, patch provided by Bill > Yerazunis > > New Features > > o New V8 Sequencer module, provided by Bill Yerazunis > o New Analog Memory module, provided by Bill Yerazunis > o New Bitgrinder module, provided by Bill Yerazunis > o New Hysteresis module, provided by Bill Yerazunis > o New VC-Delay module, provided by Bill Yerazunis > o Add Pulsetrain Noise type to Noise 2 module, patch provided by > Bill Yerazunis > o New FFT Vocoder module, provided by Bill Yerazunis Aur?lien From temps.jo at gmail.com Sat Mar 22 22:07:36 2014 From: temps.jo at gmail.com (pierre jocelyn andre) Date: Sat, 22 Mar 2014 23:07:36 +0100 Subject: [LAU] lm3jo , DNA format Message-ID: Hello, I announced the release of version 1.1.1-9 lm3jo sources are here: http://www.letime.net/vocale/paquet_deb/sources.lm3jo_deb.tar.gz deb 386 is http://www.letime.net/vocale/paquet_deb/lm3jo.deb In the instance creating a library of 8-byte audio built on DNA format. Or 256 x 256 x 256 x 256 x 256 x 256 x 256 x 256 possible melodies with 8 bytes -------------- next part -------------- An HTML attachment was scrubbed... URL: From excalibas at gmail.com Sat Mar 22 22:26:02 2014 From: excalibas at gmail.com (F. Medeiros) Date: Sat, 22 Mar 2014 22:26:02 +0000 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: <532E0DFA.80001@gmail.com> Hello, I'm @vjx Cheers! F. Medeiros On 03/21/2014 06:17 PM, Nils wrote: > Hello users (and developers), > > do you have a Twitter account remotely connected to Linux Audio, music > or programming in general? > > I would like to follow you, maybe write a blog article recommending > Twitter accounts from the LA-community. > > Nils > > http://nilsgey.de > https://twitter.com/NilsGey > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From james at jwm-art.net Sat Mar 22 22:26:09 2014 From: james at jwm-art.net (James Morris) Date: Sat, 22 Mar 2014 22:26:09 +0000 Subject: [LAU] lm3jo , DNA format In-Reply-To: References: Message-ID: <20140322222609.6dabfe23@Scrapyard.lan> On Sat, 22 Mar 2014 23:07:36 +0100 pierre jocelyn andre wrote: > Hello, > > I announced the release of version 1.1.1-9 lm3jo > sources are here: > > http://www.letime.net/vocale/paquet_deb/sources.lm3jo_deb.tar.gz > deb 386 is > http://www.letime.net/vocale/paquet_deb/lm3jo.deb Please take a look at other software release announcements for examples of how to announce a software release properly. At a minimum you should provide the following information: * the basics of what the software does * where to find more information about the software * where to download it You might also supply information about what changes have been made in the new release. > In the instance creating a library of 8-byte audio built on DNA > format. Or 256 x 256 x 256 x 256 x 256 x 256 x 256 x 256 possible > melodies with 8 bytes Without knowing what the software does, that entire statement is pretty meaningless. regards, James. From gnome at hawaii.rr.com Sat Mar 22 22:48:10 2014 From: gnome at hawaii.rr.com (david) Date: Sat, 22 Mar 2014 12:48:10 -1000 Subject: [LAU] lm3jo , DNA format In-Reply-To: <20140322222609.6dabfe23@Scrapyard.lan> References: <20140322222609.6dabfe23@Scrapyard.lan> Message-ID: <532E132A.4040100@hawaii.rr.com> On 03/22/2014 12:26 PM, James Morris wrote: > On Sat, 22 Mar 2014 23:07:36 +0100 > pierre jocelyn andre wrote: > >> Hello, >> >> I announced the release of version 1.1.1-9 lm3jo >> sources are here: >> >> http://www.letime.net/vocale/paquet_deb/sources.lm3jo_deb.tar.gz >> deb 386 is >> http://www.letime.net/vocale/paquet_deb/lm3jo.deb > > > Please take a look at other software release announcements for examples > of how to announce a software release properly. > > At a minimum you should provide the following information: > > * the basics of what the software does > * where to find more information about the software > * where to download it > > You might also supply information about what changes have been made in > the new release. > >> In the instance creating a library of 8-byte audio built on DNA >> format. Or 256 x 256 x 256 x 256 x 256 x 256 x 256 x 256 possible >> melodies with 8 bytes > > Without knowing what the software does, that entire statement is > pretty meaningless. > > regards, > James. I took it to mean he was distributing a library of 8-byte audio in DNA format. Not necessarily software. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From temps.jo at gmail.com Sat Mar 22 23:31:28 2014 From: temps.jo at gmail.com (pierre jocelyn andre) Date: Sun, 23 Mar 2014 00:31:28 +0100 Subject: [LAU] lm3jo , DNA format In-Reply-To: <532E132A.4040100@hawaii.rr.com> References: <20140322222609.6dabfe23@Scrapyard.lan> <532E132A.4040100@hawaii.rr.com> Message-ID: thank you, for answers. My project is to build an audio format for linux. Audio format built on modeling The new version brings a better piano touch. The new version brings a little more dna codes, and one of its modes of operation ( library of 8 bytes ) the project is presented in French here http://www.linuxmao.org/lm3jo I placed on wikimedia some new sounds (200KB ogg) that are 3 bytes audio format dna cordially 2014-03-22 23:48 GMT+01:00 david : > On 03/22/2014 12:26 PM, James Morris wrote: > >> On Sat, 22 Mar 2014 23:07:36 +0100 >> pierre jocelyn andre wrote: >> >> Hello, >>> >>> I announced the release of version 1.1.1-9 lm3jo >>> sources are here: >>> >>> http://www.letime.net/vocale/paquet_deb/sources.lm3jo_deb.tar.gz >>> deb 386 is >>> http://www.letime.net/vocale/paquet_deb/lm3jo.deb >>> >> >> >> Please take a look at other software release announcements for examples >> of how to announce a software release properly. >> >> At a minimum you should provide the following information: >> >> * the basics of what the software does >> * where to find more information about the software >> * where to download it >> >> You might also supply information about what changes have been made in >> the new release. >> >> In the instance creating a library of 8-byte audio built on DNA >>> format. Or 256 x 256 x 256 x 256 x 256 x 256 x 256 x 256 possible >>> melodies with 8 bytes >>> >> >> Without knowing what the software does, that entire statement is >> pretty meaningless. >> >> regards, >> James. >> > > I took it to mean he was distributing a library of 8-byte audio in DNA > format. Not necessarily software. > > -- > David W. Jones > gnome at hawaii.rr.com > authenticity, honesty, community > http://dancingtreefrog.com > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From blablack at gmail.com Sun Mar 23 00:02:56 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sun, 23 Mar 2014 00:02:56 +0000 Subject: [LAU] Drums - EDrums - Hydrogen Message-ID: Hi all, I have been using Hydrogen for a few years now, both by programming drum patterns or playing live with an e-drums and there are few things I'm missing... I wouldn't mind having a look into coding them, but I'd like to get the feedback/comments/opinions/screams of despair from the community. HiHat & EDrums Hydrogen only deals with close/open for the hihat control. It should be easy enough to mark the instruments as being part of a hihat and provide for each hihat instruments a range to define which one is triggered. For example close from 0 to 10, half-open from 11 to 40, open from 41 to 127. EDrum sends a cc message for how close the hihat is closed. There is already a script to do something very similar (http://www.hydrogen-music.org/hcms/node/2807) using mididings, but having it in Hydrogen would be more user friendly I guess? EDrums & Cymbal choke I believe edrums are using aftertouch for that (to confirm) - sample could simply be muted? Multi-mics drum samples Drums recording often uses several mic placements and depending how they are mixed it changes the mood of the drums. For example mixing using overhead and direct samples. I have drum samples where: - the snare is composed of under, above and overhead takes - cymbals are direct and overhead takes - kick is front, back, inside - etc... Per instrument, we could have groups of samples: - Instrument would have one fader per group and one fader for the whole instrument - Each instrument group would have its own output - Each group would have its fader - if we have multiple instruments with the group "overhead", this fader can control the gain for the overall overhead - I don't know what to do with panner - would anybody have an opinion on that? - Trigger/velocity (programmed or played live) would trigger would trigger all necessary samples What do you guys think of these proposals? Aur?lien From cbannister at slingshot.co.nz Sun Mar 23 04:48:18 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Sun, 23 Mar 2014 17:48:18 +1300 Subject: [LAU] Drums - EDrums - Hydrogen In-Reply-To: References: Message-ID: <20140323044817.GA32019@tal> On Sun, Mar 23, 2014 at 12:02:56AM +0000, Aur?lien Leblond wrote: > Hi all, > > I have been using Hydrogen for a few years now, both by programming > drum patterns or playing live with an e-drums and there are few things > I'm missing... > > I wouldn't mind having a look into coding them, but I'd like to get > the feedback/comments/opinions/screams of despair from the community. Wouldn't it make more sense to include: http://lists.sourceforge.net/mailman/listinfo/hydrogen-users and http://lists.sourceforge.net/mailman/listinfo/hydrogen-devel in the discussion? -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X From dj_kaza at hotmail.com Sun Mar 23 06:24:58 2014 From: dj_kaza at hotmail.com (Kaza Kore) Date: Sun, 23 Mar 2014 06:24:58 +0000 Subject: [LAU] Basic Bash/Find and batch CLI questions. Message-ID: So I'm travelling with a laptop with fairly limited space and thought I would save some space by converting my flacs to 320kbs mp3 (please don't bother with the "why mp3 and not ogg" comments here) to save at least a little space, and again try and get my head around a little Bashing. So I search online and find this for Bash: #!/bin/bash if [ -d "${1}" ] ; then cd "${1}" && for f in *.flac; do ffmpeg -i "$f" -f wav - | lame -b 320 -h - "${f%.flac}.mp3"; done fi Which I've saved and from the same article use this command to activate when in the desired folder: find ./ -type d -exec ~/bin/flac2mp3 "{}" \; Can't claim to fully understand it (hence been generally playing around in a copied test folder.) So find is passing all directories, via the type argument, onto my bash script (find is not actually being used for any searching, just to recursively send all folders, right?) I don't understand the bit after the exec call, assume that's something to do with keeping the filenames?? I need to readup on Bash again! But it doesn't quite live up to my needs. This method is case sensitive so wont find .Flac or .FLAC files, of which I'm sure I have a few. Find can do this happily with -iname though! But this brings me to something weird I've just encountered with Find... *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor$ find -iname *wey* ./flac/test/weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).flac ./flac/weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).flac ./flac/weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).mp3 ./flac/weyheyhey !! - Wearing A Shirt That Says 'Microphyst'.flac ./flac/weyheyhey !! - Wearing A Shirt That Says 'Microphyst'.mp3 ./flac/[225] Weyheyhey !! - I'm Your Daddy.flac ./flac/[225] Weyheyhey !! - I'm Your Daddy.mp3 *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor$ cd flac/ *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor/flac$ find -iname *wey* find: paths must precede expression: weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).mp3 Usage: find [-H] [-L] [-P] [-Olevel] [-D help|tree|search|stat|rates|opt|exec] [path...] [expression] I hope you can see what I think is weird there. if I do the same for *wan* I get correct results in both folders. Noticed this as trying to find *.flac wouldn't work within the folder I had moved these too for testing and wondered if it was because the search was the same name as the root folder but it's clearly not. Any idea what's going wrong here? I had hoped to use find ./ -iname *.flac -exec ~/bin/flac2mp3 "{}" \; to activate my above script but find not reliably searching has temporarily scuppered this idea... Once I am happy with this obviously it will be time to delete the original flacs. Should I use "find ./ -type f -iname *.flac -delete" or is there a reason most guides seem to suggest using -exec rm as argument? And while I'm here... As you can probably see the collection here is for trying to get a laptop DJ set together, for which I plan to use Mixxx. Firstly, does Mixxx support the Replay Gain Tag in mp3s (I think that's the right name.) If so it would obviously make sense to set all tracks to a similar RMS. What would be the best software with which to do this? First I would want to find the loudest section (of say 2-5 seconds long, not the whole song and not too short to catch something like brief feedback as being the calibration level) and then take the RMS value of that section and set the Replay Gain so that all of these match a reasonable value. Regards, Dale. -------------- next part -------------- An HTML attachment was scrubbed... URL: From guido-scholz at gmx.net Sun Mar 23 08:03:25 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Sun, 23 Mar 2014 09:03:25 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: <20140317221931.GA4205@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> <20140317221931.GA4205@traun.gscholz.bayernline.de> Message-ID: <20140323080325.GA6391@traun.gscholz.bayernline.de> Am Mon, 17. Mar 2014 um 23:19:31 +0100 schrieb Guido Scholz: > Am Mon, 17. Mar 2014 um 23:50:31 +0400 schrieb Alexandre Prokoudine: > > Hi Alexandre, > > > Granted, I'm probably expected to have those plugins installed, but > > lack thereof shouldn't crash the app :) > > OK, I found an other patch file crashing here. I will see what I can do > (but not today ;). This is fixed in CVS now, thanks for your report Alexandre. Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From guido-scholz at gmx.net Sun Mar 23 08:05:53 2014 From: guido-scholz at gmx.net (Guido Scholz) Date: Sun, 23 Mar 2014 09:05:53 +0100 Subject: [LAU] AlsaModularSynth (ams) 2.1.0 released In-Reply-To: References: Message-ID: <20140323080553.GB6391@traun.gscholz.bayernline.de> Am Sat, 22. Mar 2014 um 16:17:26 +0000 schrieb Aur?lien Leblond: Hi Aur?lien, > That's great! > Is it ok to port the selected below into ams-lv2? > http://github.com/blablack/ams-lv2 yes of course, the code is GPL-2. You can do what you want, as long as you keep the license. Guido -- http://wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From james at jwm-art.net Sun Mar 23 10:18:05 2014 From: james at jwm-art.net (James Morris) Date: Sun, 23 Mar 2014 10:18:05 +0000 Subject: [LAU] Basic Bash/Find and batch CLI questions. In-Reply-To: References: Message-ID: <20140323101805.2d7e506e@Scrapyard.lan> On Sun, 23 Mar 2014 06:24:58 +0000 Kaza Kore wrote: > So I'm travelling with a laptop with fairly limited space and thought > I would save some space by converting my flacs to 320kbs mp3 (please > don't bother with the "why mp3 and not ogg" comments here) to save at > least a little space, and again try and get my head around a little > Bashing. > > So I search online and find this for Bash: > #!/bin/bash > if [ -d "${1}" ] ; then > cd "${1}" && for f in *.flac; do ffmpeg -i "$f" -f wav - | lame -b > 320 -h - "${f%.flac}.mp3"; done fi > > Which I've saved and from the same article use this command to > activate when in the desired folder: find ./ -type d -exec > ~/bin/flac2mp3 "{}" \; > Personally I'd just put it all in the script and just pass a single path to the script. > > Can't claim to fully understand it (hence been generally playing > around in a copied test folder.) So find is passing all directories, > via the type argument, onto my bash script (find is not actually > being used for any searching, just to recursively send all folders, > right?) I don't understand the bit after the exec call, assume that's > something to do with keeping the filenames?? I need to readup on Bash > again! > Yes, the "-type d" tells find to find directories only, "-type f" for files. THe part after the exec call simply tells find to execute the command for each result, (as opposed to passing all results to command in a single go). \; terminates what find considers to be part of the command (unlike many other programs where a command to be executed is simply specified last). > > But it doesn't quite live up to my needs. This method is case > sensitive so wont find .Flac or .FLAC files, of which I'm sure I have > a few. Find can do this happily with -iname though! But this brings > me to something weird I've just encountered with Find... You could replace the "for f in *.flac; do" part in the script with another find command, but then you'll need to get clever to do the piping, this might help: http://stackoverflow.com/questions/307015/how-do-i-include-a-pipe-in-my-linux-find-exec-command (I've found the answer by flolo has met my needs adequately before). Regarding your other problems, you need to quote the fileglob that you pass to find, otherwise Bash (or your shell of choice) will expand it, aswell as pass a path to find: find . -iname '*wey*' Regards, James. > > *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor$ find > -iname *wey* ./flac/test/weyheyhey !! - Little Batty Foo Foo (ft. > TechDiff's Modest Loft Conversion remix).flac ./flac/weyheyhey !! - > Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion > remix).flac ./flac/weyheyhey !! - Little Batty Foo Foo (ft. > TechDiff's Modest Loft Conversion remix).mp3 ./flac/weyheyhey !! - > Wearing A Shirt That Says 'Microphyst'.flac ./flac/weyheyhey !! - > Wearing A Shirt That Says 'Microphyst'.mp3 ./flac/[225] Weyheyhey !! > - I'm Your Daddy.flac ./flac/[225] Weyheyhey !! - I'm Your Daddy.mp3 > *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor$ cd flac/ > *@*:/media/Data/Music/Laptop DJ Tracks/DJ Audio/Dancefloor/flac$ find > -iname *wey* find: paths must precede expression: weyheyhey !! - > Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion > remix).mp3 Usage: find [-H] [-L] [-P] [-Olevel] [-D > help|tree|search|stat|rates|opt|exec] [path...] [expression] > > I hope you can see what I think is weird there. if I do the same for > *wan* I get correct results in both folders. Noticed this as trying > to find *.flac wouldn't work within the folder I had moved these too > for testing and wondered if it was because the search was the same > name as the root folder but it's clearly not. Any idea what's going > wrong here? I had hoped to use find ./ -iname *.flac -exec > ~/bin/flac2mp3 "{}" \; to activate my above script but find not > reliably searching has temporarily scuppered this idea... > > > Once I am happy with this obviously it will be time to delete the > original flacs. Should I use "find ./ -type f -iname *.flac -delete" > or is there a reason most guides seem to suggest using -exec rm as > argument? > > > And while I'm here... As you can probably see the collection here is > for trying to get a laptop DJ set together, for which I plan to use > Mixxx. Firstly, does Mixxx support the Replay Gain Tag in mp3s (I > think that's the right name.) If so it would obviously make sense to > set all tracks to a similar RMS. What would be the best software with > which to do this? First I would want to find the loudest section (of > say 2-5 seconds long, not the whole song and not too short to catch > something like brief feedback as being the calibration level) and > then take the RMS value of that section and set the Replay Gain so > that all of these match a reasonable value. > > Regards, Dale. > From gnome at hawaii.rr.com Sun Mar 23 10:22:04 2014 From: gnome at hawaii.rr.com (david) Date: Sun, 23 Mar 2014 00:22:04 -1000 Subject: [LAU] jack no longer starting with Audiophile 2496 on desktop PC Message-ID: <532EB5CC.5070700@hawaii.rr.com> It was working. Now it's not. JACK starts up fine with the built-in audio @ 96K and ~2.33msec latency. Doesn't work with 2496 @ 96K (or 48K) at much higher latencies. Using kernel 3.2.0-4 AMD64 Debian Sid via Aptosid, with and without RT; also tried kernel 3.13 (non-RT). I am a member of the audio group. alsa-base shows v1.0.25+3, things like alsa-tools-gui report 1.0.27-2, alsa-utils 1.027.2-1. Starting from command line gives me this: david at sempronbox:~$ /usr/bin/jackd -dalsa -dhw:M2496 -r96000 -p512 -n2 -Xseq jackdmp 1.9.10 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2013 Grame. jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details no message buffer overruns no message buffer overruns no message buffer overruns JACK server starting in realtime mode with priority 10 audio_reservation_init Acquire audio card Audio3 creating alsa driver ... hw:M2496|hw:M2496|512|2|96000|0|0|nomon|swmeter|-|32bit configuring for 96000Hz, period = 512 frames (5.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback port created: Midi-Through:midi/playback_1 port created: Midi-Through:midi/capture_1 port created: E-MU-XMidi1X1:midi/playback_1 port created: E-MU-XMidi1X1:midi/capture_1 port created: M-Audio-Audiophile-24/96:midi/playback_1 port created: M-Audio-Audiophile-24/96:midi/capture_1 ALSA: poll time out, polled for 15999020 usecs JackAudioDriver::ProcessAsync: read error, stopping... ^CJack main caught signal 2 port deleted: E-MU-XMidi1X1:midi/playback_1 port deleted: M-Audio-Audiophile-24/96:midi/playback_1 port deleted: Midi-Through:midi/playback_1 port deleted: E-MU-XMidi1X1:midi/capture_1 port deleted: M-Audio-Audiophile-24/96:midi/capture_1 port deleted: Midi-Through:midi/capture_1 Released audio card Audio3 audio_reservation_finish Was doing it in previous 3.2 non RT kernel, updated to 3.13 kernel and still doing it. QJackCtl reports this: 21:24:33.243 Patchbay deactivated. 21:24:33.254 Statistics reset. 21:24:33.255 ALSA connection change. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started 21:24:39.134 Startup script... 21:24:39.135 artsshell -q terminate Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started sh: 1: artsshell: not found 21:24:39.537 Startup script terminated with exit status=32512. 21:24:39.537 JACK is starting... 21:24:39.537 /usr/bin/jackd -dalsa -dhw:M2496 -r96000 -p512 -n2 -Xseq 21:24:39.539 JACK was started with PID=4684. no message buffer overruns no message buffer overruns no message buffer overruns jackdmp 1.9.10 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2013 Grame. jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK server starting in realtime mode with priority 10 audio_reservation_init Acquire audio card Audio3 creating alsa driver ... hw:M2496|hw:M2496|512|2|96000|0|0|nomon|swmeter|-|32bit configuring for 96000Hz, period = 512 frames (5.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback port created: Midi-Through:midi/playback_1 port created: Midi-Through:midi/capture_1 port created: E-MU-XMidi1X1:midi/playback_1 port created: E-MU-XMidi1X1:midi/capture_1 port created: M-Audio-Audiophile-24/96:midi/playback_1 21:24:39.733 ALSA connection graph change. port created: M-Audio-Audiophile-24/96:midi/capture_1 21:24:46.671 Could not connect to JACK server as client. - Overall operation failed. - Server communication error. Please check the messages window for more info. JackPosixProcessSync::LockedTimedWait error usec = 5000000 err = Connection timed out Driver is not running Cannot create new client Cannot read socket fd = 16 err = Success CheckRes error JackSocketClientChannel read fail Cannot open qjackctl client 21:24:48.496 JACK is stopping... Jack main caught signal 15 ALSA: poll time out, polled for 15999021 usecs JackAudioDriver::ProcessAsync: read error, stopping... port deleted: E-MU-XMidi1X1:midi/playback_1 port deleted: M-Audio-Audiophile-24/96:midi/playback_1 port deleted: Midi-Through:midi/playback_1 port deleted: E-MU-XMidi1X1:midi/capture_1 port deleted: M-Audio-Audiophile-24/96:midi/capture_1 port deleted: Midi-Through:midi/capture_1 21:24:55.791 ALSA connection graph change. Released audio card Audio3 audio_reservation_finish 21:24:55.923 JACK was stopped successfully. 21:24:55.928 Post-shutdown script... 21:24:55.929 killall jackd jackd: no process found 21:24:56.335 Post-shutdown script terminated with exit status=256. aplay -l shows this: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1828S Analog [VT1828S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: VT1828S Digital [VT1828S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: VT1828S Alt Analog [VT1828S Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 Figuring that maybe it was a piece of dead hardware, I booted a stock Ubuntu 13.04 AMD64 live DVD, added JACK and some other pertinent stuff, set it up with the same settings I use in the installed Aptosid setup - and the damn thing worked. Linux audio is very fragile, apparently. Ideas? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From jmckernon at gmail.com Sun Mar 23 11:18:12 2014 From: jmckernon at gmail.com (James Mckernon) Date: Sun, 23 Mar 2014 11:18:12 +0000 Subject: [LAU] Basic Bash/Find and batch CLI questions. In-Reply-To: References: Message-ID: > > And while I'm here... As you can probably see the collection here is for > trying to get a laptop DJ set together, for which I plan to use Mixxx. > Firstly, does Mixxx support the Replay Gain Tag in mp3s (I think that's the > right name.) If so it would obviously make sense to set all tracks to a > similar RMS. What would be the best software with which to do this? First I > would want to find the loudest section (of say 2-5 seconds long, not the > whole song and not too short to catch something like brief feedback as > being the calibration level) and then take the RMS value of that section > and set the Replay Gain so that all of these match a reasonable value. > I recently used a CLI program called mp3gain to normalize the replay gain tag of some of my mp3s and found it fairly painfree. I'm not familiar enough with find -exec etc. to help with your first query, though. James -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmouneyres at gmail.com Sun Mar 23 12:25:12 2014 From: rmouneyres at gmail.com (raf) Date: Sun, 23 Mar 2014 13:25:12 +0100 Subject: [LAU] linuxsampler : EG envelopes in SFZ format Message-ID: hello, to continue my discussion from last october list... http://linuxaudio.org/mailarchive/lau/2013/11/3/202707 I've managed to use the envelope generator to achieve a good hihat pedal feeling, still not finished though. This is what i have for a group : key=41 loop_mode=one_shot eg8_time0=0 eg8_level0=1 eg8_time1=0.1 eg8_time1_oncc4=2 eg8_level1=0 eg8_volume=0 I have a question about how linuxsampler deal with the envelopes, especially with the egN_timeX_onccY opcode : It looks like the envelope generator is determined with the last CCY value just before the note_on message. Then changing the CCY's value won't change the opcode value, it has be fixed when the sample was triggered. So in my case with that opcode, the length of the volume envelope is determined when the sample is triggered. I would like to be able to modify this envelope length even after the sample has been triggered, is there a way to do that ? Rapha?l From clemens at ladisch.de Sun Mar 23 12:59:54 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Sun, 23 Mar 2014 13:59:54 +0100 Subject: [LAU] Basic Bash/Find and batch CLI questions. In-Reply-To: References: Message-ID: <532EDACA.20000@ladisch.de> Kaza Kore wrote: > find ./ -type d -exec ~/bin/flac2mp3 "{}" \; > > So find is passing all directories, via the type argument The type argument is a filter. > find is not actually being used for any searching, just to recursively send all folders, right? The default action is to print what has been found. With -exec, the specified program is executed for each found item. > I don't understand the bit after the exec call {} is the file name, ; ends the command to be executed. > $ find -iname *wey* > find: paths must precede expression: weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).mp3 The shell expands *wey* into multiple arguments, but -iname expects only one. It worked in the parent directory because *wey* could not be expanded there. To prevent expansion, quote it: $ find . -iname '*wey*' Regards, Clemens From dj_kaza at hotmail.com Sun Mar 23 13:50:02 2014 From: dj_kaza at hotmail.com (Kaza Kore) Date: Sun, 23 Mar 2014 13:50:02 +0000 Subject: [LAU] Basic Bash/Find and batch CLI questions. In-Reply-To: <532EDACA.20000@ladisch.de> References: , <532EDACA.20000@ladisch.de> Message-ID: > Date: Sun, 23 Mar 2014 13:59:54 +0100 > From: clemens at ladisch.de > To: dj_kaza at hotmail.com > CC: linux-audio-user at lists.linuxaudio.org > Subject: Re: [LAU] Basic Bash/Find and batch CLI questions. > > Kaza Kore wrote: > > find ./ -type d -exec ~/bin/flac2mp3 "{}" \; > > > > So find is passing all directories, via the type argument > > The type argument is a filter. Yep got that much. Filter for Directories. > > find is not actually being used for any searching, just to recursively send all folders, right? > > The default action is to print what has been found. But there is no search term. Therefore I take it it's an assumed -name *.* and passes all directory paths (but no filenames) to the executed file. > With -exec, the specified program is executed for each found item. Each found item? So I could use [-type f -iname *.flac] and it would run through on each file in turn? (Been meaning to test this out, which is what I was trying when I came across the problem answered below. Not had a chance again yet. Obviously I'd have to at least remove the [cd "${1}" && for f in *.flac; do] section but replace it with what? Are programs such as ffmpeg and lame not designed to accept a list of files? I thought many Linux programs were designed to accept exactly this as input... I'll try and do some experimenting after dinner. At least if my headache goes a bit... > > > I don't understand the bit after the exec call > > {} is the file name, ; ends the command to be executed. > Thanks. That's pretty much the conclusions I had come to. Although still need to read up on bash again as how this then relates to $f etc still confused me a bit... > > $ find -iname *wey* > > find: paths must precede expression: weyheyhey !! - Little Batty Foo Foo (ft. TechDiff's Modest Loft Conversion remix).mp3 > > The shell expands *wey* into multiple arguments, but -iname expects only one. > It worked in the parent directory because *wey* could not be expanded there. > > To prevent expansion, quote it: > > $ find . -iname '*wey*' > Still not sure while it baulks with *wey* and not with *wan* but thanks both you and James, using quotation marks cured that problem when testing earlier so all is good on that front. I do agree having a single call to the script and the search-term (find command) being included in that would make sense, rather than having it called the method I currently am. Pretty sure most CLI commands can be used in bash but again the small bits I once learn have been lost to me currently, so more reading and testing needed! And thanks other James, will look into mp3gain, it's also the first solution I found with Google. I know the subject of RMS normalisation has come up on here a few times so should probably spend some time searching the archives for this mail-list for that one (part of the reason I felt a little cheeky asking but as I had other questions too... ;) ) Dale. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mauser at smoors.de Sun Mar 23 18:37:20 2014 From: mauser at smoors.de (Sebastian Moors) Date: Sun, 23 Mar 2014 19:37:20 +0100 Subject: [LAU] [LAD] Drums - EDrums - Hydrogen In-Reply-To: References: Message-ID: <532F29E0.90107@smoors.de> Hi Aur?lien, Aur?lien Leblond wrote: > Hi all, > > I have been using Hydrogen for a few years now, both by programming > drum patterns or playing live with an e-drums and there are few things > I'm missing... > > I wouldn't mind having a look into coding them, but I'd like to get > the feedback/comments/opinions/screams of despair from the community. > > > > HiHat & EDrums > > Hydrogen only deals with close/open for the hihat control. It should > be easy enough to mark the instruments as being part of a hihat and > provide for each hihat instruments a range to define which one is > triggered. For example close from 0 to 10, half-open from 11 to 40, > open from 41 to 127. > > EDrum sends a cc message for how close the hihat is closed. > > There is already a script to do something very similar > (http://www.hydrogen-music.org/hcms/node/2807) using mididings, but > having it in Hydrogen would be more user friendly I guess? This would be really handy. I'm one of the hydrogen devs and i had a look into this problem some years ago.. At that time i was experimenting with a roland edrum and ran into the same issue. But i switched back to accoustic drums and got rid of that problem the easy way :) What i was asking myself at that time: Is there a standard way of sending hihat pressure information?? IIRC the roland was sending similar CC messages. > > > EDrums & Cymbal choke > > I believe edrums are using aftertouch for that (to confirm) - sample > could simply be muted? > No idea.. Never inspected the midi stream of my edrum when using chokes. I think sending a "virtual" note-off could be better then to mute it (then you don't have the trouble to unmute it afterwards). > > Multi-mics drum samples > > Drums recording often uses several mic placements and depending how > they are mixed it changes the mood of the drums. For example mixing > using overhead and direct samples. > I have drum samples where: > - the snare is composed of under, above and overhead takes > - cymbals are direct and overhead takes > - kick is front, back, inside > - etc... > > Per instrument, we could have groups of samples: > - Instrument would have one fader per group and one fader for the > whole instrument > - Each instrument group would have its own output > - Each group would have its fader - if we have multiple instruments > with the group "overhead", this fader can control the gain for the > overall overhead > - I don't know what to do with panner - would anybody have an opinion on that? > - Trigger/velocity (programmed or played live) would trigger would > trigger all necessary samples Well, yes.. its a nice idea. This has been proposed quite often, but no one really worked out a complete concept or even wrote a patch. I think the first stage would be to allow more then one sample per layer and to define how to choose between those layers (random, in parallel (your approach) or with a user-defined script. I would suggest to start with your first two ideas, they would be a good start to get into hydrogen development, if you're interested in that. Btw. if you want, you could create "enhancement" tickets on the hydrogen issue tracker at github. Best regards, Sebastian > > > > What do you guys think of these proposals? > > Aur?lien > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-dev From len at ovenwerks.net Sun Mar 23 21:35:35 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 23 Mar 2014 14:35:35 -0700 (PDT) Subject: [LAU] [LAD] Drums - EDrums - Hydrogen In-Reply-To: <532F29E0.90107@smoors.de> References: <532F29E0.90107@smoors.de> Message-ID: On Sun, 23 Mar 2014, Sebastian Moors wrote: > Aur?lien Leblond wrote: >> EDrums & Cymbal choke >> >> I believe edrums are using aftertouch for that (to confirm) - sample >> could simply be muted? >> > No idea.. Never inspected the midi stream of my edrum when using chokes. > I think sending a "virtual" note-off could be better then to mute it (then > you don't > have the trouble to unmute it afterwards). Wow, that opens a whole can of worms... (in my mind anyway). I am thinking the whole keyboard to play struck instruments fails to some extent, not just in drums, but anything normally played with mallets like vibes or steel drums. The keyboard is the best controller for piano or organ/synth type sounds because they all have a note off event. Because they do, a second playing of the same note will always find a note that has some time in the past found a note off. This is not the case with a struck instrument, though I am sure most synth patches that try to emulate a struck instrument effectively ignore the note off (or set the note off envelope to the same as if the note stays on). I would think on a polyphonic synth hitting the same note again would give a mix of the first note event still fading out plus the same note new event playing the same thing. So it might sound like the player hit a second cymbal sounding exactly the same as the first rather the same cymbal where hitting it a second time would stop or at least modify the sound of the first strike. Maybe keyboards (or synths) already take care of this, but what should happen is that one of them should generate a note off followed by the new note on information. In the case of a cymbal, this would be very hard to emulate, because a crash followed by a ride hit would not completely stop the crash sound, just change it. It may take several ride hits (which on a real set might each sound different in their own right) before the crash sound is not a signifcant part of the sound. Of course making a short span of keyboard (range of notes really) monophonic may work for some of this. If only cymbal sounds from the same cymbal are in that range. This would make a high hat work for example, because in a monophonic keyboard the last note is always cut off by the new one. The problem is you have to effectively have a span of keys for each drum... Well just some thoughts. While I have a few keyboards, I am not a player, I can do some string pads here and there :) I have also played drums, but it has been years since I have had a live set to play and I have not taken the time to think about why the cheap set of pads I have sound so bad... I just expect that since they are cheap. -- Len Ovens www.ovenwerks.net From rmouneyres at gmail.com Sun Mar 23 21:44:27 2014 From: rmouneyres at gmail.com (raf) Date: Sun, 23 Mar 2014 22:44:27 +0100 Subject: [LAU] [LAD] Drums - EDrums - Hydrogen In-Reply-To: References: Message-ID: <22533031-0448-496B-80D2-0DA02299BB7C@gmail.com> > I believe edrums are using aftertouch for that (to confirm) - sample > could simply be muted? most of edrums do so with an aftertouch message, mainly because edrums modules have used those messages. Often, edrum soundbanks also offer a separate key to grab (mute) the cymbal, thus another note_on message. It's simpler when you program drums instead of playing them with an electronic set. > What do you guys think of these proposals? moving Hydrogen to a more dedicated drums sampler could be great, but as for today, i find sampler more versatile to achieve good sounding results. You can for example use linuxsampler, and try out many GIG of SFZ free presets. Rapha?l From listac at nebelschwaden.de Sun Mar 23 22:01:13 2014 From: listac at nebelschwaden.de (Ede Wolf) Date: Sun, 23 Mar 2014 23:01:13 +0100 Subject: [LAU] Midi Filter & Merger In-Reply-To: <20140323080325.GA6391@traun.gscholz.bayernline.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> <20140317221931.GA4205@traun.gscholz.bayernline.de> <20140323080325.GA6391@traun.gscholz.bayernline.de> Message-ID: <532F59A9.1010708@nebelschwaden.de> Hello, I am wondering, wether there is programm that can merge events from physical midiports, preferably based on Channels. Lets say, I want to merge midi in ports 1 to 4, but all messages from any of those on channel 1 are merged & routed to midi out 1, while those on channel 2 are going to out ports 2 and 3. And can additionally do some filtering, like realtime messages are only accepted on midi in 2 and routed to all outs, while only midi in 3 accepts note on/off events and adheres to above merging rules, as ports 1 & 4, which accept CC. I am aware, that a software solution maybe won't have the performance of a hardware solution, but then again, I am not aware of any, unless you are lucky and run across an old miditemp device. Ede From rmouneyres at gmail.com Sun Mar 23 22:07:47 2014 From: rmouneyres at gmail.com (raf) Date: Sun, 23 Mar 2014 23:07:47 +0100 Subject: [LAU] Midi Filter & Merger In-Reply-To: <532F59A9.1010708@nebelschwaden.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> <20140317221931.GA4205@traun.gscholz.bayernline.de> <20140323080325.GA6391@traun.gscholz.bayernline.de> <532F59A9.1010708@nebelschwaden.de> Message-ID: > I am wondering, wether there is programm that can merge events from physical midiports, preferably based on Channels. Lets say, I want to merge midi in ports 1 to 4, but all messages from any of those on channel 1 are merged & routed to midi out 1, while those on channel 2 are going to out ports 2 and 3. > > And can additionally do some filtering, like realtime messages are only accepted on midi in 2 and routed to all outs, while only midi in 3 accepts note on/off events and adheres to above merging rules, as ports 1 & 4, which accept CC. here mididings is your software of choice : http://das.nasophon.de/mididings/ unless you're not allergic to command line software. > I am aware, that a software solution maybe won't have the performance of a hardware solution, but then again, I am not aware of any, unless you are lucky and run across an old miditemp device. i know several hardware would could do that, but all of them need programming : - a not produced anymore Soundart Chameleon with your own soundskin http://chameleon.synth.net/english/index.shtml - a community project like Midibox http://www.ucapps.de/ - a well know arduino board of your choice there are probably more possible answers cheers, Rapha?l From len at ovenwerks.net Sun Mar 23 22:09:41 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 23 Mar 2014 15:09:41 -0700 (PDT) Subject: [LAU] Midi Filter & Merger In-Reply-To: <532F59A9.1010708@nebelschwaden.de> References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> <20140317221931.GA4205@traun.gscholz.bayernline.de> <20140323080325.GA6391@traun.gscholz.bayernline.de> <532F59A9.1010708@nebelschwaden.de> Message-ID: On Sun, 23 Mar 2014, Ede Wolf wrote: > Hello, > > I am wondering, wether there is programm that can merge events from physical > midiports, preferably based on Channels. Lets say, I want to merge midi in > ports 1 to 4, but all messages from any of those on channel 1 are merged & > routed to midi out 1, while those on channel 2 are going to out ports 2 and > 3. If you are using jack for routing, then putting two inputs(captures) into one output (playback) will merge the two streams already. All you need to do is to filter the streams first. In this case you would take the input to two filters to separate the channels and filter them however you want and then put the outputs where you want. Something like qmidiroute should work just fine. > I am aware, that a software solution maybe won't have the performance of a > hardware solution, but then again, I am not aware of any, unless you are > lucky and run across an old miditemp device. Any solution is SW, this is digital stream stuff and relatively slow. Any "hardware" solution is still a small processor in a box. One of my first hardware projects was a small midi filter using a Z8 (not z80) as the processor. -- Len Ovens www.ovenwerks.net From fede2001 at hotmail.com Mon Mar 24 03:32:00 2014 From: fede2001 at hotmail.com (Federico lopez) Date: Mon, 24 Mar 2014 03:32:00 +0000 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: @OpenSourceFOH ---------------------------------------- > Date: Fri, 21 Mar 2014 19:17:18 +0100 > From: list at nilsgey.de > To: linux-audio-user at lists.linuxaudio.org > Subject: [LAU] Your Twitter accounts, I want them! :) > > Hello users (and developers), > > do you have a Twitter account remotely connected to Linux Audio, music > or programming in general? > > I would like to follow you, maybe write a blog article recommending > Twitter accounts from the LA-community. > > Nils > > http://nilsgey.de > https://twitter.com/NilsGey > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From fede2001 at hotmail.com Mon Mar 24 03:35:12 2014 From: fede2001 at hotmail.com (Federico lopez) Date: Mon, 24 Mar 2014 03:35:12 +0000 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: References: <532C822E.7010505@nilsgey.de>, Message-ID: I hit send accidentally @OpenSourceFOH mostly tweets about Linux tools for live sound, and programming. Federico ---------------------------------------- > From: fede2001 at hotmail.com > To: list at nilsgey.de; linux-audio-user at lists.linuxaudio.org > Date: Mon, 24 Mar 2014 03:32:00 +0000 > Subject: Re: [LAU] Your Twitter accounts, I want them! :) > > @OpenSourceFOH > > > ---------------------------------------- >> Date: Fri, 21 Mar 2014 19:17:18 +0100 >> From: list at nilsgey.de >> To: linux-audio-user at lists.linuxaudio.org >> Subject: [LAU] Your Twitter accounts, I want them! :) >> >> Hello users (and developers), >> >> do you have a Twitter account remotely connected to Linux Audio, music >> or programming in general? >> >> I would like to follow you, maybe write a blog article recommending >> Twitter accounts from the LA-community. >> >> Nils >> >> http://nilsgey.de >> https://twitter.com/NilsGey >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From poeticintensity at gmail.com Mon Mar 24 05:31:24 2014 From: poeticintensity at gmail.com (Jason Jones) Date: Sun, 23 Mar 2014 23:31:24 -0600 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: @Utahstudio is the account for artcitysound.com. a studio that uses Linux for production. On Mar 21, 2014 12:17 PM, "Nils" wrote: > Hello users (and developers), > > do you have a Twitter account remotely connected to Linux Audio, music or > programming in general? > > I would like to follow you, maybe write a blog article recommending > Twitter accounts from the LA-community. > > Nils > > http://nilsgey.de > https://twitter.com/NilsGey > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david at olofson.net Mon Mar 24 09:29:49 2014 From: david at olofson.net (David Olofson) Date: Mon, 24 Mar 2014 10:29:49 +0100 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: References: <532C822E.7010505@nilsgey.de> Message-ID: Not entirely sure what you're looking for, but Audiality 2 (http://audiality.org/) is really rather audio related (realtime scripted modular synth/sound engine), developed on Linux, and supports JACK... ;-) Using that for music and sound effects in my games. Some WIP tracks here: https://soundcloud.com/david-olofson/sets/audiality-2-projects @OlofsonArcade is my personal account where I talk about that, game development, and various other random nonsense. @ArcadeCat is my boss, who doesn't say much, but makes the occasional announcement related to my projects. On Mon, Mar 24, 2014 at 6:31 AM, Jason Jones wrote: > @Utahstudio is the account for artcitysound.com. a studio that uses Linux > for production. > > On Mar 21, 2014 12:17 PM, "Nils" wrote: >> >> Hello users (and developers), >> >> do you have a Twitter account remotely connected to Linux Audio, music or >> programming in general? >> >> I would like to follow you, maybe write a blog article recommending >> Twitter accounts from the LA-community. >> >> Nils >> >> http://nilsgey.de >> https://twitter.com/NilsGey >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------' From gabbe.nord at gmail.com Mon Mar 24 09:31:13 2014 From: gabbe.nord at gmail.com (Gabriel Nordeborn) Date: Mon, 24 Mar 2014 10:31:13 +0100 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: References: <532C822E.7010505@nilsgey.de> Message-ID: Mine's @zth_music :) I'm not very active, but occasionally I'll tweet more than once a month ;) On Mon, Mar 24, 2014 at 10:29 AM, David Olofson wrote: > Not entirely sure what you're looking for, but Audiality 2 > (http://audiality.org/) is really rather audio related (realtime > scripted modular synth/sound engine), developed on Linux, and supports > JACK... ;-) > > Using that for music and sound effects in my games. Some WIP tracks here: > https://soundcloud.com/david-olofson/sets/audiality-2-projects > > @OlofsonArcade is my personal account where I talk about that, game > development, and various other random nonsense. > @ArcadeCat is my boss, who doesn't say much, but makes the occasional > announcement related to my projects. > > On Mon, Mar 24, 2014 at 6:31 AM, Jason Jones > wrote: > > @Utahstudio is the account for artcitysound.com. a studio that uses > Linux > > for production. > > > > On Mar 21, 2014 12:17 PM, "Nils" wrote: > >> > >> Hello users (and developers), > >> > >> do you have a Twitter account remotely connected to Linux Audio, music > or > >> programming in general? > >> > >> I would like to follow you, maybe write a blog article recommending > >> Twitter accounts from the LA-community. > >> > >> Nils > >> > >> http://nilsgey.de > >> https://twitter.com/NilsGey > >> _______________________________________________ > >> Linux-audio-user mailing list > >> Linux-audio-user at lists.linuxaudio.org > >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > > > -- > //David Olofson - Consultant, Developer, Artist, Open Source Advocate > > .--- Games, examples, libraries, scripting, sound, music, graphics ---. > | http://consulting.olofson.net http://olofsonarcade.com | > '---------------------------------------------------------------------' > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From neil at neilcsmith.net Mon Mar 24 09:57:56 2014 From: neil at neilcsmith.net (Neil C Smith) Date: Mon, 24 Mar 2014 09:57:56 +0000 Subject: [LAU] Your Twitter accounts, I want them! :) In-Reply-To: <532C822E.7010505@nilsgey.de> References: <532C822E.7010505@nilsgey.de> Message-ID: On 21 March 2014 18:17, Nils wrote: > do you have a Twitter account remotely connected to Linux Audio, music or > programming in general? Praxis LIVE has a twitter account at @PraxisLIVE (what else? :-) ) It's a graphical environment for developing audio / video projects, including live coding. My primary development and personal usage is on Linux, though it's cross-platform. Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source intermedia development - www.praxislive.org Digital Prisoners - interactive spaces and projections - www.digitalprisoners.co.uk From perodog at gmx.net Mon Mar 24 15:54:16 2014 From: perodog at gmx.net (Dragan Noveski) Date: Mon, 24 Mar 2014 16:54:16 +0100 Subject: [LAU] compiling cursynth 1.4 on linux In-Reply-To: References: <53298B29.8040402@gmx.net> <1403191938420.14622@freeshell.de> <5329EEDF.6020206@gmx.net> Message-ID: <53305528.2010409@gmx.net> On 22.03.2014 21:22, Matt Tytel wrote: > Hey! Cursynth developer here. > > Sorry to the people who've been having problems getting Cursynth > working with JACK. > I have to say I hadn't tested it with JACK in a long time so it must > have broken along the way. > > Fixing problems with JACK is top priority for the next release. hallo, this is good to hear. just keep us up-to-date about newer vesions. cheers, doc > Also planned is the ability to select which audio output device and > MIDI input devices you want to use from within Cursynth. > > If anyone finds any more issues I'm using the github issue tracking > for now: > https://github.com/iyoko/cursynth/issues/ > > Thanks for anyone who tried it out! > Matt Tytel > http://tytel.org > > > On Wed, Mar 19, 2014 at 12:24 PM, Dragan Noveski > wrote: > > On 19.03.2014 19:43, F. Silvain wrote: > > Dragan Noveski, Mar 19 2014: > > trying to compile, i get an error about missing 'soundcard.h'. > > I compiled with: > ./configure --with-jack --with-alsa > and it translated and linked correctly. Pulse and OSS are > installed on my system though. > Yet subjecting the executable to ldd only shows libasound and > libjack for audio interfaces. > HTH > > Ta-ta > ---- > Ffanci > * Internet: http://freeshell.de/~silvain > > > all right, thanks for the help. that way it compiles here too. no > oss and pulse installed here. > but now - does it work with jack for you? > > here, with alsa the synth makes nice noise, but when started with > jack, noise is coming out of the speakers immediately, as soon the > cursynth is started. > > cheers, > doc > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From listac at nebelschwaden.de Tue Mar 25 21:39:51 2014 From: listac at nebelschwaden.de (Ede Wolf) Date: Tue, 25 Mar 2014 22:39:51 +0100 Subject: [LAU] Midi Filter & Merger In-Reply-To: References: <20140315222109.GA20139@traun.gscholz.bayernline.de> <20140317193720.GA3858@traun.gscholz.bayernline.de> <20140317221931.GA4205@traun.gscholz.bayernline.de> <20140323080325.GA6391@traun.gscholz.bayernline.de> <532F59A9.1010708@nebelschwaden.de> Message-ID: <5331F7A7.9040309@nebelschwaden.de> I am not allergic to command line, *, but to me mididings seems more like programming, which by far may overstrain me. Which unfortunately is also true for DIY Hardware. However, I am sure mididings would be the definite answer to all my needs, once understood. Would be cool to have a kind of mididings plugin in qjackctl for each midi in and out port. Or think jack rack. However, I haven't been aware (maybe I should just have tried), that jack would do somewhat reliable midi merging by itself. Still, qmidiroute as filter seems rather limited. Or I haven't realised its full potential yet. Quite possible. Sure, everything is software, even dedivated patchbays, but we do have the additional latency of travelling through USB/Parallel/Serial and back. Still, with an external clock I am confident it will be sufficient enough for me. Thanks for your input * in fact, cursynth, which I haven't been aware of, may break my "no softsynth for any reason" dogma, though I will admit, it's not really command line at all, but more terminal graphics. From unaudio at gmail.com Tue Mar 25 23:24:12 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Wed, 26 Mar 2014 01:24:12 +0200 Subject: [LAU] A spectral delay VST plug-in for Linux Message-ID: Here is a link to a public beta of my new VST plug-in, a spectral delay effect - http://lanternfishaudio.wordpress.com/downloads/. It is only available on Linux for the time being and the GUI will stop loading in 2014.07.01 but audio processing part will still work. After that time either the beta period will be extended, the plug-in released commercially or if there is not enough interest as open source. There is a manual explaining how to use or install it. It is still in fairly early stages of developments but should be usable for most users with decent hosts. It will work on Renoise and Ardour 3 and does not work on Qtractor 0.5.11 which reports it's sampling rate as 0 which, while a perfectly respectable number, is not a valid sampling rate IMO. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Wed Mar 26 06:50:53 2014 From: gnome at hawaii.rr.com (david) Date: Tue, 25 Mar 2014 20:50:53 -1000 Subject: [LAU] Musician who uses sooperlooper Message-ID: <533278CD.3060606@hawaii.rr.com> I don't know if he uses the Linux or Mac OS X version: Owen Pallett https://en.wikipedia.org/wiki/Owen_Pallett -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From unaudio at gmail.com Wed Mar 26 10:27:11 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Wed, 26 Mar 2014 12:27:11 +0200 Subject: [LAU] Musician who uses sooperlooper In-Reply-To: <533278CD.3060606@hawaii.rr.com> References: <533278CD.3060606@hawaii.rr.com> Message-ID: "|who performs solo as *Owen Pallett* or, before 2010, under the name *Final Fantasy*." Ouch... On Wed, Mar 26, 2014 at 8:50 AM, david wrote: > I don't know if he uses the Linux or Mac OS X version: > > Owen Pallett > https://en.wikipedia.org/wiki/Owen_Pallett > > -- > David W. Jones > gnome at hawaii.rr.com > authenticity, honesty, community > http://dancingtreefrog.com > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- "Cheshire-Puss," she began, "would you tell me, please, which way I ought to go from here?" "That depends a good deal on where you want to get to," said the Cat. "I don't care much where--" said Alice. "Then it doesn't matter which way you go," said the Cat. -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Wed Mar 26 10:43:53 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Wed, 26 Mar 2014 10:43:53 +0000 Subject: [LAU] Musician who uses sooperlooper In-Reply-To: <533278CD.3060606@hawaii.rr.com> References: <533278CD.3060606@hawaii.rr.com> Message-ID: On 26 Mar 2014 07:51, "david" wrote: > > I don't know if he uses the Linux or Mac OS X version: Article mentions Max/MSP, so OsX. Still cool to know SL is being used by him. Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Wed Mar 26 11:15:19 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Wed, 26 Mar 2014 12:15:19 +0100 Subject: [LAU] Musician who uses sooperlooper Message-ID: <1395832519.1005.22.camel@archlinux> On Wed, 2014-03-26 at 10:43 +0000, Harry van Haaren wrote: > so OsX Likely for good reasons. The Linux driver for my RME card is completely crap, x-runs, ADAT doesn't work correctly. New kernel-rt's lock my machine, the last kernel-rt's I can use are 3.8.13 kernel's, currently it's 3.8.13.14-rt30. That's ok for my private needs, but the absolutely no-go, if you make a living from making music. Linux is for hobbyist enthusiasts only. I'm one ;), but I also made a living by making and mixing music and soundtracks. When doing it, it always was done completely without Linux ... at least without Linux PCs, some stand alone gear using Linux, is something completely different. I'm aware that now many people from this list will disagree. What are they doing, for what companies/schools are they doing the engineering, who will pay the contract penalty when they fail? Jump into the "real" music industry shark tank and use Linux _PCs_! Good luck! Regards, Ralf From alexandre.prokoudine at gmail.com Wed Mar 26 12:25:46 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Wed, 26 Mar 2014 16:25:46 +0400 Subject: [LAU] Bitwig 1.0 is out Message-ID: Hello, audiocrowd :) Bitwig is finally released. Only DEB file (and apparently for i686 only) is available at this time for Linux. https://www.bitwig.com/en/bitwig-studio/download.html Alexandre From falktx at gmail.com Wed Mar 26 12:27:46 2014 From: falktx at gmail.com (Filipe Coelho) Date: Wed, 26 Mar 2014 12:27:46 +0000 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: Message-ID: <5332C7C2.9020607@gmail.com> On 03/26/2014 12:25 PM, Alexandre Prokoudine wrote: > Hello, audiocrowd :) > > Bitwig is finally released. Only DEB file (and apparently for i686 > only) is available at this time for Linux. > > https://www.bitwig.com/en/bitwig-studio/download.html Actually, it's 64bit only. -------------- next part -------------- An HTML attachment was scrubbed... URL: From brunogola at gmail.com Wed Mar 26 18:58:44 2014 From: brunogola at gmail.com (Bruno Gola) Date: Wed, 26 Mar 2014 15:58:44 -0300 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5332C7C2.9020607@gmail.com> References: <5332C7C2.9020607@gmail.com> Message-ID: running it on archlinux (after extracting the .deb with deb2targz) working great :) just got access to the beta as well (can save and export) On Wed, Mar 26, 2014 at 9:27 AM, Filipe Coelho wrote: > On 03/26/2014 12:25 PM, Alexandre Prokoudine wrote: > > Hello, audiocrowd :) > > Bitwig is finally released. Only DEB file (and apparently for i686 > only) is available at this time for Linux. > https://www.bitwig.com/en/bitwig-studio/download.html > > Actually, it's 64bit only. > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -- Bruno Gola http://bgo.la/ | +55 11 9-5552-3599 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at parisson.com Wed Mar 26 21:13:28 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Wed, 26 Mar 2014 22:13:28 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: <5332C7C2.9020607@gmail.com> Message-ID: <533342F8.8010505@parisson.com> On 26/03/2014 19:58, Bruno Gola wrote: > running it on archlinux (after extracting the .deb with deb2targz) > > working great :) > On Debian Wheezy too! For info, I had to install glibc 2.18 from Testing... Congrats to the BitWig team, even it is not open sourced (yet)! ;) Could be cool to have a demo project in the demo package. G From rosea.grammostola at gmail.com Thu Mar 27 09:08:56 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Thu, 27 Mar 2014 10:08:56 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <533342F8.8010505@parisson.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> Message-ID: Looks nice, but I doubt whether it is a all-in-one-packet as people on this list talked about before. I can only find a 7 instruments, handful of plugins and a few 100MB with samples On Wed, Mar 26, 2014 at 10:13 PM, Guillaume Pellerin wrote: > On 26/03/2014 19:58, Bruno Gola wrote: > > running it on archlinux (after extracting the .deb with deb2targz) > > > > working great :) > > > > On Debian Wheezy too! > For info, I had to install glibc 2.18 from Testing... > > Congrats to the BitWig team, even it is not open sourced (yet)! ;) > Could be cool to have a demo project in the demo package. > > G > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlphillips at woh.rr.com Thu Mar 27 12:54:57 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Thu, 27 Mar 2014 08:54:57 -0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> Message-ID: <53341FA1.60001@woh.rr.com> On 03/27/2014 05:08 AM, rosea grammostola wrote: > Looks nice, but I doubt whether it is a all-in-one-packet as people on > this list talked about before. > I can only find a 7 instruments, handful of plugins and a few 100MB > with samples > The instrument count includes five e-drum instruments, a bit bogus sort of list in my opinion. I must say that the other instruments are quite serviceable, but I'll guess that most users will want to deploy their favorite VST/VSTi plugins. I tested its automatic beat matching with five audio samples and a MIDI track. Bitwig performed nicely on my old TurionX2 laptop, with smooth and responsive controls for the synths and effects modules. Btw, the system here is Fedora 19 x86_64, with Planet CCCRMA stuff. The "handful of plugins" includes 25 audio fx, 9 "containers" (racks?), and 11 "modulators, generators, note FX, and routers", all apparently selected for their common utility. The automation controls are cool. I'm just getting into the program, I've looked at Ableton only briefly, so I've everything to learn about using this kind of program. Btw, LMMS has reached its 1.0 milestone, with nicely updated GUI and overall performance. Best, dp From rosea.grammostola at gmail.com Thu Mar 27 13:06:03 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Thu, 27 Mar 2014 14:06:03 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53341FA1.60001@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> Message-ID: I also wonder how much devaluation there is in sound quality. Are these plugins products for the 'mp3-generation' or are they producing the same sound quality as in AMS and Zynaddsubfx? LMMS 1.0, with JACK support I suppose? On Thu, Mar 27, 2014 at 1:54 PM, Dave Phillips wrote: > > On 03/27/2014 05:08 AM, rosea grammostola wrote: > >> Looks nice, but I doubt whether it is a all-in-one-packet as people on >> this list talked about before. >> I can only find a 7 instruments, handful of plugins and a few 100MB with >> samples >> >> > The instrument count includes five e-drum instruments, a bit bogus sort of > list in my opinion. I must say that the other instruments are quite > serviceable, but I'll guess that most users will want to deploy their > favorite VST/VSTi plugins. > > I tested its automatic beat matching with five audio samples and a MIDI > track. Bitwig performed nicely on my old TurionX2 laptop, with smooth and > responsive controls for the synths and effects modules. Btw, the system > here is Fedora 19 x86_64, with Planet CCCRMA stuff. > > The "handful of plugins" includes 25 audio fx, 9 "containers" (racks?), > and 11 "modulators, generators, note FX, and routers", all apparently > selected for their common utility. > > The automation controls are cool. I'm just getting into the program, I've > looked at Ableton only briefly, so I've everything to learn about using > this kind of program. > > Btw, LMMS has reached its 1.0 milestone, with nicely updated GUI and > overall performance. > > Best, > > dp > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlphillips at woh.rr.com Thu Mar 27 13:56:10 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Thu, 27 Mar 2014 09:56:10 -0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53341FA1.60001@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> Message-ID: <53342DFA.2020905@woh.rr.com> On 03/27/2014 08:54 AM, Dave Phillips wrote: > > On 03/27/2014 05:08 AM, rosea grammostola wrote: >> Looks nice, but I doubt whether it is a all-in-one-packet as people >> on this list talked about before. >> I can only find a 7 instruments, handful of plugins and a few 100MB >> with samples >> > > The instrument count includes five e-drum instruments, a bit bogus > sort of list in my opinion. Strike that statement from the record. The list is much longer in the registered version, the demo version's list is a limited subset of what's available. > I must say that the other instruments are quite serviceable, but I'll > guess that most users will want to deploy their favorite VST/VSTi > plugins. > > I tested its automatic beat matching with five audio samples and a > MIDI track. Bitwig performed nicely on my old TurionX2 laptop, with > smooth and responsive controls for the synths and effects modules. > Btw, the system here is Fedora 19 x86_64, with Planet CCCRMA stuff. > > The "handful of plugins" includes 25 audio fx, 9 "containers" > (racks?), and 11 "modulators, generators, note FX, and routers", all > apparently selected for their common utility. > > The automation controls are cool. I'm just getting into the program, > I've looked at Ableton only briefly, so I've everything to learn about > using this kind of program. > To be clear, those comments apply only to the demo version. > Btw, LMMS has reached its 1.0 milestone, with nicely updated GUI and > overall performance. > Yes, with improved JACK support. Still not perfect, but the program overall is much improved. I'm enjoying it. Best, dp From louigi.verona at gmail.com Thu Mar 27 19:41:39 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Thu, 27 Mar 2014 23:41:39 +0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53342DFA.2020905@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> Message-ID: LMMS 1.0? Where? Website says *2013-06-12*: *LMMS 0.4.15 has been released!* Nothing about the new version. On Thu, Mar 27, 2014 at 5:56 PM, Dave Phillips wrote: > > On 03/27/2014 08:54 AM, Dave Phillips wrote: > >> >> On 03/27/2014 05:08 AM, rosea grammostola wrote: >> >>> Looks nice, but I doubt whether it is a all-in-one-packet as people on >>> this list talked about before. >>> I can only find a 7 instruments, handful of plugins and a few 100MB with >>> samples >>> >>> >> The instrument count includes five e-drum instruments, a bit bogus sort >> of list in my opinion. >> > > Strike that statement from the record. The list is much longer in the > registered version, the demo version's list is a limited subset of what's > available. > > > > I must say that the other instruments are quite serviceable, but I'll >> guess that most users will want to deploy their favorite VST/VSTi plugins. >> >> I tested its automatic beat matching with five audio samples and a MIDI >> track. Bitwig performed nicely on my old TurionX2 laptop, with smooth and >> responsive controls for the synths and effects modules. Btw, the system >> here is Fedora 19 x86_64, with Planet CCCRMA stuff. >> >> The "handful of plugins" includes 25 audio fx, 9 "containers" (racks?), >> and 11 "modulators, generators, note FX, and routers", all apparently >> selected for their common utility. >> >> The automation controls are cool. I'm just getting into the program, I've >> looked at Ableton only briefly, so I've everything to learn about using >> this kind of program. >> >> > > To be clear, those comments apply only to the demo version. > > > > Btw, LMMS has reached its 1.0 milestone, with nicely updated GUI and >> overall performance. >> >> > Yes, with improved JACK support. Still not perfect, but the program > overall is much improved. I'm enjoying it. > > > Best, > > dp > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From leoave at gmail.com Thu Mar 27 19:54:55 2014 From: leoave at gmail.com (Leonardo Palomares) Date: Thu, 27 Mar 2014 12:54:55 -0700 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> Message-ID: http://sourceforge.net/projects/lmms/files/lmms/1.0.0/ On Thu, Mar 27, 2014 at 12:41 PM, Louigi Verona wrote: > LMMS 1.0? Where? > > Website says *2013-06-12*: > > *LMMS 0.4.15 has been released!* > Nothing about the new version. > > > On Thu, Mar 27, 2014 at 5:56 PM, Dave Phillips wrote: > >> >> On 03/27/2014 08:54 AM, Dave Phillips wrote: >> >>> >>> On 03/27/2014 05:08 AM, rosea grammostola wrote: >>> >>>> Looks nice, but I doubt whether it is a all-in-one-packet as people on >>>> this list talked about before. >>>> I can only find a 7 instruments, handful of plugins and a few 100MB >>>> with samples >>>> >>>> >>> The instrument count includes five e-drum instruments, a bit bogus sort >>> of list in my opinion. >>> >> >> Strike that statement from the record. The list is much longer in the >> registered version, the demo version's list is a limited subset of what's >> available. >> >> >> >> I must say that the other instruments are quite serviceable, but I'll >>> guess that most users will want to deploy their favorite VST/VSTi plugins. >>> >>> I tested its automatic beat matching with five audio samples and a MIDI >>> track. Bitwig performed nicely on my old TurionX2 laptop, with smooth and >>> responsive controls for the synths and effects modules. Btw, the system >>> here is Fedora 19 x86_64, with Planet CCCRMA stuff. >>> >>> The "handful of plugins" includes 25 audio fx, 9 "containers" (racks?), >>> and 11 "modulators, generators, note FX, and routers", all apparently >>> selected for their common utility. >>> >>> The automation controls are cool. I'm just getting into the program, >>> I've looked at Ableton only briefly, so I've everything to learn about >>> using this kind of program. >>> >>> >> >> To be clear, those comments apply only to the demo version. >> >> >> >> Btw, LMMS has reached its 1.0 milestone, with nicely updated GUI and >>> overall performance. >>> >>> >> Yes, with improved JACK support. Still not perfect, but the program >> overall is much improved. I'm enjoying it. >> >> >> Best, >> >> dp >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> > > > > -- > Louigi Verona > http://www.louigiverona.ru/ > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Thu Mar 27 20:12:39 2014 From: gnome at hawaii.rr.com (david) Date: Thu, 27 Mar 2014 10:12:39 -1000 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> Message-ID: <53348637.8090705@hawaii.rr.com> I guess either they haven't updated the main page of their site, or they don't consider v1.0.0 "released"? On 03/27/2014 09:54 AM, Leonardo Palomares wrote: > http://sourceforge.net/projects/lmms/files/lmms/1.0.0/ > > > On Thu, Mar 27, 2014 at 12:41 PM, Louigi Verona wrote: > > LMMS 1.0? Where? > > Website says /2013-06-12/: *LMMS 0.4.15 has been released! > > * > Nothing about the new version. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From leoave at gmail.com Thu Mar 27 20:35:18 2014 From: leoave at gmail.com (Leonardo Palomares) Date: Thu, 27 Mar 2014 13:35:18 -0700 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53348637.8090705@hawaii.rr.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> <53348637.8090705@hawaii.rr.com> Message-ID: On Thu, Mar 27, 2014 at 1:12 PM, david wrote: > I guess either they haven't updated the main page of their site, or they > don't consider v1.0.0 "released"? > > On 03/27/2014 09:54 AM, Leonardo Palomares wrote: > >> http://sourceforge.net/projects/lmms/files/lmms/1.0.0/ >> >> >> Reading the archived mailing list; Tobias Doerffel wrote that 0.9.92 would be a preparation for 1.0 http://sourceforge.net/p/lmms/mailman/message/32083907/ So, even though there is no announce, I guess 1.0 is final release. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Fri Mar 28 00:43:32 2014 From: gnome at hawaii.rr.com (gnome at hawaii.rr.com) Date: Fri, 28 Mar 2014 0:43:32 +0000 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: Message-ID: <20140328004332.NQONY.114765.root@cdptpa-web27> ---- Leonardo Palomares wrote: > On Thu, Mar 27, 2014 at 1:12 PM, david wrote: > > > I guess either they haven't updated the main page of their site, or they > > don't consider v1.0.0 "released"? > > > > On 03/27/2014 09:54 AM, Leonardo Palomares wrote: > > > >> http://sourceforge.net/projects/lmms/files/lmms/1.0.0/ > >> > >> > >> > Reading the archived mailing list; Tobias Doerffel wrote that 0.9.92 would > be a preparation for 1.0 > http://sourceforge.net/p/lmms/mailman/message/32083907/ > > So, even though there is no announce, I guess 1.0 is final release. Cool. Something nice to celebrate; some FLOSS projects seems to spend forever never reaching the magic "1.0". David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From laurence.capelin at gmail.com Fri Mar 28 07:14:40 2014 From: laurence.capelin at gmail.com (laurence) Date: Fri, 28 Mar 2014 15:14:40 +0800 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <20140328004332.NQONY.114765.root@cdptpa-web27> References: <20140328004332.NQONY.114765.root@cdptpa-web27> Message-ID: Regarding bitwig, it would indeed be cool to be able to deploy one's favourite VSTs... but those are quite often Windows VSTs given the relative dearth of native linux ones. So what options does one have in this regard? vst-bridge? On 28 March 2014 08:43, wrote: > ---- Leonardo Palomares wrote: > > On Thu, Mar 27, 2014 at 1:12 PM, david wrote: > > > > > I guess either they haven't updated the main page of their site, or > they > > > don't consider v1.0.0 "released"? > > > > > > On 03/27/2014 09:54 AM, Leonardo Palomares wrote: > > > > > >> http://sourceforge.net/projects/lmms/files/lmms/1.0.0/ > > >> > > >> > > >> > > Reading the archived mailing list; Tobias Doerffel wrote that 0.9.92 > would > > be a preparation for 1.0 > > http://sourceforge.net/p/lmms/mailman/message/32083907/ > > > > So, even though there is no announce, I guess 1.0 is final release. > > Cool. Something nice to celebrate; some FLOSS projects seems to spend > forever never reaching the magic "1.0". > > David W. Jones > gnome at hawaii.rr.com > authenticity, honesty, community > http://dancingtreefrog.com > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Fri Mar 28 07:17:52 2014 From: atte at youmail.dk (Atte) Date: Fri, 28 Mar 2014 08:17:52 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5332C7C2.9020607@gmail.com> References: <5332C7C2.9020607@gmail.com> Message-ID: <53352220.8070209@youmail.dk> On 03/26/2014 01:27 PM, Filipe Coelho wrote: > Actually, it's 64bit only. Seems it's time to go 64bit (crunchbang). Are there any drawbacks, issues or considerations vs 32bits? -- Atte http://atte.dk http://modlys.dk From ralf.mardorf at rocketmail.com Fri Mar 28 08:09:41 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 28 Mar 2014 09:09:41 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53352220.8070209@youmail.dk> References: <5332C7C2.9020607@gmail.com> <53352220.8070209@youmail.dk> Message-ID: <1395994181.599.18.camel@archlinux> On Fri, 2014-03-28 at 08:17 +0100, Atte wrote: > Are there any drawbacks, issues or considerations vs 32bits? There are not really drawbacks, there are just a few exceptions 32-bit is needed for, but as you can see, there at least is one exception/drawback for 32-bit architecture, you can't run Bitwig. From david at olofson.net Fri Mar 28 08:09:54 2014 From: david at olofson.net (David Olofson) Date: Fri, 28 Mar 2014 09:09:54 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53352220.8070209@youmail.dk> References: <5332C7C2.9020607@gmail.com> <53352220.8070209@youmail.dk> Message-ID: On Fri, Mar 28, 2014 at 8:17 AM, Atte wrote: > On 03/26/2014 01:27 PM, Filipe Coelho wrote: > >> Actually, it's 64bit only. > > > Seems it's time to go 64bit (crunchbang). Probably... Being able to use more than 2-3 GB RAM is nice. If nothing else, Linux uses the "unused" RAM as a lightning fast disk cache. :-) > Are there any drawbacks, issues or considerations vs 32bits? I've been running 64 bit since the early x86_64 days (Athlon 64 and late Pentium 4 CPUs), mostly Gentoo. Back then, there were some applications and the odd library that wouldn't compile to 64 bit, but I haven't had issues with that for years now. 64 bit was tried and tested on Linux long before it went "mainstream," so people have had plenty of time to fix their code. That said, it's nice if the distro can still run 32 bit code out of the box (pretty much standard these days...?), as it's not entirely trivial to set up manually, and one may still run into the odd 32 bit binary. -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------' From lists at parisson.com Fri Mar 28 08:13:05 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Fri, 28 Mar 2014 09:13:05 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53342DFA.2020905@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> Message-ID: <53352F11.6070104@parisson.com> It seems that BitWig infringes the GPL: vamp-aubio is in there whereas the GPL is not mentioned in legal.html, neither Paul Brossier, the author of aubio... http://files.parisson.com/doc/bitwig/legal.html :'( G From blablack at gmail.com Fri Mar 28 09:59:18 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Fri, 28 Mar 2014 09:59:18 +0000 Subject: [LAU] Tutorial : VCOs and Detuning in ams-lv2 and Ingen Message-ID: Hello boys and girls, Today I released this video on how to use VCO detuning in ams-lv2 and Ingen - but these principles applies to synth in general. Hope you find it interresting, and if you have any comments please let me know! http://objectivewave.wordpress.com/2014/03/28/tutorial-vcos-and-detuning/ In addition, if you guys have any ideas for other tutorials using ams-lv2/ingen or ams, please let me know! From alexandre.prokoudine at gmail.com Fri Mar 28 12:32:00 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Fri, 28 Mar 2014 16:32:00 +0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53352F11.6070104@parisson.com> References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> <53352F11.6070104@parisson.com> Message-ID: On Fri, Mar 28, 2014 at 12:13 PM, Guillaume Pellerin wrote: > It seems that BitWig infringes the GPL: vamp-aubio is in there whereas the GPL > is not mentioned in legal.html, neither Paul Brossier, the author of aubio... https://twitter.com/Bitwig/status/449520272123916289 Alexandre From gheskett at wdtv.com Fri Mar 28 12:39:34 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Fri, 28 Mar 2014 08:39:34 -0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53352F11.6070104@parisson.com> References: <53342DFA.2020905@woh.rr.com> <53352F11.6070104@parisson.com> Message-ID: <201403280839.35032.gheskett@wdtv.com> On Friday 28 March 2014 08:34:28 Guillaume Pellerin did opine: > It seems that BitWig infringes the GPL: vamp-aubio is in there whereas > the GPL is not mentioned in legal.html, neither Paul Brossier, the > author of aubio... > > http://files.parisson.com/doc/bitwig/legal.html > > :'( > > G Two files are GPLv3. But the way I read the rest of that, its a minefield, full of bouncing betties for the un-aware. I wouldn't advise downloading it until those restrictions are lifted, every one of them. Otherwise you could find yourself in a legal quagmire. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From simonzwise at gmail.com Fri Mar 28 13:49:23 2014 From: simonzwise at gmail.com (Simon Wise) Date: Sat, 29 Mar 2014 00:49:23 +1100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <201403280839.35032.gheskett@wdtv.com> References: <53342DFA.2020905@woh.rr.com> <53352F11.6070104@parisson.com> <201403280839.35032.gheskett@wdtv.com> Message-ID: <53357DE3.60800@gmail.com> On 28/03/14 23:39, Gene Heskett wrote: > On Friday 28 March 2014 08:34:28 Guillaume Pellerin did opine: > >> It seems that BitWig infringes the GPL: vamp-aubio is in there whereas >> the GPL is not mentioned in legal.html, neither Paul Brossier, the >> author of aubio... >> >> http://files.parisson.com/doc/bitwig/legal.html >> >> :'( >> >> G > > Two files are GPLv3. But the way I read the rest of that, its a minefield, > full of bouncing betties for the un-aware. I wouldn't advise downloading it > until those restrictions are lifted, every one of them. Otherwise you > could find yourself in a legal quagmire. nothing in that document seems wrong ... most of it is BSD style licenses and the appropriate acknowledgements are there, a few are LGPL, which certainly allow use linked to closed code, and there is an offer to deliver source code at cost of delivery. Perhaps including the sources for those parts alongside the deb would have been a bit nicer, but they are libraries distributed for use with closed code and especially if they are simply used as the original binaries then all seems appropriate. None of the authors chose strict copyleft licenses. Simon From lists at parisson.com Fri Mar 28 14:06:34 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Fri, 28 Mar 2014 15:06:34 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: References: <5332C7C2.9020607@gmail.com> <533342F8.8010505@parisson.com> <53341FA1.60001@woh.rr.com> <53342DFA.2020905@woh.rr.com> <53352F11.6070104@parisson.com> Message-ID: <533581EA.5070300@parisson.com> On 28/03/2014 13:32, Alexandre Prokoudine wrote: > On Fri, Mar 28, 2014 at 12:13 PM, Guillaume Pellerin wrote: >> It seems that BitWig infringes the GPL: vamp-aubio is in there whereas the GPL >> is not mentioned in legal.html, neither Paul Brossier, the author of aubio... > > https://twitter.com/Bitwig/status/449520272123916289 > > Alexandre Thank you Alexandre, it is now clearer! G From lists at parisson.com Fri Mar 28 14:14:21 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Fri, 28 Mar 2014 15:14:21 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53357DE3.60800@gmail.com> References: <53342DFA.2020905@woh.rr.com> <53352F11.6070104@parisson.com> <201403280839.35032.gheskett@wdtv.com> <53357DE3.60800@gmail.com> Message-ID: <533583BD.3090204@parisson.com> On 28/03/2014 14:49, Simon Wise wrote: > On 28/03/14 23:39, Gene Heskett wrote: >> On Friday 28 March 2014 08:34:28 Guillaume Pellerin did opine: >> >>> It seems that BitWig infringes the GPL: vamp-aubio is in there whereas >>> the GPL is not mentioned in legal.html, neither Paul Brossier, the >>> author of aubio... >>> >>> http://files.parisson.com/doc/bitwig/legal.html >>> >>> :'( >>> >>> G >> >> Two files are GPLv3. No, only LGPLv3 is mentioned.. But the way I read the rest of that, its a minefield, >> full of bouncing betties for the un-aware. I wouldn't advise downloading it >> until those restrictions are lifted, every one of them. Otherwise you >> could find yourself in a legal quagmire. > > nothing in that document seems wrong ... most of it is BSD style licenses and > the appropriate acknowledgements are there, a few are LGPL, which certainly > allow use linked to closed code, and there is an offer to deliver source code > at cost of delivery. Perhaps including the sources for those parts alongside the > deb would have been a bit nicer, but they are libraries distributed for use with > closed code and especially if they are simply used as the original binaries then > all seems appropriate. None of the authors chose strict copyleft licenses. > Maybe I haven't been clear enough.. The current BitWig package includes vamp-aubio: $ ls /opt/bitwig-studio/bin/vamp-plugins/ transient-detector.so vamp-aubio.so but vamp-aubio embeds libaubio which is GPL and then should be a least mentioned. But as Alexandre got it from the BitWig team on twitter, aubio is not used anymore by the app and will be removed from it. G From zettberlin at linuxuse.de Fri Mar 28 16:35:00 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Fri, 28 Mar 2014 17:35:00 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5332C7C2.9020607@gmail.com> References: <5332C7C2.9020607@gmail.com> Message-ID: <5335A4B4.2040503@linuxuse.de> Am 26.03.2014 13:27, schrieb Filipe Coelho: > On 03/26/2014 12:25 PM, Alexandre Prokoudine wrote: >> Hello, audiocrowd :) >> >> Bitwig is finally released. Only DEB file (and apparently for i686 >> only) is available at this time for Linux. >> >> https://www.bitwig.com/en/bitwig-studio/download.html > Actually, it's 64bit only. I have it running here in Kubuntu 12.10 64, works amazingly stable given the experience from the beta 8 months or so ago... 30 Tracks including 2 synths running 3-4h with even playing Minecraft in the background (must hunt these chicken and make some more arrows to protect my NPCs, there is no alternative... ;-) )and there is not a single xrun with Jack set to 8ms latency.... The only bugger is a bogus error at startup regarding port audio, I only hope, they drop that for good and equip the LInuxversion with a native Jack-connection.... best regards HZN > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From email.rafa at gmail.com Fri Mar 28 17:11:57 2014 From: email.rafa at gmail.com (Rafael Vega) Date: Fri, 28 Mar 2014 12:11:57 -0500 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5335A4B4.2040503@linuxuse.de> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> Message-ID: Portaudio... hmmm.... :( On Fri, Mar 28, 2014 at 11:35 AM, Hartmut Noack wrote: > Am 26.03.2014 13:27, schrieb Filipe Coelho: > > On 03/26/2014 12:25 PM, Alexandre Prokoudine wrote: > >> Hello, audiocrowd :) > >> > >> Bitwig is finally released. Only DEB file (and apparently for i686 > >> only) is available at this time for Linux. > >> > >> https://www.bitwig.com/en/bitwig-studio/download.html > > Actually, it's 64bit only. > > I have it running here in Kubuntu 12.10 64, works amazingly stable given > the experience from the beta 8 months or so ago... 30 Tracks including 2 > synths running 3-4h with even playing Minecraft in the background (must > hunt these chicken and make some more arrows to protect my NPCs, there > is no alternative... ;-) )and there is not a single xrun with Jack set > to 8ms latency.... > > The only bugger is a bogus error at startup regarding port audio, I only > hope, they drop that for good and equip the LInuxversion with a native > Jack-connection.... > > best regards > > HZN > > > > > > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Rafael Vega email.rafa at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Fri Mar 28 21:55:14 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 28 Mar 2014 22:55:14 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <1395994181.599.18.camel@archlinux> References: <5332C7C2.9020607@gmail.com> <53352220.8070209@youmail.dk> <1395994181.599.18.camel@archlinux> Message-ID: <1396043714.579.6.camel@archlinux> On Fri, 2014-03-28 at 09:09 +0100, Ralf Mardorf wrote: > On Fri, 2014-03-28 at 08:17 +0100, Atte wrote: > > Are there any drawbacks, issues or considerations vs 32bits? > > There are not really drawbacks, there are just a few exceptions 32-bit > is needed for, but as you can see, there at least is one > exception/drawback for 32-bit architecture, you can't run Bitwig. PS: I forgot to mention multi-arch ;). A 32-bit chroot always was possible, but nowadays many distros provide multi-arch, IOW 32-bit architecture for 64-bit installs. PPS: On Wed, 2014-03-26 at 15:58 -0300, Bruno Gola wrote: > running it on archlinux (after extracting the .deb with deb2targz) Thanks for the "deb2targz" hint. JFTR it's provided by the AUR, https://aur.archlinux.org/packages/deb2targz/ I build deb2targz yesterday, but I'm not sure, if I want to spend time to test Bitwig. From atte at youmail.dk Sat Mar 29 09:46:37 2014 From: atte at youmail.dk (Atte) Date: Sat, 29 Mar 2014 10:46:37 +0100 Subject: [LAU] wine on 64bit debian Message-ID: <5336967D.60902@youmail.dk> Hi I'm trying to install wine on my fresh 64bit debian. $ sudo apt-get install wine $ wine This gives me a window with the following: ------------------ This is the wine64-bin helper package, which does not provide wine itself, but instead exists solely to provide the following information about enabling multiarch on your system in order to be able to install and run the 32-bit wine packages. The following commands should be issued as root or via sudo in order to enable multiarch (the last command installs 32-bit wine): # dpkg --add-architecture i386 # apt-get update # apt-get install wine-bin:i386 Be very careful as spaces matter above. Note that this package (wine64-bin) will be removed in the process. For more information on the multiarch conversion, see: http://wiki.debian.org/Multiarch/HOWTO --------------------------- Hmm, this sounds a little fishy, I went ahead and ran the proposed: $ sudo apt-get install wine-bin:i386 Reading package lists... Done Building dependency tree Reading state information... Done 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: wine-bin:i386 : Depends: libc6:i386 (>= 2.4) but it is not going to be installed Depends: libwine-bin:i386 (= 1.4.1-4) but it is not going to be installed E: Unable to correct problems, you have held broken packages. As a final step I tried: $ sudo apt-get install wine-bin:i386 libc6:i386 libwine-bin:i386 Reading package lists... Done Building dependency tree Reading state information... Done 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: ant : Depends: default-jre-headless or java2-runtime-headless or java5-runtime-headless or java6-runtime-headless libc6 : Depends: libgcc1 but it is not going to be installed Breaks: libc6:i386 (!= 2.18-4) but 2.13-38+deb7u1 is to be installed libc6:i386 : Breaks: libc6 (!= 2.13-38+deb7u1) but 2.18-4 is to be installed libcairo2 : Depends: libfontconfig1 (>= 2.9.0) but it is not going to be installed Depends: libfreetype6 (>= 2.3.5) but it is not going to be installed Depends: libpixman-1-0 (>= 0.21.6) but it is not going to be installed Depends: libpng12-0 (>= 1.2.13-4) but it is not going to be installed Depends: libxcb-render0 but it is not going to be installed Depends: libxcb-shm0 but it is not going to be installed Depends: libxcb1 (>= 1.6) but it is not going to be installed Depends: libxrender1 but it is not going to be installed libcurl3-gnutls : Depends: libgcrypt11 (>= 1.4.5) but it is not going to be installed Depends: libgnutls26 (>= 2.12.17-0) but it is not going to be installed Depends: libgssapi-krb5-2 (>= 1.10+dfsg~) but it is not going to be installed Depends: libidn11 (>= 1.13) but it is not going to be installed Depends: libldap-2.4-2 (>= 2.4.7) but it is not going to be installed Depends: librtmp0 (>= 2.3) but it is not going to be installed Depends: libssh2-1 (>= 1.2.6) but it is not going to be installed Recommends: ca-certificates but it is not going to be installed libdbus-glib-1-2 : Depends: libdbus-1-3 (>= 1.2.16) but it is not going to be installed libglib2.0-0 : Depends: libffi5 (>= 3.0.4) but it is not going to be installed Depends: libpcre3 (>= 8.10) but it is not going to be installed Depends: libselinux1 (>= 1.32) but it is not going to be installed Recommends: shared-mime-info libimlib2 : Depends: libbz2-1.0 but it is not going to be installed Depends: libfreetype6 (>= 2.2.1) but it is not going to be installed Depends: libgif4 (>= 4.1.4) but it is not going to be installed Depends: libid3tag0 (>= 0.15.1b) but it is not going to be installed Depends: libjpeg8 (>= 8c) but it is not going to be installed Depends: libpng12-0 (>= 1.2.13-4) but it is not going to be installed Depends: libtiff4 (> 3.9.5-3~) but it is not going to be installed liblua5.1-0 : Depends: libgcc1 (>= 1:4.1.1) but it is not going to be installed Depends: libstdc++6 (>= 4.1.1) but it is not going to be installed libwine-bin:i386 : Depends: libwine:i386 (= 1.4.1-4) but it is not going to be installed libx11-6 : Depends: libxcb1 (>= 1.2) but it is not going to be installed libxft2 : Depends: libfontconfig1 (>= 2.9.0) but it is not going to be installed Depends: libfreetype6 (>= 2.3.5) but it is not going to be installed Depends: libxrender1 but it is not going to be installed libxml2 : Depends: liblzma5 (>= 5.1.1alpha+20120614) but it is not going to be installed Recommends: xml-core but it is not going to be installed wine-bin:i386 : Depends: debconf:i386 (>= 0.5) or debconf-2.0:i386 Depends: x11-utils:i386 or xbase-clients:i386 (>= 4.0) but it is not installable or xcontrib:i386 but it is not installable E: Error, pkgProblemResolver::Resolve generated breaks, this may be caused by held packages. So it seems that I'm not really on the right track here, any ideas? Thanks in advance! -- Atte http://atte.dk http://modlys.dk From ralf.mardorf at rocketmail.com Sat Mar 29 10:06:10 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 29 Mar 2014 11:06:10 +0100 Subject: [LAU] wine on 64bit debian In-Reply-To: <5336967D.60902@youmail.dk> References: <5336967D.60902@youmail.dk> Message-ID: <1396087570.1991.6.camel@archlinux> Hi Atte, did you run sudo apt-get install --fix-broken sudo apt-get autoclean sudo apt-get autoremove sudo apt-get -f install ? OTOH "Note that this package (wine64-bin) will be removed in the process." doesn't sound promising, so I would run all apt commands with the --dry-run option first. Did you calculatedly lock a package? JFTR, the right mailing list to send this request is https://lists.debian.org/debian-user/ ;). Regards, Ralf From willgodfrey at musically.me.uk Sat Mar 29 10:08:19 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sat, 29 Mar 2014 10:08:19 +0000 Subject: [LAU] wine on 64bit debian In-Reply-To: <5336967D.60902@youmail.dk> References: <5336967D.60902@youmail.dk> Message-ID: <20140329100819.6717495a@debian> On Sat, 29 Mar 2014 10:46:37 +0100 Atte wrote: > Hi > > I'm trying to install wine on my fresh 64bit debian. > Which version of debian are you working with? I've just installed 64 bit wheezy on a machine and wine came in without problems - that 64 bit install message is quite correct. However before installing wine I did install a laundry list of of build stuff! -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From rosea.grammostola at gmail.com Sat Mar 29 10:15:50 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sat, 29 Mar 2014 11:15:50 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5335A4B4.2040503@linuxuse.de> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> Message-ID: <53369D56.1000704@gmail.com> On 03/28/2014 05:35 PM, Hartmut Noack wrote: > The only bugger is a bogus error at startup regarding port audio, I only > hope, they drop that for good and equip the LInuxversion with a native > Jack-connection.... Price is a bit high maybe for software with portaudio :/ @Atte, how are you experiences so far? Converted yet, to Bitwig? :) From rosea.grammostola at gmail.com Sat Mar 29 10:31:53 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sat, 29 Mar 2014 11:31:53 +0100 Subject: [LAU] Bitwig: what we can learn from it Message-ID: <5336A119.3080502@gmail.com> Hey, I like to open a discussion about Bitwig and what 'we' (users, but mostly developers) can learn from it. I worked just about 30min with Bitwig so far. Just a few thoughts. What is nice to notice is that the actions you try to do as a user is possible pretty often. They seems to know which actions computermusicians tend to perform and they make that possible in an easy way. Automation is very smooth it seems. It's also a nice feeling that you work with a finished product, good chance you can make and finish the project you've in mind. It's handy that it comes with a bunch of samples, but why can't we make a sample pack and make that easy to install on Linux and add it (automatically) to let's say Qtractor? The sound quality of the included instruments seems to be ok, but I doubt whether it is comparable to stuff like AMS/Ingen, zynaddsubfx, pianoteq. Making a good song is still hard, also in a sophisticated application like Bitwig. For very advanced features you probably need Bitwig, but the tools on Linux aren't bad I think. I'm asking myself how big the gap is. To me it looks like the gap itself is not that big, but the last 10% of finishing and polishing an app, makes a huge difference for the end user. The Linuxaudio Floss tools lacks that finishing touch, completeness and level of integration pretty often (which is logical if you look at the manpower). Regards, \r From pshirkey at boosthardware.com Sat Mar 29 11:25:51 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Sat, 29 Mar 2014 22:25:51 +1100 (EST) Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5336A119.3080502@gmail.com> References: <5336A119.3080502@gmail.com> Message-ID: <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> On Sat, March 29, 2014 9:31 pm, rosea.grammostola wrote: > > It's handy that it comes with a bunch of samples, but why can't we make > a sample pack and make that easy to install on Linux and add it > (automatically) to let's say Qtractor? > Sample packs take time to create and most people who can be bothered want to sell their efforts.* There is nothing to stop us from collectively building a complete sample pack and hosting it online. The cost of the server is aroundE50/month which gets 20TB of bandwidth. I am working on such a solution but other things keep getting in the way. If I had more time/money then I could probably have it done in about 160 hours. At this rate I might get it finished by the end of the year. * Conveniently ignoring CCMixter and Freesound as an option of course. -- Patrick Shirkey Boost Hardware Ltd From zettberlin at linuxuse.de Sat Mar 29 12:31:10 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Sat, 29 Mar 2014 13:31:10 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53369D56.1000704@gmail.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> Message-ID: <5336BD0E.5080009@linuxuse.de> Am 29.03.2014 11:15, schrieb rosea.grammostola: > On 03/28/2014 05:35 PM, Hartmut Noack wrote: >> The only bugger is a bogus error at startup regarding port audio, I only >> hope, they drop that for good and equip the LInuxversion with a native >> Jack-connection.... > Price is a bit high maybe for software with portaudio :/ True, Port Audio is meh... Anyway, I got an upgrade today to 1.0.3 and now I do not get any bogus error-reports from PA anymore and Bitwig is connected automatically to Jack's system ports.... Only my USB-Midi Keyboard is not connected, though detected and seen by Bitwig. Bitwig does not create any visible MIDI-Ports in ALSA or Jack MIDI... > > @Atte, how are you experiences so far? Converted yet, to Bitwig? :) > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From zettberlin at linuxuse.de Sat Mar 29 12:57:34 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Sat, 29 Mar 2014 13:57:34 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> References: <5336A119.3080502@gmail.com> <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> Message-ID: <5336C33E.5000302@linuxuse.de> Am 29.03.2014 12:25, schrieb Patrick Shirkey: > > On Sat, March 29, 2014 9:31 pm, rosea.grammostola wrote: >> >> It's handy that it comes with a bunch of samples, but why can't we make >> a sample pack and make that easy to install on Linux and add it >> (automatically) to let's say Qtractor? >> > > Sample packs take time to create and most people who can be bothered want > to sell their efforts.* There is nothing to stop us from collectively > building a complete sample pack and hosting it online. The cost of the > server is aroundE50/month which gets 20TB of bandwidth. > > I am working on such a solution but other things keep getting in the way. Feel free to use any of these: http://lapoc.de/samples/ Another problem in this regard may be, that many people in our realm are individualist, who like to do things, that are not that mainstream to fit the needs of a sample-collection for general use(whatever that may mean). > If I had more time/money then I could probably have it done in about 160 > hours. At this rate I might get it finished by the end of the year. > > > * Conveniently ignoring CCMixter and Freesound as an option of course. > > > > > > -- > Patrick Shirkey > Boost Hardware Ltd > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From itarozzi at gmail.com Sat Mar 29 13:15:21 2014 From: itarozzi at gmail.com (Ivan Tarozzi) Date: Sat, 29 Mar 2014 14:15:21 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> References: <5336A119.3080502@gmail.com> <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> Message-ID: On 29 marzo 2014 12:25:51 CET, Patrick Shirkey wrote: > >On Sat, March 29, 2014 9:31 pm, rosea.grammostola wrote: >> >> It's handy that it comes with a bunch of samples, but why can't we >make >> a sample pack and make that easy to install on Linux and add it >> (automatically) to let's say Qtractor? >> > >Sample packs take time to create and most people who can be bothered >want >to sell their efforts.* There is nothing to stop us from collectively >building a complete sample pack and hosting it online. The cost of the >server is aroundE50/month which gets 20TB of bandwidth. > >I am working on such a solution but other things keep getting in the >way. >If I had more time/money then I could probably have it done in about >160 >hours. At this rate I might get it finished by the end of the year. > > >* Conveniently ignoring CCMixter and Freesound as an option of course. > > > > > >-- >Patrick Shirkey >Boost Hardware Ltd >_______________________________________________ >Linux-audio-user mailing list >Linux-audio-user at lists.linuxaudio.org >http://lists.linuxaudio.org/listinfo/linux-audio-user Instead of include samples in a frees software what about use existing open collections like the mentioned ccmixter (or others) integrating them via api? Solution1 : use web collections via api directly from the music application Solution 2 : create a gateway application that permits to collect, rate, organize my preferred tracks/loops (eventually from several sites) and a standard interface to be used by all other musci software Ivan From ralf.mardorf at rocketmail.com Sat Mar 29 13:19:56 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 29 Mar 2014 14:19:56 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5336C33E.5000302@linuxuse.de> References: <5336A119.3080502@gmail.com> <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> <5336C33E.5000302@linuxuse.de> Message-ID: <1396099196.612.12.camel@archlinux> On Sat, 2014-03-29 at 13:57 +0100, Hartmut Noack wrote: > Another problem in this regard may be, that many people in our realm are > individualist, who like to do things, that are not that mainstream to > fit the needs of a sample-collection for general use(whatever that may > mean). I guess we all would like to have a sample library providing a complete classical orchestra, providing variations how the horns and strings etc. are played + as much ethnic/tribe instruments as possible. So the main problem seemingly is "Sample packs take time to create" + the instruments and people who can play them need to be available + good microphones need to be available. It's not easy to have the time, the instruments, the musicians who are able to play the instruments and good microphones at the same time, to record and create sample libraries. From dlphillips at woh.rr.com Sat Mar 29 13:49:27 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Sat, 29 Mar 2014 09:49:27 -0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5336A119.3080502@gmail.com> References: <5336A119.3080502@gmail.com> Message-ID: <5336CF67.8070705@woh.rr.com> On 03/29/2014 06:31 AM, rosea.grammostola wrote: > > It's handy that it comes with a bunch of samples, but why can't we > make a sample pack and make that easy to install on Linux and add it > (automatically) to let's say Qtractor? > I like Ardour's implementation of the Freesound API, it could use some better search criteria but I've done some nice work with it. Not exactly a sample pack, but it's pretty usable. > The sound quality of the included instruments seems to be ok, but I > doubt whether it is comparable to stuff like AMS/Ingen, zynaddsubfx, > pianoteq. > The demo doesn't include everything available in the purchased package. But I agree that the demo instruments are a little weak. If Bitwig gets its MIDI act together you'll be able to use external synths as well (of course). And there are always VST/VSTi plugins to deploy. > Making a good song is still hard, also in a sophisticated application > like Bitwig. > Agree++ about making a good song. > For very advanced features you probably need Bitwig, but the tools on > Linux aren't bad I think. I'm asking myself how big the gap is. To me > it looks like the gap itself is not that big, but the last 10% of > finishing and polishing an app, makes a huge difference for the end > user. The Linuxaudio Floss tools lacks that finishing touch, > completeness and level of integration pretty often (which is logical > if you look at the manpower). > Well, its automated tempo/beat-matching is something I don't see in any other Linux audio app. I can do some nice time adjustment with Ardour's tools but the job isn't automated. I haven't checked out Bitwig's pitch-shifting yet. Btw, I compared Bitwig's retail list price against the list prices for the latest versions of Cubase, Logic Pro, Ableton Live, and Pro Tools. Only Logic is priced lower than Bitwig, probably because the program has Apple's weight behind it (i.e. it could be a loss-leader for some other purpose). For example, Ableton 9 Complete lists at $749 USD. Overall I'm fascinated by its potential, and I already plan to use it for some work I have in mind. Best, dp From pshirkey at boosthardware.com Sat Mar 29 14:19:34 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Sun, 30 Mar 2014 01:19:34 +1100 (EST) Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <59058.86.107.254.57.1396092351.squirrel@boosthardware.com> Message-ID: <61635.86.107.254.57.1396102774.squirrel@boosthardware.com> On Sun, March 30, 2014 12:15 am, Ivan Tarozzi wrote: > On 29 marzo 2014 12:25:51 CET, Patrick Shirkey > wrote: >> >>On Sat, March 29, 2014 9:31 pm, rosea.grammostola wrote: >>> >>> It's handy that it comes with a bunch of samples, but why can't we >>make >>> a sample pack and make that easy to install on Linux and add it >>> (automatically) to let's say Qtractor? >>> >> >>Sample packs take time to create and most people who can be bothered >>want >>to sell their efforts.* There is nothing to stop us from collectively >>building a complete sample pack and hosting it online. The cost of the >>server is aroundE50/month which gets 20TB of bandwidth. >> >>I am working on such a solution but other things keep getting in the >>way. >>If I had more time/money then I could probably have it done in about >>160 >>hours. At this rate I might get it finished by the end of the year. >> >> >>* Conveniently ignoring CCMixter and Freesound as an option of course. >> >> >> >> >> >>-- >>Patrick Shirkey >>Boost Hardware Ltd >>_______________________________________________ >>Linux-audio-user mailing list >>Linux-audio-user at lists.linuxaudio.org >>http://lists.linuxaudio.org/listinfo/linux-audio-user > > Instead of include samples in a frees software what about use existing > open collections like the mentioned ccmixter (or others) integrating them > via api? > > Solution1 : use web collections via api directly from the music > application > Solution 2 : create a gateway application that permits to collect, rate, > organize my preferred tracks/loops (eventually from several sites) and a > standard interface to be used by all other musci software > My direction is slightly different. I'm thinking of complete midi instrument packs prepackaged (and tested) for the different open source instrument players that can be downloaded directly in app via a public API or individually via a web interface. I might also combine it with other open source multimedia packaging to provide a "one-stop-shop" for all open source multimedia content. Other sites already do similar things in the 3d multimedia/game community but they are also community driven and a bit of a mess so a consolidated system might be useful for some people who participate in those communities too. The site/server/bandwidth might be paid for with advertising or donations but the bandwidth requirements will get expensive quickly so I might have to charge for complete packages. Of course everything would be cc - licensed and of professional quality which means 48khz flac. I'm sure Monty would disagree with that last item so I might also include ogg vorbis too. Integrating with cc-mixter, freesound, blender, etc... is also a good idea. Basically a meta site for all open source multimedia production content with an open public API funded by donations, e-commerce or advertising. I'm sure some of the proprietary folks are getting nervous about that idea though so lets see if they offer me any money to find something else to do first. -- Patrick Shirkey Boost Hardware Ltd From mott at reverberant.com Sat Mar 29 14:37:35 2014 From: mott at reverberant.com (Iain Mott) Date: Sat, 29 Mar 2014 11:37:35 -0300 Subject: [LAU] getting VLC to automatically connect to multiple JACK clients Message-ID: <1396103855.2888.9.camel@espelho> Hi list, It's possible to use the following to get VLC to connect to the system outputs, via Jack, automatically with: vlc --jack-connect-regex system there's also a way of doing this in the preferences. Is there a way though of getting it to connect to multiple clients? Using the preferences method i tried entering two connections (system and jkmeter) with spaces in between the names, with a colon, comers etc. but nothing worked. Any way of doing this? Thanks, iain From rosea.grammostola at gmail.com Sat Mar 29 14:43:05 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sat, 29 Mar 2014 15:43:05 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336BD0E.5080009@linuxuse.de> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> Message-ID: <5336DBF9.9010706@gmail.com> On 03/29/2014 01:31 PM, Hartmut Noack wrote: > and Bitwig is connected automatically to > Jack's system ports.... This is not proper behavior if you can't turn it off From dlphillips at woh.rr.com Sat Mar 29 14:50:33 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Sat, 29 Mar 2014 10:50:33 -0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DBF9.9010706@gmail.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> Message-ID: <5336DDB9.4050105@woh.rr.com> On 03/29/2014 10:43 AM, rosea.grammostola wrote: > On 03/29/2014 01:31 PM, Hartmut Noack wrote: >> and Bitwig is connected automatically to >> Jack's system ports.... > This is not proper behavior if you can't turn it off Unfortunately it's how many (most?) Portaudio clients behave with their JACK connections. Very irritating with Audacity, for instance. Best, dp From simonzwise at gmail.com Sat Mar 29 14:51:43 2014 From: simonzwise at gmail.com (Simon Wise) Date: Sun, 30 Mar 2014 01:51:43 +1100 Subject: [LAU] wine on 64bit debian In-Reply-To: <5336967D.60902@youmail.dk> References: <5336967D.60902@youmail.dk> Message-ID: <5336DDFF.4000603@gmail.com> On 29/03/14 20:46, Atte wrote: > Hi > > I'm trying to install wine on my fresh 64bit debian. > > $ sudo apt-get install wine > $ wine > > This gives me a window with the following: > > ------------------ > This is the wine64-bin helper package, which does not provide wine itself, > but instead exists solely to provide the following information about > enabling multiarch on your system in order to be able to install and run > the 32-bit wine packages. > > The following commands should be issued as root or via sudo in order to > enable multiarch (the last command installs 32-bit wine): > > # dpkg --add-architecture i386 > # apt-get update > # apt-get install wine-bin:i386 > > Be very careful as spaces matter above. Note that this package > (wine64-bin) will be removed in the process. For more information on > the multiarch conversion, see: http://wiki.debian.org/Multiarch/HOWTO > --------------------------- > > Hmm, this sounds a little fishy, I went ahead and ran the proposed: > > > $ sudo apt-get install wine-bin:i386 you did also run the other two commands first? you need to tell apt that you want both architectures, then you need to update its list of available packages or it will not be able to find any packages for i386. Simon From ralf.mardorf at rocketmail.com Sat Mar 29 14:54:13 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 29 Mar 2014 15:54:13 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DBF9.9010706@gmail.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> Message-ID: <1396104853.612.33.camel@archlinux> On Sat, 2014-03-29 at 15:43 +0100, rosea.grammostola wrote: > On 03/29/2014 01:31 PM, Hartmut Noack wrote: > > and Bitwig is connected automatically to Jack's system ports.... > This is not proper behavior if you can't turn it off Assumed "proper behaviour" shouldn't be supported, then perhaps aj-snapshot removing everything, before restoring might be an option ;)?! I don't care about session mangers or what ever apps provide, I still prefer scripts to (re)start audio sessions. From ralf.mardorf at rocketmail.com Sat Mar 29 15:03:28 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 29 Mar 2014 16:03:28 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DDB9.4050105@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> Message-ID: <1396105408.612.38.camel@archlinux> On Sat, 2014-03-29 at 10:50 -0400, Dave Phillips wrote: > Audacity Ok, an Audacity like behaviour even can't be handled by software like aj-snapshot. Or am I'm mistaken? I never used "-d,--daemon Restore ALSA and/or JACK connections until terminated". Jackd usage for Audacity simply is broken, but perhaps aj-snapshot is able to handle this, I never tested it. From zettberlin at linuxuse.de Sat Mar 29 15:11:15 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Sat, 29 Mar 2014 16:11:15 +0100 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DBF9.9010706@gmail.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> Message-ID: <5336E293.8080101@linuxuse.de> Am 29.03.2014 15:43, schrieb rosea.grammostola: > On 03/29/2014 01:31 PM, Hartmut Noack wrote: >> and Bitwig is connected automatically to Jack's system ports.... > This is not proper behavior if you can't turn it off It needs to be turned on actually in Bitwig. This did not work before though, even if you choosed system to be automatically connected, you had to connect the ports by hand. > _______________________________________________ Linux-audio-user > mailing list Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > From fons at linuxaudio.org Sat Mar 29 15:13:28 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 29 Mar 2014 15:13:28 +0000 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DDB9.4050105@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> Message-ID: <20140329151327.GD4434@linuxaudio.org> On Sat, Mar 29, 2014 at 10:50:33AM -0400, Dave Phillips wrote: > Unfortunately it's how many (most?) Portaudio clients behave with > their JACK connections. Very irritating with Audacity, for instance. Or even some apps with a 'native' Jack interface, Pd for example. And it gets it even worse for midi. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From len at ovenwerks.net Sat Mar 29 15:26:04 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 29 Mar 2014 08:26:04 -0700 (PDT) Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DDB9.4050105@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> Message-ID: On Sat, 29 Mar 2014, Dave Phillips wrote: > On 03/29/2014 10:43 AM, rosea.grammostola wrote: >> On 03/29/2014 01:31 PM, Hartmut Noack wrote: >>> and Bitwig is connected automatically to >>> Jack's system ports.... >> This is not proper behavior if you can't turn it off > > Unfortunately it's how many (most?) Portaudio clients behave with their JACK > connections. Very irritating with Audacity, for instance. What I find most irritating about Audacity is that it opens the jack ports at the time the record/play is selected rather than at application startup. This means I have to redo the jack connections every time the record button is clicked rather than being able to set it up once when the application starts, set my levels and then do my work. Ardour is a bit much for just straight recording a stream, mhwaveedit does the jack part right, but lacks some of audacity's plugins etc. (I haven't used audacity for a while, maybe this is fixed?) -- Len Ovens www.ovenwerks.net From alexandre.prokoudine at gmail.com Sat Mar 29 15:42:45 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Sat, 29 Mar 2014 19:42:45 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5336A119.3080502@gmail.com> References: <5336A119.3080502@gmail.com> Message-ID: On Sat, Mar 29, 2014 at 2:31 PM, rosea.grammostola wrote: > Hey, > > I like to open a discussion about Bitwig and what 'we' (users, but mostly > developers) can learn from it. 1) They actually sat down and worked with keyboard vendors. I have a Novation Impulse 61, and while Bitwig only lists the 25 keys model among supported ones, I do get extra handy things like the current track name (taken from instrument/preset name) displayed on the LCD. Haven't checked if Novation already added Bitwig to Automap yet, though. 2) MIDI learn is visualized better than anything I've seen on Linux, and it's more discoverable (right-click menu). 3) After >10 years I'm still uncertain if I can permanently rename I/O for a device in JACK, but I'm sure as hell there is no simple UI for that. Bitwig fixes that at the "first run configuration" step: I can create virtual inputs/outputs from physical ones and give them sensible names. Alexandre From ralf.mardorf at rocketmail.com Sat Mar 29 15:50:57 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 29 Mar 2014 16:50:57 +0100 Subject: [LAU] OT: Bitwig 1.0 is out In-Reply-To: <20140329151327.GD4434@linuxaudio.org> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> <20140329151327.GD4434@linuxaudio.org> Message-ID: <1396108257.612.44.camel@archlinux> On Sat, 2014-03-29 at 15:13 +0000, Fons Adriaensen wrote: > And it gets it even worse for midi. Using several identical cards for MIDI, I e.g. use 2 identical Envy24 cards for MIDI, is a PITA with Linux. [rocketmouse at archlinux ~]$ aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 16:0 RME AIO_579bcc HDSPMx579bcc MIDI 1 20:0 TerraTec EWX24/96 TerraTec EWX24/96 MIDI 24:0 TerraTec EWX24/96 TerraTec EWX24/96 MIDI 28:0 USB Device 0x170b:0x11 USB Device 0x170b:0x11 MIDI 1 32:0 nanoKONTROL nanoKONTROL MIDI 1 MIDI connections for the two TerraTec cards have to be made manually all the times, automatically restore of MIDI connections is/or at least seems to be impossible. From len at ovenwerks.net Sat Mar 29 16:11:23 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 29 Mar 2014 09:11:23 -0700 (PDT) Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> Message-ID: On Sat, 29 Mar 2014, Alexandre Prokoudine wrote: > 3) After >10 years I'm still uncertain if I can permanently rename I/O > for a device in JACK, but I'm sure as hell there is no simple UI for > that. Bitwig fixes that at the "first run configuration" step: I can > create virtual inputs/outputs from physical ones and give them > sensible names. This should I think be a part of jack. I have a delta 66 which means capture 1 to 4 are analog in, capture 9 and 10 are spdif in (analog 5 and 6 for me) and capture 11 and 12 are monitor mixer in. Capture 5 to 8 are not usable and it would be nice to not show them at all, but it would be nice to at least label them "NA" or "not in use" or something. Playback has similar issues though I can at least use all the playbacks as inputs to the monitor mixer... if I actually made use of it :) But labeling (which affects ordering in displays BTW) would be really nice. I could make a fake device with 6 inputs and 6 outputs which would be next best... -- Len Ovens www.ovenwerks.net From fons at linuxaudio.org Sat Mar 29 16:29:26 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 29 Mar 2014 16:29:26 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> Message-ID: <20140329162926.GE4434@linuxaudio.org> On Sat, Mar 29, 2014 at 09:11:23AM -0700, Len Ovens wrote: > > On Sat, 29 Mar 2014, Alexandre Prokoudine wrote: > > >3) After >10 years I'm still uncertain if I can permanently rename I/O > >for a device in JACK, but I'm sure as hell there is no simple UI for > >that. Bitwig fixes that at the "first run configuration" step: I can > >create virtual inputs/outputs from physical ones and give them > >sensible names. > > This should I think be a part of jack. I have a delta 66 which means > capture 1 to 4 are analog in, capture 9 and 10 are spdif in (analog > 5 and 6 for me) and capture 11 and 12 are monitor mixer in. Capture > 5 to 8 are not usable and it would be nice to not show them at all, > but it would be nice to at least label them "NA" or "not in use" or > something. Playback has similar issues though I can at least use all > the playbacks as inputs to the monitor mixer... if I actually made > use of it :) But labeling (which affects ordering in displays BTW) > would be really nice. I could make a fake device with 6 inputs and 6 > outputs which would be next best... With latest Jack1 you can add any metadata to a port. Of course then the connection GUIs such as qjackctl should support it... Apart from that it would indeed be handy if the port names for a device could be defined somewhere, e.g. in a .jackrc file. Can't be that difficult... Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rosea.grammostola at gmail.com Sat Mar 29 21:17:31 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sat, 29 Mar 2014 22:17:31 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140329162926.GE4434@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> Message-ID: <5337386B.5070902@gmail.com> I like the way you can change views. From Arrange to Edit view. This works better for me then Qtractor for example, which opens a new window when you edit Midi events. It seems to have quite some options to help you to make a (complex) rhythm pattern. It seems to make you possible to edit every single parameter on the grid The GUI looks and feels nice and fast. From simonzwise at gmail.com Sun Mar 30 02:52:10 2014 From: simonzwise at gmail.com (Simon Wise) Date: Sun, 30 Mar 2014 13:52:10 +1100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5337386B.5070902@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> Message-ID: <533786DA.4020208@gmail.com> On 30/03/14 08:17, rosea.grammostola wrote: > I like the way you can change views. From Arrange to Edit view. This works > better for me then Qtractor for example, which opens a new window when you edit > Midi events. > > It seems to have quite some options to help you to make a (complex) rhythm pattern. > > It seems to make you possible to edit every single parameter on the grid > > The GUI looks and feels nice and fast. for a different, older, work-flow try this ... http://www.youtube.com/watch?v=M2ORkIrHUbg Simon From louigi.verona at gmail.com Sun Mar 30 10:27:35 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sun, 30 Mar 2014 14:27:35 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533786DA.4020208@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> Message-ID: What we can learn from Bitwig is that they base their work on musician's needs. And their whole application is tailored towards a musician getting his work done easier and more efficiently. In Linux Audio very often the basis is a curious technical idea that might have little to do with doing music. As a made-up example "why not create a framework that will have all the midi connections in one place and it will dynamically reassign those connections and plug them using my new format that everyone will have to adapt because it is such a great and efficient format". This is a strictly technical passion. Commercial projects tend to figure out what their users actually want. Even right now in this thread I see people suggesting many cool technical feats, but I see little interest in trying to understand what musicians might want. As I usually write, sometimes getting heated metaphors back at myself for that, often a musician needs some basic stuff first. I spoke about no Linux sampler supporting WAVE loops, although all Windows DAWs do. Or that no sf2 player has volume envelope, although most non-Linux sf2 players do. And the reason for this is because people are doing software for themselves and not necessarily for others. This is not good or bad, this is just how it is. You decide whether you want to change this or not. Louigi. http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sun Mar 30 11:00:55 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 13:00:55 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> Message-ID: <1396177255.587.3.camel@archlinux> On Sun, 2014-03-30 at 14:27 +0400, Louigi Verona wrote: > This is not good or bad, this is just how it is. Full ACK. That's why my claim is that Linux is for hobby enthusiasts. Linux audio is good, but it ships with many drawbacks. We could use Linux even for professional work, but it definitively is more time consuming to use Linux and time is money. Updating a Linux DAW and keeping it stable is another issue. This can be done by power users, but not by an averaged musician. From info at linuxdsp.co.uk Sun Mar 30 12:04:25 2014 From: info at linuxdsp.co.uk (linuxDSP) Date: Sun, 30 Mar 2014 13:04:25 +0100 Subject: [LAU] linuxDSP releases DYN4000 - LV2 and VST channel dynamics plug-in for linux Message-ID: <53380849.1010106@linuxdsp.co.uk> *linux audio users* linuxDSP releases DYN4000 - LV2 and VST channel dynamics plug-in for linux. Modelled on channel dynamics processing from one of the most successful and best-known British recording consoles, by former engineers from the company which made those consoles, the linuxDSP DYN4000 plug-in comprises compressor / limiter and expander / gate sections with soft-knee compression and gate threshold hysteresis. The compressor includes switchable fast attack, variable programme dependent release and auto make-up gain. Ardour 2 - 3 compatible, mono / stereo LV2 and linuxVST Find out more at http://www.linuxdsp.co.uk -- linuxDSP is a division of Applied Computer Music Technologies Ltd, a company registered in the UK. Company Number: 8499906 From linux at alextone.info Sun Mar 30 12:21:18 2014 From: linux at alextone.info (Alex) Date: Sun, 30 Mar 2014 14:21:18 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> Message-ID: <53380C3E.5040808@alextone.info> On 03/30/2014 12:27 PM, Louigi Verona wrote: > What we can learn from Bitwig is that they base their work on > musician's needs. And their whole application is tailored towards a > musician getting his work done easier and more efficiently. > > In Linux Audio very often the basis is a curious technical idea that > might have little to do with doing music. As a made-up example "why > not create a framework that will have all the midi connections in one > place and it will dynamically reassign those connections and plug them > using my new format that everyone will have to adapt because it is > such a great and efficient format". > This is a strictly technical passion. Commercial projects tend to > figure out what their users actually want. > > > Even right now in this thread I see people suggesting many cool > technical feats, but I see little interest in trying to understand > what musicians might want. As I usually write, sometimes getting > heated metaphors back at myself for that, often a musician needs some > basic stuff first. > > I spoke about no Linux sampler supporting WAVE loops, although all > Windows DAWs do. > Or that no sf2 player has volume envelope, although most non-Linux sf2 > players do. > > And the reason for this is because people are doing software for > themselves and not necessarily for others. This is not good or bad, > this is just how it is. You decide whether you want to change this or not. > > Louigi. > http://www.louigiverona.ru/ > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I add my now self-enforced once a year enthusiastic mention for workflow in LA. I don't necessarily agree with the notion that LA is for only hobby enthusiasts, as my nearly ten years in LA trying to piece together a professional setup took a step forward with the advent of Non-session-manager, and a couple of other breakthroughs in that time, and for the moment at least, i feel i'm "almost there." Putting aside the obvious difference between commercial and opensource DAWs in terms of philosophy, i'm with Louigi on the user/musician's workflow front. For all their faults, and there are many, made worse by the upgrade extortion commercial apps enact for what are usually bugfixes, most of them do have an eye on their users in terms of workflow, as it will directly affect their bottom line if they don't. Louigi's right, it's just how it is, no finger pointing, or accusations of incompetence implied. I will add there are exceptions in our community, and i will wave the flag for Fons, Filipe, John, Robin, and Christian, who have, when i'm interacting with them about their apps (those apps i use), kept an open mind about workflow, enthusiastically, or at least respectfully. My corner use case has elicited little enthusiasm at times, but it's these generous and decent fellows that kept my linux enthusiasm going, when i got so frustrated with the amount of work required to achieve a simple task that i considered going back to the "dark side". (and that decision is still on the table, being honest with myself) I've learned to distinguish between more academic projects, and those with an intent on wider user involvement at the musical level. (No doubt to the relief of LA devs in general) I no longer consider my unique user case relevant in LA, however i've managed to put something together that is at least usable, and with a modicum of workflow efficiency, as clumsy as that may be. Bitwig misses the mark for me, on Linux, in a couple of spots. But the basic workflow is "mainstream", fairly efficient, and they've obviously worked out the path from A to B to C in terms of navigation, and sequence of as few actions as possible to achieve a task, to the general users benefit. (At least in the 2 hours i spent trying it out) My 2 Euros worth, and now it's back in the box until the next workflow mention, sometime in 2015. Regards to all, and thanks again to those devs i mentioned. You're keeping the flame alive. Alex. From ralf.mardorf at rocketmail.com Sun Mar 30 12:29:27 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 14:29:27 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53380C3E.5040808@alextone.info> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <53380C3E.5040808@alextone.info> Message-ID: <1396182567.995.9.camel@archlinux> On Sun, 2014-03-30 at 14:21 +0200, Alex wrote: > I will add there are exceptions in our community, and i will wave the > flag for Fons, Filipe, John, Robin, and Christian Rui, Harry and many other developers care a lot about user feedback too :). That's the good thing when using Linux audio. The chance to talk to the developers of proprietary software is virtually zero. From louigi.verona at gmail.com Sun Mar 30 12:33:02 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sun, 30 Mar 2014 16:33:02 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396182567.995.9.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <53380C3E.5040808@alextone.info> <1396182567.995.9.camel@archlinux> Message-ID: "The chance to talk to the developers of proprietary software is virtually zero." Not directly, but you do "talk" to them by being their customer. And also in case of some DAWs, like FLStudio, you do have the chance to talk to them. They have a forum for that and AFAIK they do respond. On Sun, Mar 30, 2014 at 4:29 PM, Ralf Mardorf wrote: > On Sun, 2014-03-30 at 14:21 +0200, Alex wrote: > > I will add there are exceptions in our community, and i will wave the > > flag for Fons, Filipe, John, Robin, and Christian > > Rui, Harry and many other developers care a lot about user feedback > too :). That's the good thing when using Linux audio. The chance to talk > to the developers of proprietary software is virtually zero. > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sun Mar 30 12:41:54 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 14:41:54 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <53380C3E.5040808@alextone.info> <1396182567.995.9.camel@archlinux> Message-ID: <1396183314.995.15.camel@archlinux> On Sun, 2014-03-30 at 16:33 +0400, Louigi Verona wrote: > "The chance to talk to the developers of proprietary software is > virtually zero." > > > Not directly, but you do "talk" to them by being their customer. And > also in case of > > some DAWs, like FLStudio, you do have the chance to talk to them. They > have a forum > > for that and AFAIK they do respond. Sure, it's not just black and white ;). I made good experiences with several hardware vendors. They were interested in the issues I experience(d) with my hardware, when using it with Linux. E.g. Western Digital and RME seriously were interested what issues I experience and they were interested to help me, if possible. However, a lot of companies for proprietary software and/or hardware ignore customers. From tim at klingt.org Sun Mar 30 12:43:24 2014 From: tim at klingt.org (Tim Blechmann) Date: Sun, 30 Mar 2014 14:43:24 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396182567.995.9.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <53380C3E.5040808@alextone.info> <1396182567.995.9.camel@archlinux> Message-ID: >> I will add there are exceptions in our community, and i will wave the >> flag for Fons, Filipe, John, Robin, and Christian > > Rui, Harry and many other developers care a lot about user feedback > too :). That's the good thing when using Linux audio. The chance to talk > to the developers of proprietary software is virtually zero. in this regard, you may want to distinguish between a company of 50000 people like yamaha (steinberg), 3000 people like avid (protools) and companies like bitwig or cockos (each having about 3 developers). tim From ralf.mardorf at rocketmail.com Sun Mar 30 12:53:29 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 14:53:29 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <53380C3E.5040808@alextone.info> <1396182567.995.9.camel@archlinux> Message-ID: <1396184009.995.21.camel@archlinux> On Sun, 2014-03-30 at 14:43 +0200, Tim Blechmann wrote: > >> I will add there are exceptions in our community, and i will wave the > >> flag for Fons, Filipe, John, Robin, and Christian > > > > Rui, Harry and many other developers care a lot about user feedback > > too :). That's the good thing when using Linux audio. The chance to talk > > to the developers of proprietary software is virtually zero. > > in this regard, you may want to distinguish between a company of 50000 > people like yamaha (steinberg), 3000 people like avid (protools) and > companies like bitwig or cockos (each having about 3 developers). Ok, true and while some companies don't care about the averaged customer, they care if you call them when working for another company. When I worked for Brauner I could call some companies and they listened to me, but they won't listen to me, when I call them privately. That's not bad at all, since if you call company B while working for company A the averaged user will benefit too. But if you talk to companies there always is a communication chain, while for Linux audio we usually can directly get in contact with the developers, that's a nice advantage. From rosea.grammostola at gmail.com Sun Mar 30 13:10:51 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sun, 30 Mar 2014 15:10:51 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> Message-ID: <533817DB.9090503@gmail.com> On 03/30/2014 12:27 PM, Louigi Verona wrote: > What we can learn from Bitwig is that they base their work on > musician's needs. And their whole application is tailored towards a > musician getting his work done easier and more efficiently. > > In Linux Audio very often the basis is a curious technical idea that > might have little to do with doing music. As a made-up example "why > not create a framework that will have all the midi connections in one > place and it will dynamically reassign those connections and plug them > using my new format that everyone will have to adapt because it is > such a great and efficient format". > This is a strictly technical passion. Commercial projects tend to > figure out what their users actually want. > > > Even right now in this thread I see people suggesting many cool > technical feats, but I see little interest in trying to understand > what musicians might want. As I usually write, sometimes getting > heated metaphors back at myself for that, often a musician needs some > basic stuff first. > > I spoke about no Linux sampler supporting WAVE loops, although all > Windows DAWs do. > Or that no sf2 player has volume envelope, although most non-Linux sf2 > players do. > > And the reason for this is because people are doing software for > themselves and not necessarily for others. This is not good or bad, > this is just how it is. You decide whether you want to change this or not. It's true, the attention to workflow and needs of musicians of Bitwig is impressive. I'm sure I'll be aware even more of it's possibilities if someone shows me what you can do in Bitwig exactly. On the other hand, looking at Bitwig I'm not sure whether I should be more impressed by Biwtig or by the achievements of the linuxuadio community, of what is possible with Floss Linuxaudio already. And yes impressed also by the technical infrastructure. I mean, what you get from Bitwig is portaudio, that's almost a shame for a 300 euro app from a linuxaudio-user pov. Also it lacks OSC, LADSPA, LV2 and NSM support. Stuff you'll find in Ardour for instance. Also the quality in sound(instruments/samples) they give you for your 300 euro, is not impressive for me, don't get fooled. But this seems to be cultural thing, the acceptance of low quality sound. My conclusion so far is that Bitwig gives you what Linuxaudio lack too often, smooth workflow and 'completeness' of features. This is a major thing for people who want to make music! On the other hand, apart from very sophisticated features for making beats etc., a lot could be possible with Linuxaudio tools or is already possible today. The challenge today is to make Linuxaudio tools more friendly and complete for musicians and integrate them better with each other. With metadata in JACK, NSM, OSC it should be possible to improve this more and more. It would be nice to launch Non-Timeline in a NSM session with Carla and control Carla by Non-Timeline via OSC for instance. \r From robin at gareus.org Sun Mar 30 13:47:58 2014 From: robin at gareus.org (Robin Gareus) Date: Sun, 30 Mar 2014 15:47:58 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396177255.587.3.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> Message-ID: <5338208E.8030204@gareus.org> Please excuse this somewhat off-topic rant, On 03/30/2014 01:00 PM, Ralf Mardorf wrote: > We could use Linux even for professional work, There are many *do* use Linux for professional audio work and A/V post-production. Ever heard of Pixar? It's also not uncommon in various huge post-prod houses, heck even the NASA control center uses Ardour [1]. They're certainly not hobbyists nor amateurs. Then there's the Miraverse [2] and the list goes on and on. Some vendors use embedded linux inside instruments. Just because there's no Linux on the label does not mean it's not used. Opening the field a bit more to GNU/Linux Multimedia: blender.org is also doing a fanatic job. There are likely more non-professional (not making a living with) users of Linux Audio but that does not imply that Linux Audio is not professional or can only be used by some elite for professional work. > but it definitively is more time consuming to use Don't confuse setup with use. The big strength of Linux is the possibility to customize it. You can tailor it for a given workflow which indeed requires investing (time|money). But once it is set up, it can be very efficient to *use*. I think it's not a professional vs hobby distinction but rather style (and requirements) dependent. > Updating a Linux DAW and keeping it stable is another issue. You just don't. There's no need for any casual musician to do that. Just like no few musician update the firmware of his/her Yamaha keyboard and most guitarists have a roadie to replace the strings of their guitars and tune them... Tweaking a distro today is no different than a 70's guitarist replacing the pickups of his guitar or changing the tubes or soldering some wah-pedal tweaks. It's also not Linux specific: getting ASIO to work, installing re-wire or whatnot on other OSses is similar. If you're not interested in this, just use one of the excellent ready-to-run Linux-Studio distribution and be done with it. 2c, robin [1] http://richwielgosz.com/2011/06/nasa-and-jpl-use-the-ardour-daw.html [2] http://manifoldrecording.com/studio/control-room.php From ralf.mardorf at rocketmail.com Sun Mar 30 14:05:40 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 16:05:40 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5338208E.8030204@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> Message-ID: <1396188340.995.46.camel@archlinux> Hi Robin :) Pixar makes audio productions using Linux? The NASA is using Ardour to record communication, not to produce MIDI/hard disk recording music ;). Replacing a pickup for your Stratocaster doesn't need that much knowhow as setting up a Linux DAW and btw. if I wouldn't set up the Linux DAW by myself, but use a Linux audio distro, Linux audio wouldn't work on my machine. Not upgrading a Linux audio DAW is stupid, when you're missing features new versions of software will provide, not to mention that for good reasons I prefer Arch over Debian. I'm using both, but the rolling release Arch is much more stable, than Debian, since for audio Debian stable IMHO is a no-go you need to mix stable, testing and unstable. YMMV, but you can't ignore that _all_ famous and even most hobby studios don't use Linux, but they use proprietary Apple or Windows solutions instead. A college and some other institutions, homerecording folks, have some luxury the averaged musician, engineer who needs to make a living from audio production, doesn't have. 2 Cents, Ralf From ralf.mardorf at rocketmail.com Sun Mar 30 14:22:56 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 16:22:56 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5338208E.8030204@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> Message-ID: <1396189376.995.50.camel@archlinux> PS: Robin, I guess you remember that you once made a remix and sent it to LAU, claiming you didn't have enough time to find out how to mix the bass correctly ;). I sometimes need days to do a Linux mix that is less good, than doing a mix in 5 minutes on a Neve. From robin at gareus.org Sun Mar 30 14:44:38 2014 From: robin at gareus.org (Robin Gareus) Date: Sun, 30 Mar 2014 16:44:38 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396189376.995.50.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> Message-ID: <53382DD6.30602@gareus.org> On 03/30/2014 04:22 PM, Ralf Mardorf wrote: > PS: Robin, I guess you remember that you once made a remix and sent it > to LAU, claiming you didn't have enough time to find out how to mix the > bass correctly ;). I sometimes need days to do a Linux mix that is less > good, than doing a mix in 5 minutes on a Neve. > Ralf, I'm sorry, but you keep missing the point, besides I said neither of this. For reference: http://linuxaudio.org/mailarchive/lau/2012/7/20/191557 Mixing a (almost) solo double-bass is not a problem of tools. It's my lack of experience at mixing+mastering this kind of sound. In retrospect I would not have done any better using ProTools or Bitwig or even a Neve Desk. ciao, robin From cbannister at slingshot.co.nz Sun Mar 30 14:49:59 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Mon, 31 Mar 2014 03:49:59 +1300 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396188340.995.46.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396188340.995.46.camel@archlinux> Message-ID: <20140330144958.GA32457@tal> On Sun, Mar 30, 2014 at 04:05:40PM +0200, Ralf Mardorf wrote: > features new versions of software will provide, not to mention that for > good reasons I prefer Arch over Debian. I'm using both, but the rolling > release Arch is much more stable, than Debian ... A rolling release by its very nature is *not* 'stable'. In the Debian sense, stable does not mean unlikely to crash (although this is a pleasant side effect) but that it will stay the same on a daily basis while still keeping the system updated with security patches. It is my understanding that a rolling release is, by its nature, the complete opposite. -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X From ralf.mardorf at rocketmail.com Sun Mar 30 14:54:04 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 16:54:04 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53382DD6.30602@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> Message-ID: <1396191244.995.58.camel@archlinux> On Sun, 2014-03-30 at 16:44 +0200, Robin Gareus wrote: > On 03/30/2014 04:22 PM, Ralf Mardorf wrote: > > PS: Robin, I guess you remember that you once made a remix and sent it > > to LAU, claiming you didn't have enough time to find out how to mix the > > bass correctly ;). I sometimes need days to do a Linux mix that is less > > good, than doing a mix in 5 minutes on a Neve. > > > > Ralf, > > I'm sorry, but you keep missing the point, besides I said neither of > this. For reference: http://linuxaudio.org/mailarchive/lau/2012/7/20/191557 > > Mixing a (almost) solo double-bass is not a problem of tools. It's my > lack of experience at mixing+mastering this kind of sound. In retrospect > I would not have done any better using ProTools or Bitwig or even a Neve > Desk. My apologies, however, I'm sure using a professional analog mixing console or even a middle class Soundcraft you would have been able to do a better mix, by needing less time to do it. From ralf.mardorf at rocketmail.com Sun Mar 30 15:06:33 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 17:06:33 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53382DD6.30602@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> Message-ID: <1396191993.995.64.camel@archlinux> A last note, before I go offline: It already takes time to find a usable EQ plugin when e.g. using Qtractor or Ardour. Proprietary DAWs usually provide relatively usable EQs by default. So IMO it might be a good idea to add by default that kind of EQ to the Linux DAW's mixers, we will find on good analog mixing consoles, resp. something similar. From ralf.mardorf at rocketmail.com Sun Mar 30 15:10:51 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 17:10:51 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396191993.995.64.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> <1396191993.995.64.camel@archlinux> Message-ID: <1396192251.995.66.camel@archlinux> On Sun, 2014-03-30 at 17:06 +0200, Ralf Mardorf wrote: > A last note, before I go offline: > > It already takes time to find a usable EQ plugin when e.g. using > Qtractor or Ardour. Proprietary DAWs usually provide relatively usable > EQs by default. > So IMO it might be a good idea to add by default that kind of EQ to the > Linux DAW's mixers, we will find on good analog mixing consoles, resp. > something similar. PS: Assumed you are aware about a EQ you like, you still need time to add it. An EQ should be something that is a default feature, even for a virtual mixing console. From len at ovenwerks.net Sun Mar 30 16:19:52 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 30 Mar 2014 09:19:52 -0700 (PDT) Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396191993.995.64.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> <1396191993.995.64.camel@archlinux> Message-ID: On Sun, 30 Mar 2014, Ralf Mardorf wrote: > It already takes time to find a usable EQ plugin when e.g. using > Qtractor or Ardour. Proprietary DAWs usually provide relatively usable > EQs by default. > So IMO it might be a good idea to add by default that kind of EQ to the > Linux DAW's mixers, we will find on good analog mixing consoles, resp. > something similar. You mean like mixbus? Not free, I know, but it is there. -- Len Ovens www.ovenwerks.net From rosea.grammostola at gmail.com Sun Mar 30 16:42:04 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sun, 30 Mar 2014 18:42:04 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5338208E.8030204@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> Message-ID: <5338495C.30902@gmail.com> On 03/30/2014 03:47 PM, Robin Gareus wrote: > [1]http://richwielgosz.com/2011/06/nasa-and-jpl-use-the-ardour-daw.html Nice, would even be better if there are pictures showing serious (electronic) musicians using Floss linuxaudio music production tools. :) That's part of the debate in this topic, if linuxaudio floss tools are good enough for musicians/music production. Regards, \r From jk at jasonkahn.net Sun Mar 30 16:53:56 2014 From: jk at jasonkahn.net (jk at jasonkahn.net) Date: Sun, 30 Mar 2014 18:53:56 +0200 Subject: [LAU] Linux-audio-user Digest, Vol 85, Issue 16 In-Reply-To: References: Message-ID: <53384C24.4050109@jasonkahn.net> > Message: 4 > Date: Sun, 30 Mar 2014 13:00:55 +0200 > From: Ralf Mardorf > To: linux-audio-user at lists.linuxaudio.org > Subject: Re: [LAU] Bitwig: what we can learn from it > Message-ID: <1396177255.587.3.camel at archlinux> > Content-Type: text/plain; charset="UTF-8" > > > time is money. how pathetic From ralf.mardorf at rocketmail.com Sun Mar 30 16:54:02 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 18:54:02 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> <1396191993.995.64.camel@archlinux> Message-ID: <1396198442.578.15.camel@archlinux> On Sun, 2014-03-30 at 09:19 -0700, Len Ovens wrote: > On Sun, 30 Mar 2014, Ralf Mardorf wrote: > > > It already takes time to find a usable EQ plugin when e.g. using > > Qtractor or Ardour. Proprietary DAWs usually provide relatively usable > > EQs by default. > > So IMO it might be a good idea to add by default that kind of EQ to the > > Linux DAW's mixers, we will find on good analog mixing consoles, resp. > > something similar. > > You mean like mixbus? Not free, I know, but it is there. Mixbus, EnergyXT and perhaps Bitwig might or might not provide it. Assumed I will go off-line within the next minutes, I'll likely have time to test Bitwig tomorrow. They aren't free as in beer, so I would call them "proprietary" DAWs. It's ok to pay for a Linux DAW, especially if people should use it for professional work. For my home recording I prefer to use legal gratis software. If we look out what other home recording folks use, we often will see them using cracked software for other operating systems. Btw. we could use templates for Qtractor, Ardour and other Linux DAWs that provide tracks with EQs, I don't claim that using "free" Linux software doesn't provide a smoothly work-flow, but it takes some preparation and can't be done for all the musician's needs. Using Linux means to care about many things and it's not always easy. From ralf.mardorf at rocketmail.com Sun Mar 30 17:00:29 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 30 Mar 2014 19:00:29 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53384C24.4050109@jasonkahn.net> References: <53384C24.4050109@jasonkahn.net> Message-ID: <1396198829.578.18.camel@archlinux> On Sun, 2014-03-30 at 18:53 +0200, jk at jasonkahn.net wrote: > > Message: 4 > > Date: Sun, 30 Mar 2014 13:00:55 +0200 > > From: Ralf Mardorf > > To: linux-audio-user at lists.linuxaudio.org > > Subject: Re: [LAU] Bitwig: what we can learn from it > > Message-ID: <1396177255.587.3.camel at archlinux> > > Content-Type: text/plain; charset="UTF-8" > > > > > > time is money. > > how pathetic Perhaps a little bit polemic, but pathos, emotionalism is something musicians tend to have a lot ;). From jeremy at autostatic.com Sun Mar 30 17:56:18 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sun, 30 Mar 2014 19:56:18 +0200 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <5336DDB9.4050105@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> Message-ID: <53385AC2.1040708@autostatic.com> On 03/29/2014 03:50 PM, Dave Phillips wrote: > > On 03/29/2014 10:43 AM, rosea.grammostola wrote: >> On 03/29/2014 01:31 PM, Hartmut Noack wrote: >>> and Bitwig is connected automatically to >>> Jack's system ports.... >> This is not proper behavior if you can't turn it off > > Unfortunately it's how many (most?) Portaudio clients behave with their > JACK connections. Very irritating with Audacity, for instance. > > Best, > > dp You can disable this, in Audacity autoconnecting is done here: http://code.google.com/p/audacity/source/browse/audacity-src/trunk/lib-src/portaudio-v19/src/hostapi/jack/pa_jack.c And then at line 1771. Iirc uncommenting the two if stanzas should do the trick. At line 88 you can change the client name btw, could be handy too. Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From rdxesy at yahoo.de Sun Mar 30 18:10:24 2014 From: rdxesy at yahoo.de (Janus) Date: Sun, 30 Mar 2014 20:10:24 +0200 Subject: [LAU] prodatum v2.0.0 released Message-ID: <53385E10.8050102@yahoo.de> prodatum is an EMU Proteus 2000 patch editor for Linux/Mac/Windows. v2 no longer depends on a patched Fl_File_Chooser and links fine with existing FLTK 1.1 or 1.3 installations (for the looks I recommend linking statically with a scheme-patched FLTk version though, patch is in the source tree). If you own one of those old E-Mu synths, give it a try. I'm sure you'll like it. For more information, binaries and source visit http://prodatum.sf.net From jeremy at autostatic.com Sun Mar 30 18:38:04 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sun, 30 Mar 2014 20:38:04 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite Message-ID: <5338648C.6020405@autostatic.com> Dear all, Finally got around finishing a new track. And it's just 65BPM so no four to the floor this time. http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg This song is about one of the most beautiful places I know on this planet, Cala Del Aceite in the most southern part of Spain (http://www.conilplaya.com/fotos/playasdeconil/caladelaceite/playaCaladelAceiteConil.htm). Tools used: * Qtractor for recording and mixing * seq24 for sequencing * The necessary plugins: - drumkv1 to hold the drum samples (drum samples are all from http://samples.kb6.de/) - a lot of plugins that are part of Distrho or Carla: Noize Maker, Tal Reverb III, ZynAddSubFX-LV2, Nekobi - MDA subsynth - FluidSynth DSSI for the Rhodes - linuxDSP plugins (EQ500, DYN500, MBC2B on the master bus) - Calf Vintage Delay - LADSPA comb filter, Fast Lookahead Limiter - GxZitaReverb The background vocals for the choruses are sung by my wife. The ocean sample is from Freesound: http://www.freesound.org/people/dobroide/sounds/93653/ C?diz is pretty close to Conil, hence the choice. Lyrics ------ Making promises that I can't keep It's pushing me, pushing me into a deep State of sadness, state of doubt A state of awareness I can't live without Making mistakes, it's so hard to bear It's driving me, driving me to a point where I can't escape, I can't shy away From the daemons I refuse to obey All is forgiven, all is well... Awaiting the day that I'll be relieved From this burden, this burden that has grieved So many loved ones, so many friends All the people on which I depend Stand up, act now, it's time for a change Lingering won't help, help to rearrange The current imbalance, the current state Of things so rush now don't hesitate All is forgiven, all is well ------ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From dlphillips at woh.rr.com Sun Mar 30 18:39:14 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Sun, 30 Mar 2014 14:39:14 -0400 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <53385AC2.1040708@autostatic.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> <53385AC2.1040708@autostatic.com> Message-ID: <533864D2.4060902@woh.rr.com> On 03/30/2014 01:56 PM, Jeremy Jongepier wrote: > On 03/29/2014 03:50 PM, Dave Phillips wrote: >> On 03/29/2014 10:43 AM, rosea.grammostola wrote: >>> On 03/29/2014 01:31 PM, Hartmut Noack wrote: >>>> and Bitwig is connected automatically to >>>> Jack's system ports.... >>> This is not proper behavior if you can't turn it off >> Unfortunately it's how many (most?) Portaudio clients behave with their >> JACK connections. Very irritating with Audacity, for instance. >> >> Best, >> >> dp > You can disable this, in Audacity autoconnecting is done here: > http://code.google.com/p/audacity/source/browse/audacity-src/trunk/lib-src/portaudio-v19/src/hostapi/jack/pa_jack.c > > And then at line 1771. Iirc uncommenting the two if stanzas should do > the trick. At line 88 you can change the client name btw, could be handy > too. > > Bye, > > Jeremy > Thanks, Jeremy. I do sometimes build Audacity, I'll edit the code next time and let you know how the changes worked. Best, dp From joelz at pobox.com Sun Mar 30 18:44:35 2014 From: joelz at pobox.com (Joel Roth) Date: Sun, 30 Mar 2014 08:44:35 -1000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396198442.578.15.camel@archlinux> References: <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396189376.995.50.camel@archlinux> <53382DD6.30602@gareus.org> <1396191993.995.64.camel@archlinux> <1396198442.578.15.camel@archlinux> Message-ID: <20140330184435.GA15683@sprite> Ralf Mardorf wrote: > Btw. we could use templates for Qtractor, > Ardour and other Linux DAWs that provide tracks with EQs, I don't claim > that using "free" Linux software doesn't provide a smoothly work-flow, > but it takes some preparation and can't be done for all the musician's > needs. Using Linux means to care about many things and it's not always > easy. Yes, for example, Nama's mastering network and effect defaults[1] are a black box to me, yet I would guess a common enough configuration that I would hope someone else can do the work that I might not be able to accomplish myself. Users of many other DAWs would benefit as well. And users of Ecasound might like to have volume/pan controls that respond smoothly to realtime parameters changes. Finding suitable LADSPA/LV2 plugins need only take a contribution for one savvy Linux audio user who has already solved these issues for him/herself. Cheers, Joel 1. Nama's mastering network. +- Low -+ | | Master_in --- Eq --+- Mid -+--- Boost -> soundcard/wav_out | | +- High + The Eq track hosts an equalizer. The Low, Mid and High tracks each apply a bandpass filter, a compressor and a spatialiser. The Boost track applies gain and a limiter. eq: Parametric1 1 0 0 40 1 0 0 200 1 0 0 600 1 0 0 3300 1 0 low_pass: lowpass_iir 106 2 mid_pass: bandpass_iir 520 800 2 high_pass: highpass_iir 1030 2 compressor: sc4 0 3 16 0 1 3.25 0 spatialiser: matrixSpatialiser 0 limiter: tap_limiter 0 0 -- Joel Roth From jeremy at autostatic.com Sun Mar 30 19:09:57 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sun, 30 Mar 2014 21:09:57 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396188340.995.46.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <1396177255.587.3.camel@archlinux> <5338208E.8030204@gareus.org> <1396188340.995.46.camel@archlinux> Message-ID: <53386C05.1080303@autostatic.com> On 03/30/2014 04:05 PM, Ralf Mardorf wrote: > Replacing a pickup for your Stratocaster doesn't need that much knowhow > as setting up a Linux DAW It does. It takes several skills: - being able to hear subtle differences between different pickups (unless you replace the pickup with an exact copy) - being able to choose a pickup that matches your guitar (unless you replace the pickup with an exact copy) - knowing how to restring your guitar - knowing how to solder - basic knowledge of electronics, unless you want to go for that Peter Green sound - knowing how to mount a PU with regard to the distance to the strings And then we're not talking about actually playing the guitar. Bye, Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From accensi at gmail.com Sun Mar 30 20:41:51 2014 From: accensi at gmail.com (A. C. Censi) Date: Sun, 30 Mar 2014 17:41:51 -0300 Subject: [LAU] Imitone In-Reply-To: References: Message-ID: http://arstechnica.com/gadgets/2014/03/hands-on-with-imitone-if-you-can-hum-you-can-make-music-with-it/ Hands on with Imitone: if you can hum, you can make music with it Kickstarted app aims to help folks capture the songs they hear in their heads. -- A C Censi [on the move] accensi [em] g m a i l [ponto] com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlphillips at woh.rr.com Sun Mar 30 21:06:12 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Sun, 30 Mar 2014 17:06:12 -0400 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <5338648C.6020405@autostatic.com> References: <5338648C.6020405@autostatic.com> Message-ID: <53388744.6030100@woh.rr.com> On 03/30/2014 02:38 PM, Jeremy Jongepier wrote: > Dear all, > > Finally got around finishing a new track. And it's just 65BPM so no four > to the floor this time. > > http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg > > This song is about one of the most beautiful places I know on this > planet, Cala Del Aceite in the most southern part of Spain That's lovely work, Jeremy. Harmonies are sweet, well-recorded. Production overall is excellent. Fine lyrics & tune, had to hear it again after the first play. Thank you ! Best, dp From rosea.grammostola at gmail.com Sun Mar 30 21:25:37 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Sun, 30 Mar 2014 23:25:37 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <53388744.6030100@woh.rr.com> References: <5338648C.6020405@autostatic.com> <53388744.6030100@woh.rr.com> Message-ID: <53388BD1.4080000@gmail.com> One of your best songs, if not the best. Nice work. From list at nilsgey.de Mon Mar 31 00:44:20 2014 From: list at nilsgey.de (Nils) Date: Mon, 31 Mar 2014 02:44:20 +0200 Subject: [LAU] Imitone In-Reply-To: References: Message-ID: <5338BA64.20800@nilsgey.de> So much money wasted on practically aubio with a "hip" UI. On 30.03.2014 22:41, A. C. Censi wrote: > > > http://arstechnica.com/gadgets/2014/03/hands-on-with-imitone-if-you-can-hum-you-can-make-music-with-it/ > > Hands on with Imitone: if you can hum, you can make music with it > > Kickstarted app aims to help folks capture the songs they hear in > their heads. > > -- > A C Censi [on the move] > accensi [em] g m a i l [ponto] com > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Mon Mar 31 01:38:14 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 30 Mar 2014 18:38:14 -0700 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it Message-ID: On Sun, March 30, 2014 11:44 am, Joel Roth wrote: > 1. Nama's mastering network. > > +- Low -+ > | | > Master_in --- Eq --+- Mid -+--- Boost -> soundcard/wav_out > | | > +- High + > This was most interesting. I will say I was surprised that the low/mid/high eq is in parallel but do remember that many of the analogue versions do look like parallel circuits too. My question is two (maybe three or four?) fold then. As I had always assumed that these filters would be in series (something about how they are laid out on the desks maybe?) ... How is this done in high end analogue desks? How is this done in the LV2 (etc.) plug-ins? (those with more than one filter in them) If some do it one way over the other, how can I tell which is which without going through code I don't understand? Ardour, nonmixer, jackrack etc. are set up linear, one plugin into the next. So if I want to use a lowpass, highpass and bandpass filter, they will be one after the other. How will this affect the sound? Is one way better than another? will they turnout to have the same effect anyway? This is hard to visualize in my head and more of my experience has been live sound reinforcement than recording. -- Len Ovens www.OvenWerks.net From dlphillips at woh.rr.com Mon Mar 31 03:47:07 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Sun, 30 Mar 2014 23:47:07 -0400 Subject: [LAU] a little remix Message-ID: <5338E53B.1060703@woh.rr.com> Greetings, https://soundcloud.com/davephillips69/i-received-a-letter 1'49" of downloadable CC-licensed all-original mashed-up words and music, IIRC. Best, dp From ralf.mardorf at rocketmail.com Mon Mar 31 05:58:25 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 07:58:25 +0200 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: References: Message-ID: <1396245505.594.11.camel@archlinux> On Sun, 2014-03-30 at 18:38 -0700, Len Ovens wrote: > How is this done in high end analogue desks? Search the Internet for EQ circuit diagrams, ignore block diagrams for desks. You don't need to understand the complete circuit diagram, you anyway will be able to answer how it is done yourself. From nettings at stackingdwarves.net Mon Mar 31 07:37:05 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Mon, 31 Mar 2014 09:37:05 +0200 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: References: Message-ID: <53391B21.9070509@stackingdwarves.net> On 03/31/2014 03:38 AM, Len Ovens wrote: > Ardour, nonmixer, jackrack etc. are set up linear, one plugin into the > next. So if I want to use a lowpass, highpass and bandpass filter, they > will be one after the other. How will this affect the sound? not at all. > Is one way > better than another? analog parallel filters can have some benefits wrt signal-to-noise ratio, but you need to get the phase alignment right. for digital, it's mostly an unnecessary complicaton, unless you are going for effects that really require parallel processing (such as upwards compression). > will they turnout to have the same effect anyway? > This is hard to visualize in my head and more of my experience has been > live sound reinforcement than recording. unless you know very well what you are doing, parallel signal paths are full of dangerous pitfalls. a while ago, someone on some mailing list brought up the concept of parallel equalising (as in, split the signal, filter one half, recombine it with the original). reasoning was, hey, parallel compression is what the pros do, so parallel eq'ing must be great too, right? makes no sense, and produces something like this (the blue line is what you set your eq to, the red line is what you get when combining it with an unprocessed signal): http://stackingdwarves.net/download/4th-order%20highpass.png http://stackingdwarves.net/download/4th-order%20lowpass.png http://stackingdwarves.net/download/Parametric%20Mitra-Regalia.png morale: anything that introduces group delay shouldn't be used in parallel processing unless you know what you are doing. hth, j?rn -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From jeremy at autostatic.com Mon Mar 31 08:31:19 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 10:31:19 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <53388744.6030100@woh.rr.com> References: <5338648C.6020405@autostatic.com> <53388744.6030100@woh.rr.com> Message-ID: <533927D7.4040804@autostatic.com> On 03/30/2014 11:06 PM, Dave Phillips wrote: > > On 03/30/2014 02:38 PM, Jeremy Jongepier wrote: >> Dear all, >> >> Finally got around finishing a new track. And it's just 65BPM so no four >> to the floor this time. >> >> http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg >> >> >> This song is about one of the most beautiful places I know on this >> planet, Cala Del Aceite in the most southern part of Spain > > That's lovely work, Jeremy. Harmonies are sweet, well-recorded. > Production overall is excellent. Fine lyrics & tune, had to hear it > again after the first play. > > Thank you ! > > Best, > > dp Hi Dave, Thanks! Had quite a hard time achieving a mix to my liking. The kick could need some more attention and in hindsight I could've adjusted some levels. And my new audio interface is merciless, I really needed some time to adapt to that. Also found out that my cans and monitors don't work well with it :( Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From jeremy at autostatic.com Mon Mar 31 08:32:04 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 10:32:04 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <53388BD1.4080000@gmail.com> References: <5338648C.6020405@autostatic.com> <53388744.6030100@woh.rr.com> <53388BD1.4080000@gmail.com> Message-ID: <53392804.3060100@autostatic.com> On 03/30/2014 11:25 PM, rosea.grammostola wrote: > One of your best songs, if not the best. Nice work. Thanks Rosea! There is more where this came from ;) Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From moshwe at gmail.com Mon Mar 31 08:39:35 2014 From: moshwe at gmail.com (Moshe Werner) Date: Mon, 31 Mar 2014 11:39:35 +0300 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <533927D7.4040804@autostatic.com> References: <5338648C.6020405@autostatic.com> <53388744.6030100@woh.rr.com> <533927D7.4040804@autostatic.com> Message-ID: Beautiful piece Jeremy, I really enjoyed this one. On Mon, Mar 31, 2014 at 11:31 AM, Jeremy Jongepier wrote: > On 03/30/2014 11:06 PM, Dave Phillips wrote: > > > > On 03/30/2014 02:38 PM, Jeremy Jongepier wrote: > >> Dear all, > >> > >> Finally got around finishing a new track. And it's just 65BPM so no four > >> to the floor this time. > >> > >> > http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg > >> > >> > >> This song is about one of the most beautiful places I know on this > >> planet, Cala Del Aceite in the most southern part of Spain > > > > That's lovely work, Jeremy. Harmonies are sweet, well-recorded. > > Production overall is excellent. Fine lyrics & tune, had to hear it > > again after the first play. > > > > Thank you ! > > > > Best, > > > > dp > > Hi Dave, > > Thanks! Had quite a hard time achieving a mix to my liking. The kick > could need some more attention and in hindsight I could've adjusted some > levels. And my new audio interface is merciless, I really needed some > time to adapt to that. Also found out that my cans and monitors don't > work well with it :( > > Jeremy > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Mar 31 09:03:51 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 11:03:51 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: References: <5338648C.6020405@autostatic.com> <53388744.6030100@woh.rr.com> <533927D7.4040804@autostatic.com> Message-ID: <53392F77.9070401@autostatic.com> On 03/31/2014 10:39 AM, Moshe Werner wrote: > Beautiful piece Jeremy, > I really enjoyed this one. Thanks Moshe! Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From pshirkey at boosthardware.com Mon Mar 31 09:11:06 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Mon, 31 Mar 2014 20:11:06 +1100 (EST) Subject: [LAU] music from image metadata Message-ID: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> Hi, Can anyone think of a way to automate the creation of a music track from the metadata embedded in an image track? I'm thinking of this one in particular. http://www.jimstonefreelance.com/phillipwood.html -- Patrick Shirkey Boost Hardware Ltd From fons at linuxaudio.org Mon Mar 31 09:22:27 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 09:22:27 +0000 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: References: Message-ID: <20140331092227.GA31837@linuxaudio.org> On Sun, Mar 30, 2014 at 06:38:14PM -0700, Len Ovens wrote: > My question is two (maybe three or four?) fold then. As I had always > assumed that these filters would be in series (something about how they > are laid out on the desks maybe?) ... How is this done in high end > analogue desks? They almost always have a series connection of individual sections. Putting the sections in series means that they are independent of each other, the gains (in dB) just add, and the order doesn't matter. Parallel arrangements are used in e.g. 1/3 oct band equalisers. Putting filter in parallel requires very careful design because strange things may happen in the regions between the center frequencies due to phase differences. For the same reason, combining an equalised signal with the original may produce unexpected results. Things are different *within a single EQ section*. For example implementing a parametric or shelf filter as the sum of the input and a filtered version of the same has some advantages, even if you get exactly the same response (including phase) as when using another scheme. For one, if the gain is set to zero, you get a bit-exact copy of the input as output. This is not the case for e.g. a biquad set for a flat response - this will act as two filters that just happen to cancel each other. But even then such sections should be put in series. My 4-band LADSPA (also available as an LV2) works that way. > How is this done in the LV2 (etc.) plug-ins? (those with more than one > filter in them) If some do it one way over the other, how can I tell which > is which without going through code I don't understand? The advertising blurb should at least tell you what the thing is doing... > Ardour, nonmixer, jackrack etc. are set up linear, one plugin into the > next. So if I want to use a lowpass, highpass and bandpass filter, they > will be one after the other. How will this affect the sound? If you put filters in series with nothing non-linear in between them then the order doesn't matter, provided 1) you use a sample format with nearly unlimited dynamic range such as floats, and 2) none of the filters has gross defects. The second condition is not satisfied with all LADSPA and LV2 plugins, there are some that will happily add noise at -50 dB relative to the signal for some combinations of inputs and settings. Parallel arrangements have their place, e.g. for multiband compressors used for mastering. Regarding these you should be aware that such processing makes nonsense of whatever careful EQ you have done before. Multiband compression amounts an EQ that is changing all the time, and you usually have no idea of what exactly it is doing. If the mixing and mastering are done by the same person (as they usually are in our context), everything done in mastering could have been done better during the mix, e.g. by compressing tracks instead of the mix. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From rosea.grammostola at gmail.com Mon Mar 31 09:23:01 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Mon, 31 Mar 2014 11:23:01 +0200 Subject: [LAU] Bitwig 1.0 is out In-Reply-To: <533864D2.4060902@woh.rr.com> References: <5332C7C2.9020607@gmail.com> <5335A4B4.2040503@linuxuse.de> <53369D56.1000704@gmail.com> <5336BD0E.5080009@linuxuse.de> <5336DBF9.9010706@gmail.com> <5336DDB9.4050105@woh.rr.com> <53385AC2.1040708@autostatic.com> <533864D2.4060902@woh.rr.com> Message-ID: <533933F5.6090205@gmail.com> It's possible to drag a midi file into a track, so you can use Bitwig for advanced midi editing if you need it for your hit record. Same for audio of course. That's good to know and have on Linux. From rosea.grammostola at gmail.com Mon Mar 31 10:12:47 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Mon, 31 Mar 2014 12:12:47 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533817DB.9090503@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> Message-ID: <53393F9F.1020100@gmail.com> On 03/30/2014 03:10 PM, rosea.grammostola wrote: > > It's true, the attention to workflow and needs of musicians of Bitwig > is impressive. > I'm sure I'll be aware even more of it's possibilities if someone > shows me what you can do in Bitwig exactly. > > On the other hand, looking at Bitwig I'm not sure whether I should be > more impressed by Biwtig or by the achievements of the linuxuadio > community, of what is possible with Floss Linuxaudio already. And yes > impressed also by the technical infrastructure. I mean, what you get > from Bitwig is portaudio, that's almost a shame for a 300 euro app > from a linuxaudio-user pov. Also it lacks OSC, LADSPA, LV2 and NSM > support. Stuff you'll find in Ardour for instance. > > Also the quality in sound(instruments/samples) they give you for your > 300 euro, is not impressive for me, don't get fooled. But this seems > to be cultural thing, the acceptance of low quality sound. > > My conclusion so far is that Bitwig gives you what Linuxaudio lack too > often, smooth workflow and 'completeness' of features. This is a major > thing for people who want to make music! > > On the other hand, apart from very sophisticated features for making > beats etc., a lot could be possible with Linuxaudio tools or is > already possible today. > > The challenge today is to make Linuxaudio tools more friendly and > complete for musicians and integrate them better with each other. With > metadata in JACK, NSM, OSC it should be possible to improve this more > and more. It would be nice to launch Non-Timeline in a NSM session > with Carla and control Carla by Non-Timeline via OSC for instance. Oh and we need to improve and polish our MIDI sequencers on Linux From alexandre.prokoudine at gmail.com Mon Mar 31 10:16:47 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 14:16:47 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53393F9F.1020100@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53393F9F.1020100@gmail.com> Message-ID: On Mon, Mar 31, 2014 at 2:12 PM, rosea.grammostola wrote: > Oh and we need to improve and polish our MIDI sequencers on Linux So you want to Polish something that's already full of Czechnology? :) Alexandre From rosea.grammostola at gmail.com Mon Mar 31 10:20:17 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Mon, 31 Mar 2014 12:20:17 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53393F9F.1020100@gmail.com> Message-ID: <53394161.1090902@gmail.com> On 03/31/2014 12:16 PM, Alexandre Prokoudine wrote: > On Mon, Mar 31, 2014 at 2:12 PM, rosea.grammostola wrote: > >> Oh and we need to improve and polish our MIDI sequencers on Linux > So you want to Polish something that's already full of Czechnology? :) Heh, yeah. In terms of workflow, looks and feel. I guess that's hiding a big part of the Czechnology inside it for the end user :) From rosea.grammostola at gmail.com Mon Mar 31 10:22:01 2014 From: rosea.grammostola at gmail.com (rosea.grammostola) Date: Mon, 31 Mar 2014 12:22:01 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53394161.1090902@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53393F9F.1020100@gmail.com> <53394161.1090902@gmail.com> Message-ID: <533941C9.4040001@gmail.com> On 03/31/2014 12:20 PM, rosea.grammostola wrote: > On 03/31/2014 12:16 PM, Alexandre Prokoudine wrote: >> On Mon, Mar 31, 2014 at 2:12 PM, rosea.grammostola wrote: >> >>> Oh and we need to improve and polish our MIDI sequencers on Linux >> So you want to Polish something that's already full of Czechnology? :) > Heh, yeah. In terms of workflow, looks and feel. I guess that's hiding > a big part of the Czechnology inside it for the end user :) Ah Polish ... got it :) From robin at gareus.org Mon Mar 31 10:23:15 2014 From: robin at gareus.org (Robin Gareus) Date: Mon, 31 Mar 2014 12:23:15 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533817DB.9090503@gmail.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> Message-ID: <53394213.5080701@gareus.org> On 03/30/2014 03:10 PM, rosea.grammostola wrote: > My conclusion so far is that Bitwig gives you what Linuxaudio lack > too often, smooth workflow and 'completeness' of features. This is a > major thing for people who want to make music! only for some kind of music. Bitwig is just a toy compared to for example Csound and Supercollider. > The challenge today is to make Linuxaudio tools more friendly and > complete for musicians and integrate them better with each other. I don't think it's much of a challenge. At least not challenge enough to motivate any of the people who volunteer their time to linux audio. Bitwig has a clear motivation for what they do. AFAIK, the interest of most linux-audio developers is otherwise. What we can learn from Bitwig is that, it can be done if you get a good team together, motivate them towards a common goal and make compromises.. but we knew that, didn't we? 2c, robin From gordonjcp at gjcp.net Mon Mar 31 10:33:02 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Mon, 31 Mar 2014 11:33:02 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53394213.5080701@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> Message-ID: <20140331103302.GA2641@gjcp.net> On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: > On 03/30/2014 03:10 PM, rosea.grammostola wrote: > > > My conclusion so far is that Bitwig gives you what Linuxaudio lack > > too often, smooth workflow and 'completeness' of features. This is a > > major thing for people who want to make music! > > only for some kind of music. > > Bitwig is just a toy compared to for example Csound and Supercollider. > But Csound and Supercollider are not suitable for making music. They're fine if you're some kind of autistic savant computer genius, but utterly fucking useless if you're a musician. -- Gordonjcp MM0YEQ From harryhaaren at gmail.com Mon Mar 31 10:38:14 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Mon, 31 Mar 2014 11:38:14 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331103302.GA2641@gjcp.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: On Mon, Mar 31, 2014 at 11:33 AM, Gordon JC Pearce wrote: > But Csound and Supercollider are not suitable for making music. They're > fine if you're some kind of autistic savant computer genius, but utterly > fucking useless if you're a musician. > Lets keep "Use-case" in mind. Bitwig contributes in to certain use case, for making music (geared towards electronic styles, although not solely those). CSound & SuperCollider contribute to a very different use-case. Linux audio in general does not (IMO) offer a whole lot for the use-case of electro / radio styles of music, with software geared at ease of production of such music for musicians. Its a techie environment. My 2 cents :) -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From robin at gareus.org Mon Mar 31 10:42:10 2014 From: robin at gareus.org (Robin Gareus) Date: Mon, 31 Mar 2014 12:42:10 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331103302.GA2641@gjcp.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: <53394682.3090102@gareus.org> On 03/31/2014 12:33 PM, Gordon JC Pearce wrote: > On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: >> On 03/30/2014 03:10 PM, rosea.grammostola wrote: >> >>> My conclusion so far is that Bitwig gives you what Linuxaudio >>> lack too often, smooth workflow and 'completeness' of features. >>> This is a major thing for people who want to make music! >> >> only for some kind of music. >> >> Bitwig is just a toy compared to for example Csound and >> Supercollider. >> > > But Csound and Supercollider are not suitable for making music. > They're fine if you're some kind of autistic savant computer > genius, but utterly fucking useless if you're a musician. > It really depends what you think is music and what style you like and want to make. I recently saw an amazing performance of four opera singers + supercollider (+ some python, set with lilypond, etc etc) All of them musicians. Genius, yes. Autistic savant computer genius, no. http://www.ensemble101.fr/ blew me away. 2c, robin From fons at linuxaudio.org Mon Mar 31 11:26:45 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 11:26:45 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331103302.GA2641@gjcp.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: <20140331112645.GA26586@linuxaudio.org> On Mon, Mar 31, 2014 at 11:33:02AM +0100, Gordon JC Pearce wrote: > On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: > > > only for some kind of music. > > > > Bitwig is just a toy compared to for example Csound and Supercollider. > > > > But Csound and Supercollider are not suitable for > making music. They're fine if you're some kind of > autistic savant computer genius, but utterly fucking > useless if you're a musician. Define musician. The people who are able to use Csound and Supercollider can do it because they have invested time and effort in learning to do it. As has anyone who can play whatever instrument in a passable way (doesn't matter if it is a violin or a bass guitar). As has a composer who can arrange a song and write a score for it without needing a battery of synths to know how it will sound, or to check if his harmony is right. And no matter how you turn it, learning to do something difficult has beneficial side effects, apart from the primary result. Your 'musician' seems to be one for whom everything has to be prepared before and easy, so the only thing that remains to be done is some clicking on a screen. And then think him/herself a musician just as the kids wasting their time with shoot-and-kill games imagine they are soldiers. Your 'musician' is in fact just cannon fodder for an industry that is about making fast money and little else. And he wouldn't even be able to exist without the efforts of those who can rightly call themselves musicians and be proud of it. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Mon Mar 31 11:48:59 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 13:48:59 +0200 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <20140331092227.GA31837@linuxaudio.org> References: <20140331092227.GA31837@linuxaudio.org> Message-ID: <1396266539.568.3.camel@archlinux> J?rn, Fons, was there the need to reply detailed today? Tomorrow is the first of April and it would have been nice to wait with detailed explanations until tomorrow and to correct them the day after tomorrow. Geeks! Spoilsports! From dlphillips at woh.rr.com Mon Mar 31 11:55:27 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 31 Mar 2014 07:55:27 -0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331112645.GA26586@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <533957AF.3040503@woh.rr.com> On 03/31/2014 07:26 AM, Fons Adriaensen wrote: > On Mon, Mar 31, 2014 at 11:33:02AM +0100, Gordon JC Pearce wrote: > >> On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: >> >>> only for some kind of music. >>> >>> Bitwig is just a toy compared to for example Csound and Supercollider. >>> >> But Csound and Supercollider are not suitable for >> making music. They're fine if you're some kind of >> autistic savant computer genius, but utterly fucking >> useless if you're a musician. > Define musician. I was going to respond with the simpler "Bullshit!" but since you've dilated my original notion... > > Your 'musician' seems to be one for whom everything > has to be prepared before and easy, so the only thing > that remains to be done is some clicking on a screen. > And then think him/herself a musician just as the > kids wasting their time with shoot-and-kill games > imagine they are soldiers. I'm not so sure it's that simple. Even well-trained classical and jazz musicians display evident prejudice towards certain kinds of music within their own genres. One "jazz" fan loves his Dixieland, another can't do without Sun Ra. The Beethoven freak down the street absolutely despises Schoenberg, while my modernist buddy can't stand to hear Mozart. De gustibus and all that aside, it's a multi-format world. I get mine through YouTube channels such as pelodelperro, musicaignotus, and the Wellesz Company channels. No Vevo ads near those bad boys ! :) > Your 'musician' is in fact just cannon fodder for > an industry that is about making fast money and little > else. Again, I think it's a little more complicated. I respect skill and enthusiasm where-ever I can find it these days. I hear great efforts from pop/rock musicians, film score composers, game sound designers, and so forth. Musically it's not often my preferred listening choices but I can set aside prejudice long enough to try to get into the music per se. I'm always learning, and lessons can be found everywhere. Recently a pop musician friend claimed it was impossible for anyone to know and sing 60 songs a night. I had to remind him that his experience was limited to a few performances a year, not club dates, and certainly nothing like the 6-night-a-week 6-sets-per-night gigs I played in Chicago, two to three weeks at a booking. We were doing ~15 songs per set then, the math puts his assertion to rest. And about that "quick & efficient" thing : I read an interview with Philippe Leroux in which he mentioned working 20 months on 20 minutes of music. It reminded me of a statement by Morton Feldman where he said he wanted to get to the point where he put in the exact amount of time the music required, maybe only for 5 minutes one day and for 20 hours the next. Sometimes we can move so quickly we don't see the surround at all, and it can matter. EP: "To the man with something to say no vers is libre enough." > Ciao, > Always good to read your messages, Fons. Ciao ciao, dp From ralf.mardorf at rocketmail.com Mon Mar 31 12:21:26 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 14:21:26 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533957AF.3040503@woh.rr.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> Message-ID: <1396268486.568.22.camel@archlinux> On Mon, 2014-03-31 at 07:55 -0400, Dave Phillips wrote: > On 03/31/2014 07:26 AM, Fons Adriaensen wrote: > > Your 'musician' seems to be one for whom everything > > has to be prepared before and easy, so the only thing > > that remains to be done is some clicking on a screen. > > And then think him/herself a musician just as the > > kids wasting their time with shoot-and-kill games > > imagine they are soldiers. > > I'm not so sure it's that simple. Even well-trained classical and jazz > musicians display evident prejudice towards certain kinds of music > within their own genres. One "jazz" fan loves his Dixieland, another > can't do without Sun Ra. The Beethoven freak down the street absolutely > despises Schoenberg, while my modernist buddy can't stand to hear Mozart. > > > Your 'musician' is in fact just cannon fodder for > > an industry that is about making fast money and little > > else. > > Again, I think it's a little more complicated. I respect skill and > enthusiasm where-ever I can find it these days. I hear great efforts > from pop/rock musicians, film score composers, game sound designers, and > so forth. Musically it's not often my preferred listening choices but I > can set aside prejudice long enough to try to get into the music per se. > I'm always learning, and lessons can be found everywhere. Music has less to do with the skills to use a computer or the skills about music theory and skills to play an instrument. Creativity is independent of those skills. A lot of genius painters try to paint as naive as unskilled children do. They have the skills to paint what ever they want and they know all about the theory, but the intellectual aspiration often kills the fantasy. Musicians more often than painters tend to ignore that fact, likely because they feel the need to demonstrate that they are skilled musicians. Skills are not always good for creative work. A basic craftsmanship is needed, but it's not all that is needed to make good music. What is the message of music? I'm better than you? I learned to use computers, music theory and how to handle my musical instruments? Or do I want to send a message, like love, rage, jealousy etc.? Emotions have less to do with know-how. For some kinds of music I need to be able to compose and to play my instruments, then I only need a button to start the audio recording, for other kinds of music I need a good work-flow to edit MIDI data, to piece together the music. In both cases most important is the fantasy. A lot of people who are able to compose and to play instruments did edited tape compositions that had less to do with the music theory they learned and the ability to play instruments they are able to play. There are different kinds and motivations to make art by using "noise". From david at olofson.net Mon Mar 31 12:44:07 2014 From: david at olofson.net (David Olofson) Date: Mon, 31 Mar 2014 14:44:07 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331103302.GA2641@gjcp.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: On Mon, Mar 31, 2014 at 12:33 PM, Gordon JC Pearce wrote: [...] > But Csound and Supercollider are not suitable for making music. They're fine if you're some kind of autistic savant computer genius, but utterly fucking useless if you're a musician. Wouldn't say I'm either of those; just a rather experienced programmer with an interest in music... Either way; I used to think of this kind of tools as basically just powerful synthesizers, and controlling them from a "mainstream style" sequencer with a master keyboard and stuff pretty much seemed like the only viable setup. I've used various versions of Cakewalk and Sonar through the years, but obviously, I'd much rather use Linux, as that's what I use for practically everything these days. I just haven't been able to find a stable Linux sequencer that does what I need. (Would seem like very basic stuff, but I guess not... o.O) Then I started writing the synth/sound engine Audiality 2. It has a realtime scripting language that was originally just intended as a small but flexible replacement for the usual "five million" hardwired features you need to do much more than playing back dry samples. I intended to use it for both sound and music, but I was planning on using a MIDI sequencer for the latter. However, when testing and playing around, I discovered that just hacking music in the same scripting language was a surprisingly viable option, and as a bonus, offered tremendous flexibility through the seamless integration of sounds and music. So, I actually haven't bothered much with sequencers since! It would be nice to have the option of recording from MIDI controllers, since that's definitely the easiest, quickest and most natural way of doing many things - but I think I'd actually prefer a "sequencer" that just pastes code into my editor. :-) What weirdness came out of this episode of insanity, then? Well, some works in progress, mostly "chip inspired" stuff for a game I'm working on: https://soundcloud.com/david-olofson/sets/audiality-2-projects Oh, and that engine; Free/Open Source (zlib), of course: http://audiality.org/ -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------' From ralf.mardorf at rocketmail.com Mon Mar 31 12:54:05 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 14:54:05 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533957AF.3040503@woh.rr.com> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> Message-ID: <1396270445.568.30.camel@archlinux> On Mon, 2014-03-31 at 07:55 -0400, Dave Phillips wrote: > while my modernist buddy can't stand to hear Mozart. At least tow people from this list, Julien C. and Ralf M. dislike Wolfgang. > I hear great efforts > from pop/rock musicians, film score composers, game sound designers, and > so forth. Musically it's not often my preferred listening choices but I > can set aside prejudice long enough to try to get into the music per se. You never ever will find a Marc Bolan recording here, but while I don't own it and don't listen to it, IMO he's one of the very important contemporary musicians and he had impact to the music I make, perhaps more impact than Sch?nberg has got and I like Sch?nberg a lot. From gerhard.zintel at web.de Mon Mar 31 13:09:50 2014 From: gerhard.zintel at web.de (Gerhard Zintel) Date: Mon, 31 Mar 2014 15:09:50 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331112645.GA26586@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <201403311509.50956.gerhard.zintel@web.de> On Monday 31 March 2014, Fons Adriaensen wrote: > Your 'musician' is in fact just cannon fodder for > an industry that is about making fast money and little > else. And he wouldn't even be able to exist without > the efforts of those who can rightly call themselves > musicians and be proud of it. > Very well put From djdualcore at gmail.com Mon Mar 31 13:17:08 2014 From: djdualcore at gmail.com (Neil) Date: Mon, 31 Mar 2014 08:17:08 -0500 Subject: [LAU] music from image metadata In-Reply-To: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> References: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> Message-ID: This musicians does a lot with images. So far as I know he does it with the images themselves, not metadata. A lot of the stuff he posts on SoundCloud includes notes about his methods. https://soundcloud.com/dara-o-shayda On Mon, Mar 31, 2014 at 4:11 AM, Patrick Shirkey wrote: > Hi, > > Can anyone think of a way to automate the creation of a music track from > the metadata embedded in an image track? > > I'm thinking of this one in particular. > > http://www.jimstonefreelance.com/phillipwood.html > > > > > -- > Patrick Shirkey > Boost Hardware Ltd > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- DJ Dual Core's Blog http://oldmixtapes.blogspot.com/ Order without government; Peace without violence. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Mon Mar 31 13:22:05 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 15:22:05 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <201403311509.50956.gerhard.zintel@web.de> References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <201403311509.50956.gerhard.zintel@web.de> Message-ID: <1396272125.568.38.camel@archlinux> On Mon, 2014-03-31 at 15:09 +0200, Gerhard Zintel wrote: > On Monday 31 March 2014, Fons Adriaensen wrote: > > Your 'musician' is in fact just cannon fodder for > > an industry that is about making fast money and little > > else. And he wouldn't even be able to exist without > > the efforts of those who can rightly call themselves > > musicians and be proud of it. > > > Very well put No, it isn't. Fons is not completely mistaken, but it isn't that black and white. I'm a dino, able to compose and to play some instruments, since this was the only way to make music, when I was young. Nowadays there are other possibilities to make music. There is also a way to make music by trail and error. The end result is important, the know-how, needed time, good or bad intention is unimportant, if people, or at least the author like/s the result. Music isn't a competition about smartness. From cbannister at slingshot.co.nz Mon Mar 31 13:22:07 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Tue, 1 Apr 2014 02:22:07 +1300 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396268486.568.22.camel@archlinux> References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> <1396268486.568.22.camel@archlinux> Message-ID: <20140331132206.GA11734@tal> On Mon, Mar 31, 2014 at 02:21:26PM +0200, Ralf Mardorf wrote: > Music has less to do with the skills to use a computer or the skills > about music theory and skills to play an instrument. Creativity is > independent of those skills. [...] > other kinds of music I need a good work-flow to edit MIDI data, to piece > together the music. In both cases most important is the fantasy. [...] > There are different kinds and motivations to make art by using "noise". You mean like Karlheinz Stockhausen? :) http://www.youtube.com/watch?v=3XfeWp2y1Lk -- "If you're not careful, the newspapers will have you hating the people who are being oppressed, and loving the people who are doing the oppressing." --- Malcolm X From david at olofson.net Mon Mar 31 13:29:08 2014 From: david at olofson.net (David Olofson) Date: Mon, 31 Mar 2014 15:29:08 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331112645.GA26586@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: On Mon, Mar 31, 2014 at 1:26 PM, Fons Adriaensen wrote: [...] > Define musician. (My brain won't boot up properly today, so I kind of forgot what I was going to write in the first place...) "Someone who makes pleasant sounds in an organized fashion, using basically anything." -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------' From ralf.mardorf at rocketmail.com Mon Mar 31 13:30:20 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 15:30:20 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331132206.GA11734@tal> References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> <1396268486.568.22.camel@archlinux> <20140331132206.GA11734@tal> Message-ID: <1396272620.568.43.camel@archlinux> On Tue, 2014-04-01 at 02:22 +1300, Chris Bannister wrote: > On Mon, Mar 31, 2014 at 02:21:26PM +0200, Ralf Mardorf wrote: > > Music has less to do with the skills to use a computer or the skills > > about music theory and skills to play an instrument. Creativity is > > independent of those skills. > [...] > > other kinds of music I need a good work-flow to edit MIDI data, to piece > > together the music. In both cases most important is the fantasy. > [...] > > There are different kinds and motivations to make art by using "noise". > > You mean like Karlheinz Stockhausen? :) > > http://www.youtube.com/watch?v=3XfeWp2y1Lk I won't play the YouTube link, sine I'm not a fan of Stockhausen (anymore), but too funny, I indeed was thinking about some of his compositions, that btw. aren't bad, but relatively good, even while I dislike much of his music. You caught me :D. When I was young I liked Stockhausen, but I also liked Mozart *shudder*. From emviveros at gmail.com Mon Mar 31 13:33:47 2014 From: emviveros at gmail.com (Esteban Viveros) Date: Mon, 31 Mar 2014 10:33:47 -0300 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396268486.568.22.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> <1396268486.568.22.camel@archlinux> Message-ID: How I understand some points discussed here... First point: Make music is make the kind of music wich you learn to recognize how music (included the term "good music" at all). Second point: Make a perfect track can be beyond make a perfect music structure as make a good sound (included to make sound like mainstream does). I'm primarily a user, starting tending to developer recently, but my aims are make music, and help to make music easily. About the aim to make music, I'm using how first option Ableton Live9, and it can satisfy 90% of my needs, but this software are very expensive (it crashes with help to make music easily). But a great goals of this kind of software, like Live or Bitwig, are the clean interface, and drag and drop devices, witch frees the mind to create a musical or surrounding effect, depending on the know how. If the gay is starting, they have very usefull presets to impulse your sound. Another kind of think very important is the community, you can find a lot of tutorials in good quality of experient users making those tools to work well. Discriminating items in my comments that I would separate: - Interface designed at work creating the sound or musical structure as directly as possible. Here, FalkTX with KXStudio are doing a interesting work. I think all moment in something like Claudia Launcher in some space on the screen to select plugins, and in a first view, only drag and drop and it work. Can be considered an initial bundle that can only do a good sound (that good sound must be defined previously). I'm wasting long time experimenting interfaces, and I'm only starting to think this kind of thinks, but the primarly aim would an interface can access the stuff already created in linux ecosystem and organize in levels of complexity, first drag and drop, play the instruments, arrange, ou arrange and play wethever.. and play the song. The second level would a creative manipulation of outputs and inputs, like happens in patchbay or claudia. and at last third at level of programation new stuffs, like pd csound suppercollider, maybe only integrate some project of this (that's are the most difficult part I think). It's a idea and I'm collecting know how to implement it in some situation on the future... - Presets, that's a important and simple goal. If you don't know how to mix very well, first samples with pre-equalizations allowing loop a pre defined structure like Drum Bass and leader synth for example sound great only overlap. And some interesting aestetic stuff for a second level of mix, like presets to make sound more punchy, flat or muffled. We can think a lot of thinks here, both in quantity and when and how that amount available. - Tutorials in good quality. FalkTX, Rui Nuno Capella and LazerBlade are doing very good things accordingly. Every interface needs a time of learning, video tutorials in my vision is a great tool to maximize learning time. That's some thoughts I ever have, but I can't do much because I don't have know how to implement it, but gradually'm directing my ways for this to be allowed. 2014-03-31 9:21 GMT-03:00 Ralf Mardorf : > On Mon, 2014-03-31 at 07:55 -0400, Dave Phillips wrote: > > On 03/31/2014 07:26 AM, Fons Adriaensen wrote: > > > Your 'musician' seems to be one for whom everything > > > has to be prepared before and easy, so the only thing > > > that remains to be done is some clicking on a screen. > > > And then think him/herself a musician just as the > > > kids wasting their time with shoot-and-kill games > > > imagine they are soldiers. > > > > I'm not so sure it's that simple. Even well-trained classical and jazz > > musicians display evident prejudice towards certain kinds of music > > within their own genres. One "jazz" fan loves his Dixieland, another > > can't do without Sun Ra. The Beethoven freak down the street absolutely > > despises Schoenberg, while my modernist buddy can't stand to hear Mozart. > > > > > Your 'musician' is in fact just cannon fodder for > > > an industry that is about making fast money and little > > > else. > > > > Again, I think it's a little more complicated. I respect skill and > > enthusiasm where-ever I can find it these days. I hear great efforts > > from pop/rock musicians, film score composers, game sound designers, and > > so forth. Musically it's not often my preferred listening choices but I > > can set aside prejudice long enough to try to get into the music per se. > > I'm always learning, and lessons can be found everywhere. > > Music has less to do with the skills to use a computer or the skills > about music theory and skills to play an instrument. Creativity is > independent of those skills. A lot of genius painters try to paint as > naive as unskilled children do. They have the skills to paint what ever > they want and they know all about the theory, but the intellectual > aspiration often kills the fantasy. Musicians more often than painters > tend to ignore that fact, likely because they feel the need to > demonstrate that they are skilled musicians. Skills are not always good > for creative work. A basic craftsmanship is needed, but it's not all > that is needed to make good music. What is the message of music? I'm > better than you? I learned to use computers, music theory and how to > handle my musical instruments? Or do I want to send a message, like > love, rage, jealousy etc.? Emotions have less to do with know-how. > > For some kinds of music I need to be able to compose and to play my > instruments, then I only need a button to start the audio recording, for > other kinds of music I need a good work-flow to edit MIDI data, to piece > together the music. In both cases most important is the fantasy. > > A lot of people who are able to compose and to play instruments did > edited tape compositions that had less to do with the music theory they > learned and the ability to play instruments they are able to play. > > There are different kinds and motivations to make art by using "noise". > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Esteban Viveros (27) 98815 7170 | (11) 95761 4125 http://expurgacao.art.br/ https://soundcloud.com/estebanviveros http://projetobramaloka.tumblr.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Mon Mar 31 13:35:25 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 13:35:25 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <533957AF.3040503@woh.rr.com> References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> Message-ID: <20140331133525.GB21111@linuxaudio.org> On Mon, Mar 31, 2014 at 07:55:27AM -0400, Dave Phillips wrote: > On 03/31/2014 07:26 AM, Fons Adriaensen wrote: > > >Your 'musician' seems to be one for whom everything > >has to be prepared before and easy, so the only thing > >that remains to be done is some clicking on a screen. > >And then think him/herself a musician just as the > >kids wasting their time with shoot-and-kill games > >imagine they are soldiers. > > I'm not so sure it's that simple. Even well-trained classical and > jazz musicians display evident prejudice towards certain kinds of > music within their own genres. One "jazz" fan loves his Dixieland, > another can't do without Sun Ra. My remark wasn't about musical genres, to each his/her own. And I know some nice music made by arranging samples on a timeline - it's just very rare. The point is that anything that doesn't require any effort or background to make it is very likely not going to be interesting [*]. Those who know their art by training, study and experience will always be at an advantage and dominate a scene, just as a trained sportsman will win any competition against someone who doesn't care about training. Ciao, [*] except maybe as a overhyped fashion, as happens in the figurative arts. -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Mon Mar 31 13:42:33 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 15:42:33 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331133525.GB21111@linuxaudio.org> References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <533957AF.3040503@woh.rr.com> <20140331133525.GB21111@linuxaudio.org> Message-ID: <1396273353.568.47.camel@archlinux> On Mon, 2014-03-31 at 13:35 +0000, Fons Adriaensen wrote: > Those who know their art by training,study and experience will always > be at an advantage and dominate a scene At least it seldom does harm ;). I don't completely disagree, but there are exceptions. Old school hip hop, 80s no future punk and several others more. Now I sent 12 of 13. Bye, Ralf From zotz at 100jamz.com Mon Mar 31 14:03:26 2014 From: zotz at 100jamz.com (drew Roberts) Date: Mon, 31 Mar 2014 10:03:26 -0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> Message-ID: <201403311003.26792.zotz@100jamz.com> On Sunday 30 March 2014 06:27:35 Louigi Verona wrote: > Commercial projects tend to figure > out what their users actually want. Not disputing the general thrust of your post but: Commercial projects try to figure out what they need to do to get money out of their customers. If that involves giving their users what thjey want at times, they will try and do that. If they can get more money but not giving customers what they want, they will try and do that. all the best, drew -- This is drew's personal email account and is not related to Tribune Radio Ltd. From alexandre.prokoudine at gmail.com Mon Mar 31 14:02:40 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 18:02:40 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331112645.GA26586@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: On Mon, Mar 31, 2014 at 3:26 PM, Fons Adriaensen wrote: > Your 'musician' seems to be one for whom everything > has to be prepared before and easy, so the only thing > that remains to be done is some clicking on a screen. > And then think him/herself a musician just as the > kids wasting their time with shoot-and-kill games > imagine they are soldiers. > > Your 'musician' is in fact just cannon fodder for > an industry that is about making fast money and little > else. And he wouldn't even be able to exist without > the efforts of those who can rightly call themselves > musicians and be proud of it. For some reason the Linux community tends to promote radical views without the need to resort to the "golden middle", as seen above. For instance, a plenty of presets and samples shipped with a DAW means: 1) It's easy to learn how to use the DAW and its tools and then move on. 2) Granted, it's easy to abuse presets and prerecorded samples. Robin might correct me if I'm wrong, but the primary objective of him creating Reasonable Synth shipped with A3 is that A3 used to produce _no sound whatsoever_ when you tried to play back a .mid file in the importing dialog. And there had been no synths in the binary bundle before either, AFAIK. Moreover, not producing any sound by default from MIDI data is exactly the problem that Rosegarden specifically had to address in the _FAQ_ (not even actual features). And Qtractor doesn't make it a lot easier either. Don't you see what's wrong with this picture? If getting started and/or having a smooth composition workflow is too difficult, it doesn't really matter how powerful your software is. People just give up. That said, Linux DAWs are not yet ready for instruments with tons of presets anyway. Try navigating Amsynth with its hundreds of patches from a plain list in either A3 or Qtractor, and you'll soon give up too :) Alexandre From fons at linuxaudio.org Mon Mar 31 14:14:35 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 14:14:35 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <20140331141435.GC21111@linuxaudio.org> On Mon, Mar 31, 2014 at 06:02:40PM +0400, Alexandre Prokoudine wrote: > On Mon, Mar 31, 2014 at 3:26 PM, Fons Adriaensen wrote: > > > Your 'musician' seems to be one for whom everything > > has to be prepared before and easy, so the only thing > > that remains to be done is some clicking on a screen. > > And then think him/herself a musician just as the > > kids wasting their time with shoot-and-kill games > > imagine they are soldiers. > > > > Your 'musician' is in fact just cannon fodder for > > an industry that is about making fast money and little > > else. And he wouldn't even be able to exist without > > the efforts of those who can rightly call themselves > > musicians and be proud of it. > > For some reason the Linux community tends to promote radical views > without the need to resort to the "golden middle", as seen above. What you quote was a reaction to a 'radical view' without any motivation. And to make your point you deleted the motivation I provided for my 'radical view'. The 'golden middle' is a device invented by politicians for their own purposes. It usually amounts to condoning mediocrity. > Moreover, not producing any sound by default from MIDI data is exactly > the problem that Rosegarden specifically had to address in the _FAQ_ > (not even actual features). And Qtractor doesn't make it a lot easier > either. > > Don't you see what's wrong with this picture? I very much prefer just having to wire up the synth I want to having to remove the one I don't want first. > People just give up. Little is lost on someone who gives up so easily. Except maybe sales. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From alexandre.prokoudine at gmail.com Mon Mar 31 14:23:02 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 18:23:02 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <201403311003.26792.zotz@100jamz.com> References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> <201403311003.26792.zotz@100jamz.com> Message-ID: On Mon, Mar 31, 2014 at 6:03 PM, drew Roberts wrote: > Commercial projects try to figure out what they need to do to get money out of > their customers. If that involves giving their users what thjey want at > times, they will try and do that. That surely explains the army of "shut up and take my money" folks in Bitwig threads all across the interwebz :) Simply put, a successful service/software is the one that was built around a particular need (of people) that hadn't been taken care of before. How that service/software evolves later is an entirely different topic. In production software, if your app doesn't assist in getting shit down, you're out of the game. Sorry, but it's that simple. Users can put up with a lot of stuff as long as shit gets done. Make a simple test. Try doing the most basic thing in existing DAWs on Linux, including Bitwig -- getting sound out of a MIDI track -- and count steps, then multiply it by 20 tracks to get a better idea of boring work you need to do every time (Hint: A3 will win, because I specifically bugged Paul about it). Alexandre From alexandre.prokoudine at gmail.com Mon Mar 31 14:32:35 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 18:32:35 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331141435.GC21111@linuxaudio.org> References: <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> Message-ID: On Mon, Mar 31, 2014 at 6:14 PM, Fons Adriaensen wrote: > The 'golden middle' is a device invented by politicians > for their own purposes. It usually amounts to condoning > mediocrity. *facepalm* >> Moreover, not producing any sound by default from MIDI data is exactly >> the problem that Rosegarden specifically had to address in the _FAQ_ >> (not even actual features). And Qtractor doesn't make it a lot easier >> either. >> >> Don't you see what's wrong with this picture? > > I very much prefer just having to wire up the synth I want to > having to remove the one I don't want first. Seriously, Fons? Basically you've just admitted that you have no hands-on experience with Bitwig. What's worse, you appear to have no hands-on experience with Ardour 3. The thing is, neither Bitwig nor Ardour 3 use default synths for MIDI tracks (A3 does it only for importing). In both cases you specifically pick the ones that you want. The difference between Bitwig and e.g. Qtractor is how long it takes you to do that. >> People just give up. > > Little is lost on someone who gives up so easily. Except maybe > sales. *double facepalm* Alexandre From fons at linuxaudio.org Mon Mar 31 14:33:23 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 14:33:23 +0000 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <1396266539.568.3.camel@archlinux> References: <20140331092227.GA31837@linuxaudio.org> <1396266539.568.3.camel@archlinux> Message-ID: <20140331143323.GD21111@linuxaudio.org> On Mon, Mar 31, 2014 at 01:48:59PM +0200, Ralf Mardorf wrote: > J?rn, Fons, was there the need to reply detailed today? Tomorrow is the > first of April and it would have been nice to wait with detailed > explanations until tomorrow and to correct them the day after tomorrow. Because tomorrow is my birthday, but of course nobody will believe that. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From jamshark70 at gmail.com Mon Mar 31 14:33:02 2014 From: jamshark70 at gmail.com (James Harkins) Date: Mon, 31 Mar 2014 14:33:02 +0000 (UTC) Subject: [LAU] Bitwig: what we can learn from it References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: Gordon JC Pearce writes: > But Csound and Supercollider are not suitable for making music. They're fine if you're some kind of > autistic savant computer genius, but utterly fucking useless if you're a musician. Well, in that case, it's a good thing I didn't read your message before I started... making music with SuperCollider tonight :-p This one's in 4/4 time, even. Oh, and if you heard BT's "This Binary Universe," you heard Csound in the first track (as in, that track is *all* Csound). Sure sounded a lot like A aeolian to me... but maybe it actually is unmusical just because of the tool he used. I thought it was pretty. I'd draw a parallel between programming languages for music and Western notation. Both are technical, complex symbolic systems that take a hefty time investment to master. It's not at all farfetched to argue that writing tens of thousands of dots in precise places on ruled paper, as, say, Mahler did in any single symphony, constitutes an act of autistic-savant persistence (in the days before computer notation, when scores and parts had to be copied by hand before the premiere). When you really think about it, it's insane... but it makes music. Writing code to make music is also insane... but in the hands of someone who works at it, it makes music. hjh From gheskett at wdtv.com Mon Mar 31 14:36:48 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Mon, 31 Mar 2014 10:36:48 -0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331112645.GA26586@linuxaudio.org> References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <201403311036.48445.gheskett@wdtv.com> On Monday 31 March 2014 10:36:30 Fons Adriaensen did opine: > On Mon, Mar 31, 2014 at 11:33:02AM +0100, Gordon JC Pearce wrote: > > On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: > > > only for some kind of music. > > > > > > Bitwig is just a toy compared to for example Csound and > > > Supercollider. > > > > But Csound and Supercollider are not suitable for > > making music. They're fine if you're some kind of > > autistic savant computer genius, but utterly fucking > > useless if you're a musician. > > Define musician. > > The people who are able to use Csound and Supercollider > can do it because they have invested time and effort in > learning to do it. As has anyone who can play whatever > instrument in a passable way (doesn't matter if it is > a violin or a bass guitar). As has a composer who can > arrange a song and write a score for it without needing > a battery of synths to know how it will sound, or to > check if his harmony is right. > > And no matter how you turn it, learning to do something > difficult has beneficial side effects, apart from the > primary result. > > Your 'musician' seems to be one for whom everything > has to be prepared before and easy, so the only thing > that remains to be done is some clicking on a screen. > And then think him/herself a musician just as the > kids wasting their time with shoot-and-kill games > imagine they are soldiers. > > Your 'musician' is in fact just cannon fodder for > an industry that is about making fast money and little > else. And he wouldn't even be able to exist without > the efforts of those who can rightly call themselves > musicians and be proud of it. > > > Ciao, +1 Fons. Cheers, Gene -- "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) Genes Web page From robin at gareus.org Mon Mar 31 14:42:02 2014 From: robin at gareus.org (Robin Gareus) Date: Mon, 31 Mar 2014 16:42:02 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <53397EBA.8020908@gareus.org> On 03/31/2014 04:02 PM, Alexandre Prokoudine wrote: > Robin might correct me if I'm wrong, but the primary objective of him > creating Reasonable Synth shipped with A3 is that A3 used to produce > _no sound whatsoever_ when you tried to play back a .mid file in the > importing dialog. Any MIDI track, actually (midi auditioning only came later). > And there had been no synths in the binary bundle > before either, AFAIK. well yeah. The primary objective was to no longer answer the recurring question "Why can I not hear MIDI?" Even though the way we preempted that question won't make the user any wiser about why s/he cannot hear MIDI, adding a synth was deemed the most _reasonable_ reply :( > Moreover, not producing any sound by default from MIDI data is exactly > the problem that Rosegarden specifically had to address in the _FAQ_ > (not even actual features). And Qtractor doesn't make it a lot easier > either. > > Don't you see what's wrong with this picture? yes, using proprietary DAWs dumbs users down. 2c, robin From ralf.mardorf at rocketmail.com Mon Mar 31 14:44:09 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 31 Mar 2014 16:44:09 +0200 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <20140331143323.GD21111@linuxaudio.org> References: <20140331092227.GA31837@linuxaudio.org> <1396266539.568.3.camel@archlinux> <20140331143323.GD21111@linuxaudio.org> Message-ID: <1396277049.2845.8.camel@archlinux> On Mon, 2014-03-31 at 14:33 +0000, Fons Adriaensen wrote: > On Mon, Mar 31, 2014 at 01:48:59PM +0200, Ralf Mardorf wrote: > > > J?rn, Fons, was there the need to reply detailed today? Tomorrow is the > > first of April and it would have been nice to wait with detailed > > explanations until tomorrow and to correct them the day after tomorrow. > > Because tomorrow is my birthday, but of course nobody will believe that. I guess this is the absolutely last mail I'm allowed to sent now. Ok, that's a good excuse, resp. reason and indeed it's hard to believe, but OTOH some people for sure are born at April the first. I hope I'm allowed to sent a mail tomorrow, to send congratulations. I disagree with many of your points of view, but your software still is superb :). From alexandre.prokoudine at gmail.com Mon Mar 31 14:48:53 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 18:48:53 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: On Mon, Mar 31, 2014 at 6:33 PM, James Harkins wrote: > Well, in that case, it's a good thing I didn't read your message before I > started... making music with SuperCollider tonight :-p This one's in 4/4 > time, even. Er, isn't it just a matter of taste? :) Gordon basically summarized (in a rather arguable manner) a point that we've discussed time and time again: if Linux audio is for geeks or for full-time musicians. In the eyes of slightly informed masses, Linux audio either doesn't exist, or is closely related to experimental electronic music (alternatively, to death metal bands). The big win for LMMS and Qtractor (with all their disadvantages) is that they are changing this perception towards (reluctantly) including EDM to this list. So what did you expect? :) Alexandre From fons at linuxaudio.org Mon Mar 31 14:51:13 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 14:51:13 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> Message-ID: <20140331145113.GE21111@linuxaudio.org> On Mon, Mar 31, 2014 at 06:32:35PM +0400, Alexandre Prokoudine wrote: > On Mon, Mar 31, 2014 at 6:14 PM, Fons Adriaensen wrote: > > > I very much prefer just having to wire up the synth I want to > > having to remove the one I don't want first. > > Seriously, Fons? Seriously. Having to remove any defaults I don't want is a waste of time. Setting up what I do want is necessary anyway. > Basically you've just admitted that you have no hands-on experience > with Bitwig. What's worse, you appear to have no hands-on experience > with Ardour 3. I didn't mention either of them. But for Bitwig that's absolutely true. It's not the sort of thing I need. Regarding A3, I've got lots of hands-on experience with it. Even if it lacks the features that would make *my* workflow a lot easier, and there is little chance they will ever be added. I just learn to get the job done using the tools that are available. In fact, A3 lacking the tools I need is much less a problem than it growing obese by having a lot of (for me) useless stuff built-in in a non-modular way. I've done multitrack recording on my laptop using A2. Today, even a completely empty A3 session will drive it into swapping. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From alexandre.prokoudine at gmail.com Mon Mar 31 14:56:08 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 18:56:08 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53397EBA.8020908@gareus.org> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <53397EBA.8020908@gareus.org> Message-ID: On Mon, Mar 31, 2014 at 6:42 PM, Robin Gareus wrote: >> Moreover, not producing any sound by default from MIDI data is exactly >> the problem that Rosegarden specifically had to address in the _FAQ_ >> (not even actual features). And Qtractor doesn't make it a lot easier >> either. >> >> Don't you see what's wrong with this picture? > > yes, using proprietary DAWs dumbs users down. I'm sorry, but I heard so few professionally composed and mixed pieces done with Linux audio software that I can't possibly imagine where this notion of intellectual superiority comes from. Alexandre From alexandre.prokoudine at gmail.com Mon Mar 31 15:07:18 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 19:07:18 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331145113.GE21111@linuxaudio.org> References: <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> <20140331145113.GE21111@linuxaudio.org> Message-ID: On Mon, Mar 31, 2014 at 6:51 PM, Fons Adriaensen wrote: >> Basically you've just admitted that you have no hands-on experience >> with Bitwig. What's worse, you appear to have no hands-on experience >> with Ardour 3. > > I didn't mention either of them. They were in the context that you appear to arbitrarily keep or drop :) OTOH, I'm not sure why it should matter at all given that you seem to have very little (if any) use for MIDI tracks. Given your background, we might as well argue whether Pianoteq is a blasphemy :) This is simply not the point. The point is that free software could be made easier to use without (much) introducing the "dumbing down" aspect. The question is if developers choose the sorry status quo or if they are more open-minded. Because pretty much all lessons we can learn from Bitwig end up in actually changing source code. Alexandre From list at nilsgey.de Mon Mar 31 17:16:16 2014 From: list at nilsgey.de (Nils) Date: Mon, 31 Mar 2014 19:16:16 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> <20140331145113.GE21111@linuxaudio.org> Message-ID: <5339A2E0.5090704@nilsgey.de> >Fons Adriaensen >Alexandre Prokoudine >Fons Adriaensen >Alexandre Prokoudine >Fons Adriaensen >Alexandre Prokoudine I would love to read an article based on the continuation of this discussion in private with well written arguments, a bit back and forth and conclusions and suggestions for the future. Too much work? Too much work. Nils From fons at linuxaudio.org Mon Mar 31 15:23:06 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 15:23:06 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> <20140331145113.GE21111@linuxaudio.org> Message-ID: <20140331152306.GF21111@linuxaudio.org> On Mon, Mar 31, 2014 at 07:07:18PM +0400, Alexandre Prokoudine wrote: > OTOH, I'm not sure why it should matter at all given that you seem to > have very little (if any) use for MIDI tracks. Given your background, > we might as well argue whether Pianoteq is a blasphemy :) It isn't in my view, because it's done rather well. BTW, some of the research that went into it was done here in Parma. I wasn't involved directly except for one recording session in the anechoic room in Ferrara where we had to go out to all stationary shops in the city to buy a few hundred rubber erasers, used to damp all the strings in a Steinway grand (we needed the IR of the soundboard). > The point is that free software could be made easier to use without > (much) introducing the "dumbing down" aspect. It could in some cases. > The question is if developers choose the sorry status quo or > if they are more open-minded. It has more to do with available time and resources than with open-mindedness. I will spend my time by trying to get the best DSP code. And in some cases by trying to make something more configurable and avoid artificial limits, even if that makes the end result harder to use. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From fons at linuxaudio.org Mon Mar 31 15:36:31 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 15:36:31 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <5339A2E0.5090704@nilsgey.de> References: <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331141435.GC21111@linuxaudio.org> <20140331145113.GE21111@linuxaudio.org> <5339A2E0.5090704@nilsgey.de> Message-ID: <20140331153631.GG21111@linuxaudio.org> On Mon, Mar 31, 2014 at 07:16:16PM +0200, Nils wrote: > I would love to read an article based on the continuation of this > discussion in private with well written arguments, a bit back and > forth and conclusions and suggestions for the future. Come to Karlsruhe, offer beers and/or wine. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From nettings at stackingdwarves.net Mon Mar 31 15:40:11 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Mon, 31 Mar 2014 17:40:11 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <53397EBA.8020908@gareus.org> Message-ID: <53398C5B.6070400@stackingdwarves.net> On 03/31/2014 04:56 PM, Alexandre Prokoudine wrote: > On Mon, Mar 31, 2014 at 6:42 PM, Robin Gareus wrote: > >>> Don't you see what's wrong with this picture? >> >> yes, using proprietary DAWs dumbs users down. > > I'm sorry, but I heard so few professionally composed and mixed pieces > done with Linux audio software that I can't possibly imagine where > this notion of intellectual superiority comes from. the tools have nothing to do with the artistic quality of the outcome. (even though a large part of the music industry works by making people believe just that). all they can do is let you arrive at a good result more quickly or more easily, or inspire you to try new things. the problem with the latter is that many people will get inspired in pretty similar ways, and the result of that you can hear on the radio. i have worked on quite a few proprietary DAWs, and i still can't believe the fiery hoops they make me jump through. just this week i've had to decide to do a short film sound do-over on $PROPRIETARY_TOOL, because i can't import the EDL into ardour (yet), and manually syncing 75 clips is not an option (no timecode on location, unfortunately, and now all i have is an OMF file). i thought, hey, let's import the session and bounce it out with all regions expanded, so i can finish the job on ardour. alas, $PT decides to bail out halfway through the export with a totally unhelpful error message. sigh. now i'm stuck with it. latest brickwall i ran into is that $PT (after taking 10 years to support native audio hardware at all) has this totally arbitrary limit of 32 channels of simultaneous native recording. you need more? buy the box. oh, you already have far superiour audio hardware and the studio is running dante? too bad. buy the box, and buy a dante bridge. right, that's the company which has been too stupid to get their accounting done properly for more than two years and got kicked out of NASDAQ for that. now that's two epic facepalms for the product that is responsible for most of the really professional content out there, whatever that means. which, to conclude this little rant, makes it very hard for me indeed to avoid that fuzzy feeling of intellectual superiority, even though i agree it is a bad habit. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From alexandre.prokoudine at gmail.com Mon Mar 31 15:51:32 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 19:51:32 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53398C5B.6070400@stackingdwarves.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <53397EBA.8020908@gareus.org> <53398C5B.6070400@stackingdwarves.net> Message-ID: On Mon, Mar 31, 2014 at 7:40 PM, J?rn Nettingsmeier wrote: >> I'm sorry, but I heard so few professionally composed and mixed pieces >> done with Linux audio software that I can't possibly imagine where >> this notion of intellectual superiority comes from. > > the tools have nothing to do with the artistic quality of the outcome. You completely misread my point. But I'm not blaming you. Sleep deprivation might have affected the way I phrase my words. I'm sure that we all can think of quite a few horrible stories about proprietary software (I've my collection too). The net outcome, however, is that most people who get shit done in audio professionally don't use free DAWs. Deal with it :-P Alexandre From murks at tuxfamily.org Mon Mar 31 15:55:05 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 31 Mar 2014 17:55:05 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> Message-ID: <20140331175505.71f4ef69@eeyore> On Mon, 31 Mar 2014 15:29:08 +0200 David Olofson wrote: > On Mon, Mar 31, 2014 at 1:26 PM, Fons Adriaensen > wrote: [...] > > Define musician. > > (My brain won't boot up properly today, so I kind of forgot what I was > going to write in the first place...) > > "Someone who makes pleasant sounds in an organized fashion, using > basically anything." Now that's interesting. Why exactly does it need to be pleasant sound, and what do you mean by that? Pleasant to whom? I think my definition would be even more general: "Someone who makes sound on purpose." Maybe an even more extreme definition is needed: "Someone who makes sound." Regards, Philipp From rosea.grammostola at gmail.com Mon Mar 31 15:59:12 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Mon, 31 Mar 2014 17:59:12 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <53398C5B.6070400@stackingdwarves.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <53397EBA.8020908@gareus.org> <53398C5B.6070400@stackingdwarves.net> Message-ID: A software tool should make tasks easy, so you don't waste time on cumbersome workflow, bugs and missing features, but focus that time and energy on making music. This has nothing to do with the intellectual level of the music or musician. It's probably just that proprietary software projects have more money, working hours and better cooperation/teamwork and better knowledge of GUI design to get this done. Using software on Linux makes you pretty often loose time on bad workflow, bad integration, bugs and missing features. The process of creativity or producing your creative work, stucks in the middle too often. I guess this is more true for MIDI then for audio (not surprisingly that Fons, Jorn, Robin work mainly with the audio part and have little to complain). One acoustic guitar can play smooth and gives you joy and satisfaction. An other guitar can give you blisters, no good sound, it doesn't help you to produce your creativity and is frustrating ... On Mon, Mar 31, 2014 at 5:40 PM, J?rn Nettingsmeier < nettings at stackingdwarves.net> wrote: > On 03/31/2014 04:56 PM, Alexandre Prokoudine wrote: > >> On Mon, Mar 31, 2014 at 6:42 PM, Robin Gareus wrote: >> >> Don't you see what's wrong with this picture? >>>> >>> >>> yes, using proprietary DAWs dumbs users down. >>> >> >> I'm sorry, but I heard so few professionally composed and mixed pieces >> done with Linux audio software that I can't possibly imagine where >> this notion of intellectual superiority comes from. >> > > the tools have nothing to do with the artistic quality of the outcome. > (even though a large part of the music industry works by making people > believe just that). all they can do is let you arrive at a good result more > quickly or more easily, or inspire you to try new things. the problem with > the latter is that many people will get inspired in pretty similar ways, > and the result of that you can hear on the radio. > > i have worked on quite a few proprietary DAWs, and i still can't believe > the fiery hoops they make me jump through. > > just this week i've had to decide to do a short film sound do-over on > $PROPRIETARY_TOOL, because i can't import the EDL into ardour (yet), and > manually syncing 75 clips is not an option (no timecode on location, > unfortunately, and now all i have is an OMF file). i thought, hey, let's > import the session and bounce it out with all regions expanded, so i can > finish the job on ardour. alas, $PT decides to bail out halfway through the > export with a totally unhelpful error message. sigh. now i'm stuck with it. > > latest brickwall i ran into is that $PT (after taking 10 years to support > native audio hardware at all) has this totally arbitrary limit of 32 > channels of simultaneous native recording. you need more? buy the box. oh, > you already have far superiour audio hardware and the studio is running > dante? too bad. buy the box, and buy a dante bridge. > right, that's the company which has been too stupid to get their > accounting done properly for more than two years and got kicked out of > NASDAQ for that. now that's two epic facepalms for the product that is > responsible for most of the really professional content out there, whatever > that means. > > which, to conclude this little rant, makes it very hard for me indeed to > avoid that fuzzy feeling of intellectual superiority, even though i agree > it is a bad habit. > > > > -- > J?rn Nettingsmeier > Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 > > Meister f?r Veranstaltungstechnik (B?hne/Studio) > Tonmeister VDT > > http://stackingdwarves.net > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david at olofson.net Mon Mar 31 16:09:23 2014 From: david at olofson.net (David Olofson) Date: Mon, 31 Mar 2014 18:09:23 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331175505.71f4ef69@eeyore> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331175505.71f4ef69@eeyore> Message-ID: On Mon, Mar 31, 2014 at 5:55 PM, Philipp ?berbacher wrote: [...] >> "Someone who makes pleasant sounds in an organized fashion, using >> basically anything." > > Now that's interesting. Why exactly does it need to be pleasant sound, > and what do you mean by that? Pleasant to whom? Good point. Some music definitely isn't meant to sound pleasant to anyone! Many examples of that in movie and video game sound tracks. > I think my definition would be even more general: "Someone who makes > sound on purpose." > Maybe an even more extreme definition is needed: "Someone who makes > sound." Well, it depends on the motivation for coming up with a definition in the first place. It could be as general as "anyone who makes sound, even accidentally" or as specific as "someone who makes commercially viable music in the currently most popular genre." If we try to come up with a definition only for the sake of having a definition, I'm afraid all we get is infinite recursion. ;-) -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------' From egor.sanin at gmail.com Mon Mar 31 16:13:50 2014 From: egor.sanin at gmail.com (Egor Sanin) Date: Mon, 31 Mar 2014 12:13:50 -0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331175505.71f4ef69@eeyore> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331175505.71f4ef69@eeyore> Message-ID: On 3/31/14, Philipp ?berbacher wrote: >> "Someone who makes pleasant sounds in an organized fashion, using >> basically anything." > > Now that's interesting. Why exactly does it need to be pleasant sound, > and what do you mean by that? Pleasant to whom? > > I think my definition would be even more general: "Someone who makes > sound on purpose." > Maybe an even more extreme definition is needed: "Someone who makes > sound." On those dreary, sluggish mornings when life forcefully drags you out from under that impossibly comfortable blanket, the sound of coffee beens being ground to that perfect consistency, followed by the sound of water, permeated by live-giving oils and solids, pouring into a cup is music to my ears. I will say: "Barrista, you are a musician!" From james at jwm-art.net Mon Mar 31 16:17:20 2014 From: james at jwm-art.net (James Morris) Date: Mon, 31 Mar 2014 16:17:20 +0000 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331103302.GA2641@gjcp.net> References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> Message-ID: <20140331161720.34618500@Scrapyard.lan> On Mon, 31 Mar 2014 11:33:02 +0100 Gordon JC Pearce wrote: > On Mon, Mar 31, 2014 at 12:23:15PM +0200, Robin Gareus wrote: > > On 03/30/2014 03:10 PM, rosea.grammostola wrote: > > > > > My conclusion so far is that Bitwig gives you what Linuxaudio lack > > > too often, smooth workflow and 'completeness' of features. This > > > is a major thing for people who want to make music! > > > > only for some kind of music. > > > > Bitwig is just a toy compared to for example Csound and > > Supercollider. > > > > But Csound and Supercollider are not suitable for making music. > They're fine if you're some kind of autistic savant computer genius, > but utterly fucking useless if you're a musician. > From murks at tuxfamily.org Mon Mar 31 16:25:20 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 31 Mar 2014 18:25:20 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331175505.71f4ef69@eeyore> Message-ID: <20140331182520.4f27a475@eeyore> On Mon, 31 Mar 2014 18:09:23 +0200 David Olofson wrote: > On Mon, Mar 31, 2014 at 5:55 PM, Philipp ?berbacher > wrote: [...] > >> "Someone who makes pleasant sounds in an organized fashion, using > >> basically anything." > > > > Now that's interesting. Why exactly does it need to be pleasant > > sound, and what do you mean by that? Pleasant to whom? > > Good point. Some music definitely isn't meant to sound pleasant to > anyone! Many examples of that in movie and video game sound tracks. I simply wanted to include noise musicians ;) It's fascinating to me to hear sounds and not being able to tell how they were produced and consequently being unable to reproduce it. It's also fascinating that this sort of music, being 'outside' of conventional musical systems is also neither 'right' nor 'wrong', you can just like or dislike it. As reference, I'm thinking about this sort of stuff I have the music of Maurizio Bianchi in mind, not some watered-down stuff like power noise, noise rock or stuff like that. https://archive.org/details/mir017 > > I think my definition would be even more general: "Someone who makes > > sound on purpose." > > Maybe an even more extreme definition is needed: "Someone who makes > > sound." > > Well, it depends on the motivation for coming up with a definition in > the first place. It could be as general as "anyone who makes sound, > even accidentally" or as specific as "someone who makes commercially > viable music in the currently most popular genre." > > If we try to come up with a definition only for the sake of having a > definition, I'm afraid all we get is infinite recursion. ;-) I don't see that infinite recursion here. If I make one definition for the sake of having a definition I can stop right there, although I could also go on indefinitely, which I guess is what you meant :>. With the two definitions above I had woodpeckers and various other birds in mind. I also find the sounds my dishes sometimes make when they slide into the sink, under water and hit the wall of the sink rather interesting. Regards, Philipp From robin at linuxaudio.org Mon Mar 31 17:26:54 2014 From: robin at linuxaudio.org (Robin Gareus) Date: Mon, 31 Mar 2014 19:26:54 +0200 Subject: [LAU] LAC'14 program announcement Message-ID: <5339A55E.3020803@linuxaudio.org> Hi all, We're excited about the upcoming Linux Audio Conference, featuring a tightly packed diverse schedule with 77 events by over 100 persons! The conference schedule has just been published at http://lac.linuxaudio.org/2014/program Apart from presentations, there are workshops, a poster-session and five concerts in 3 days. The 4th day of the conference is dedicated to an excursion: http://lac.linuxaudio.org/2014/excursion a simple overview of the complete schedule can be found in the printable version of the program. If you want to attend and have not yet done so, please register at http://lac.linuxaudio.org/2014/registration Looking forward to seeing you all in May at the ZKM. yours truly, robin - LAC'14 team From nettings at stackingdwarves.net Mon Mar 31 17:37:13 2014 From: nettings at stackingdwarves.net (=?UTF-8?B?SsO2cm4gTmV0dGluZ3NtZWllcg==?=) Date: Mon, 31 Mar 2014 19:37:13 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140329162926.GE4434@linuxaudio.org> <5337386B.5070902@gmail.com> <533786DA.4020208@gmail.com> <533817DB.9090503@gmail.com> <53394213.5080701@gareus.org> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <20140331175505.71f4ef69@eeyore> Message-ID: <5339A7C9.3050406@stackingdwarves.net> On 03/31/2014 06:13 PM, Egor Sanin wrote: > On 3/31/14, Philipp ?berbacher wrote: >>> "Someone who makes pleasant sounds in an organized fashion, using >>> basically anything." >> >> Now that's interesting. Why exactly does it need to be pleasant sound, >> and what do you mean by that? Pleasant to whom? >> >> I think my definition would be even more general: "Someone who makes >> sound on purpose." >> Maybe an even more extreme definition is needed: "Someone who makes >> sound." > > On those dreary, sluggish mornings when life forcefully drags you out > from under that impossibly comfortable blanket, the sound of coffee > beens being ground to that perfect consistency, followed by the sound > of water, permeated by live-giving oils and solids, pouring into a cup > is music to my ears. I will say: "Barrista, you are a musician!" a slightly more mundane process (percolator), but nonetheless a modern classic: http://stackingdwarves.net/download/kai_vehmanen-making_coffee.mp3 hat tip to kai vehmanen, creator of ecasound. we used this as a stream outage clip for a LAC many years back, people were almost disappointed when the stream would come back :) kai, if you read this: how could you allow this tune to fall off the net? -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From silvain at freeshell.de Mon Mar 31 18:13:30 2014 From: silvain at freeshell.de (F. Silvain) Date: Mon, 31 Mar 2014 20:13:30 +0200 (CEST) Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <5338648C.6020405@autostatic.com> References: <5338648C.6020405@autostatic.com> Message-ID: <1403312010480.6562@freeshell.de> Jeremy Jongepier, Mar 30 2014: > Dear all, > > Finally got around finishing a new track. And it's just 65BPM so no four > to the floor this time. > > http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg Nice bit of music, very smooth. The guitar distracted me though. Perhaps a little too much reverb for my ears especialy emphasizing the sliding noise in the chorus. Still taken into the playlist. Nice atmosphere. Thank you, Jeremy. From willgodfrey at musically.me.uk Mon Mar 31 18:20:19 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 31 Mar 2014 19:20:19 +0100 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <20140331143323.GD21111@linuxaudio.org> References: <20140331092227.GA31837@linuxaudio.org> <1396266539.568.3.camel@archlinux> <20140331143323.GD21111@linuxaudio.org> Message-ID: <20140331192019.54d4e3ab@debian> On Mon, 31 Mar 2014 14:33:23 +0000 Fons Adriaensen wrote: > On Mon, Mar 31, 2014 at 01:48:59PM +0200, Ralf Mardorf wrote: > > > J?rn, Fons, was there the need to reply detailed today? Tomorrow is the > > first of April and it would have been nice to wait with detailed > > explanations until tomorrow and to correct them the day after tomorrow. > > Because tomorrow is my birthday, but of course nobody will believe that. > > Ciao, > But were you born before midday? :) -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From jeremy at autostatic.com Mon Mar 31 18:44:04 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 20:44:04 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <1403312010480.6562@freeshell.de> References: <5338648C.6020405@autostatic.com> <1403312010480.6562@freeshell.de> Message-ID: <5339B774.9050106@autostatic.com> On 03/31/2014 08:13 PM, F. Silvain wrote: > Jeremy Jongepier, Mar 30 2014: > >> Dear all, >> >> Finally got around finishing a new track. And it's just 65BPM so no four >> to the floor this time. >> >> http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg >> > Nice bit of music, very smooth. The guitar distracted me though. Perhaps > a little too much reverb for my ears especialy emphasizing the sliding > noise in the chorus. Still taken into the playlist. Nice atmosphere. > Thank you, Jeremy. You're welcome and thanks for the criticism. The reverb is indeed over the top but that was actually the sound I was looking for. I could've done something about the sliding noises but it didn't really bother me. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From willgodfrey at musically.me.uk Mon Mar 31 18:49:45 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 31 Mar 2014 19:49:45 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> <201403311003.26792.zotz@100jamz.com> Message-ID: <20140331194945.2d100826@debian> On Mon, 31 Mar 2014 18:23:02 +0400 Alexandre Prokoudine wrote: > Make a simple test. Try doing the most basic thing in existing DAWs on > Linux, including Bitwig -- getting sound out of a MIDI track -- and > count steps, then multiply it by 20 tracks to get a better idea of > boring work you need to do every time (Hint: A3 will win, because I > specifically bugged Paul about it). > > Alexandre Sorry. I don't recognise this scenario. I turn on the 'big switch' and computer, auxiliaries and amp power up in sequence. For a new composition I click on my yoshimi icon followed by the rosegarden one. I select an instrument for yoshimi and play a few times then hit record on rosegarden. Switch both to track 2 and repeat. Repeat for 'n'. Save both the yoshimi parameter set and the rosegarden file. To work on an existing track I simply click on the yoshimi and rosegarden files to pull up the programs with the files loaded and ready to go. How is that taxing? Actually this *very* little routine is exactly right to get me in the 'zone'. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Mon Mar 31 18:51:31 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 31 Mar 2014 19:51:31 +0100 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <5338648C.6020405@autostatic.com> References: <5338648C.6020405@autostatic.com> Message-ID: <20140331195131.5a951380@debian> On Sun, 30 Mar 2014 20:38:04 +0200 Jeremy Jongepier wrote: > Dear all, > > Finally got around finishing a new track. And it's just 65BPM so no four > to the floor this time. > > http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg Very nice work. Looks like a relaxing place to be as well, so a good match then :) -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Mon Mar 31 19:00:25 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 31 Mar 2014 20:00:25 +0100 Subject: [LAU] [LAD] LAC'14 program announcement In-Reply-To: <5339A55E.3020803@linuxaudio.org> References: <5339A55E.3020803@linuxaudio.org> Message-ID: <20140331200025.5cbbfbb9@debian> On Mon, 31 Mar 2014 19:26:54 +0200 Robin Gareus wrote: > Hi all, > > We're excited about the upcoming Linux Audio Conference, featuring a > tightly packed diverse schedule with 77 events by over 100 persons! > > The conference schedule has just been published at > http://lac.linuxaudio.org/2014/program > > Apart from presentations, there are workshops, a poster-session and five > concerts in 3 days. The 4th day of the conference is dedicated to an > excursion: http://lac.linuxaudio.org/2014/excursion a simple overview of > the complete schedule can be found in the printable version of the program. > > If you want to attend and have not yet done so, please register at > http://lac.linuxaudio.org/2014/registration > > Looking forward to seeing you all in May at the ZKM. > > yours truly, > robin - LAC'14 team Looks good. I'm all booked and paid up this year so it'd take a disaster to keep me away this time! -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Mon Mar 31 19:04:48 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 31 Mar 2014 20:04:48 +0100 Subject: [LAU] a little remix In-Reply-To: <5338E53B.1060703@woh.rr.com> References: <5338E53B.1060703@woh.rr.com> Message-ID: <20140331200448.58d617ec@debian> On Sun, 30 Mar 2014 23:47:07 -0400 Dave Phillips wrote: > Greetings, > > https://soundcloud.com/davephillips69/i-received-a-letter > > 1'49" of downloadable CC-licensed all-original mashed-up words and > music, IIRC. > > Best, > > dp Well that was different. Interesting but i really don't know what to make of it - I guess I'm too 'mainstream' :? -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From alexandre.prokoudine at gmail.com Mon Mar 31 19:15:43 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 31 Mar 2014 23:15:43 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <20140331194945.2d100826@debian> References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> <201403311003.26792.zotz@100jamz.com> <20140331194945.2d100826@debian> Message-ID: 31 ????? 2014 ?. 22:50 ???????????? "Will Godfrey" < willgodfrey at musically.me.uk> ???????: > Sorry. I don't recognise this scenario. > > I turn on the 'big switch' and computer, auxiliaries and amp power up in > sequence. > For a new composition I click on my yoshimi icon followed by the rosegarden one. > I select an instrument for yoshimi and play a few times then hit record on > rosegarden. > Switch both to track 2 and repeat. Repeat for 'n'. > Save both the yoshimi parameter set and the rosegarden file. > > To work on an existing track I simply click on the yoshimi and rosegarden files > to pull up the programs with the files loaded and ready to go. I, however, recognize your scenario all too well. There's no night where I do not go to sleep without praying to lord to deliver me from the dark ages of linux audio where I needed to do extra jack connection plumbing and manually restoring presets :) Alexandre -------------- next part -------------- An HTML attachment was scrubbed... URL: From james at jwm-art.net Mon Mar 31 19:19:57 2014 From: james at jwm-art.net (James Morris) Date: Mon, 31 Mar 2014 19:19:57 +0000 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <5338648C.6020405@autostatic.com> References: <5338648C.6020405@autostatic.com> Message-ID: <20140331191957.63742863@Scrapyard.lan> On Sun, 30 Mar 2014 20:38:04 +0200 Jeremy Jongepier wrote: > Dear all, > > Finally got around finishing a new track. And it's just 65BPM so no > four to the floor this time. > > http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg Where's the bass? regards, James. > > This song is about one of the most beautiful places I know on this > planet, Cala Del Aceite in the most southern part of Spain > (http://www.conilplaya.com/fotos/playasdeconil/caladelaceite/playaCaladelAceiteConil.htm). > > Tools used: > * Qtractor for recording and mixing > * seq24 for sequencing > * The necessary plugins: > - drumkv1 to hold the drum samples (drum samples are all from > http://samples.kb6.de/) > - a lot of plugins that are part of Distrho or Carla: Noize Maker, > Tal Reverb III, ZynAddSubFX-LV2, Nekobi > - MDA subsynth > - FluidSynth DSSI for the Rhodes > - linuxDSP plugins (EQ500, DYN500, MBC2B on the master bus) > - Calf Vintage Delay > - LADSPA comb filter, Fast Lookahead Limiter > - GxZitaReverb > > The background vocals for the choruses are sung by my wife. The ocean > sample is from Freesound: > http://www.freesound.org/people/dobroide/sounds/93653/ C?diz is pretty > close to Conil, hence the choice. > > Lyrics > ------ > > Making promises that I can't keep > It's pushing me, pushing me into a deep > State of sadness, state of doubt > A state of awareness I can't live without > > Making mistakes, it's so hard to bear > It's driving me, driving me to a point where > I can't escape, I can't shy away > From the daemons I refuse to obey > > All is forgiven, all is well... > > Awaiting the day that I'll be relieved > From this burden, this burden that has grieved > So many loved ones, so many friends > All the people on which I depend > > Stand up, act now, it's time for a change > Lingering won't help, help to rearrange > The current imbalance, the current state > Of things so rush now don't hesitate > > All is forgiven, all is well > > ------ > From jeremy at autostatic.com Mon Mar 31 19:22:51 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 21:22:51 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <20140331195131.5a951380@debian> References: <5338648C.6020405@autostatic.com> <20140331195131.5a951380@debian> Message-ID: <5339C08B.1060608@autostatic.com> On 03/31/2014 08:51 PM, Will Godfrey wrote: > On Sun, 30 Mar 2014 20:38:04 +0200 > Jeremy Jongepier wrote: > >> Dear all, >> >> Finally got around finishing a new track. And it's just 65BPM so no four >> to the floor this time. >> >> http://theinfiniterepeat.com/music/the_infinite_repeat-cala_del_aceite.ogg > > Very nice work. Looks like a relaxing place to be as well, so a good match > then :) > Thanks Will! And it sure is a relaxing place, especially given the fact that most Spanish coastal regions are crowded. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From jeremy at autostatic.com Mon Mar 31 19:28:05 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 21:28:05 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <20140331191957.63742863@Scrapyard.lan> References: <5338648C.6020405@autostatic.com> <20140331191957.63742863@Scrapyard.lan> Message-ID: <5339C1C5.9010505@autostatic.com> On 03/31/2014 09:19 PM, James Morris wrote: > Where's the bass? > > regards, > James. I consider this an insult towards Nekobi ;) Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From dlphillips at woh.rr.com Mon Mar 31 19:29:15 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 31 Mar 2014 15:29:15 -0400 Subject: [LAU] a little remix In-Reply-To: <20140331200448.58d617ec@debian> References: <5338E53B.1060703@woh.rr.com> <20140331200448.58d617ec@debian> Message-ID: <5339C20B.1030005@woh.rr.com> On 03/31/2014 03:04 PM, Will Godfrey wrote: > On Sun, 30 Mar 2014 23:47:07 -0400 > Dave Phillips wrote: > >> Greetings, >> >> https://soundcloud.com/davephillips69/i-received-a-letter >> >> 1'49" of downloadable CC-licensed all-original mashed-up words and >> music, IIRC. >> >> Best, >> >> dp > Well that was different. Interesting but i really don't know what to make of it > - I guess I'm too 'mainstream' :? > Thanks for lending your ears, Will. You get kudos for giving it a shot. :) Best always, dp From wizardofgosz at gmail.com Mon Mar 31 19:29:58 2014 From: wizardofgosz at gmail.com (Ricardus Vincente) Date: Mon, 31 Mar 2014 15:29:58 -0400 Subject: [LAU] a little remix In-Reply-To: <20140331200448.58d617ec@debian> References: <5338E53B.1060703@woh.rr.com> <20140331200448.58d617ec@debian> Message-ID: <5339C236.4050109@gmail.com> On 03/31/2014 03:04 PM, Will Godfrey wrote: > Dave Phillips wrote: > >> Greetings, >> >> https://soundcloud.com/davephillips69/i-received-a-letter >> >> 1'49" of downloadable CC-licensed all-original mashed-up words and >> music, IIRC. >> >> Best, >> >> dp Why remix this? You already had the ultimate mix of this song. :-P :-P Rich... From dlphillips at woh.rr.com Mon Mar 31 19:37:33 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 31 Mar 2014 15:37:33 -0400 Subject: [LAU] a little remix In-Reply-To: <5339C236.4050109@gmail.com> References: <5338E53B.1060703@woh.rr.com> <20140331200448.58d617ec@debian> <5339C236.4050109@gmail.com> Message-ID: <5339C3FD.6040408@woh.rr.com> On 03/31/2014 03:29 PM, Ricardus Vincente wrote: > On 03/31/2014 03:04 PM, Will Godfrey wrote: > >> Dave Phillips wrote: >> >>> Greetings, >>> >>> https://soundcloud.com/davephillips69/i-received-a-letter >>> >>> 1'49" of downloadable CC-licensed all-original mashed-up words and >>> music, IIRC. >>> >>> Best, >>> >>> dp > Why remix this? You already had the ultimate mix of this song. :-P > > :-P > > Rich... I wanted the harmonies. Bada-boom ! Btw, I failed to note that it's also non-GMO 100% USDA-certified organic. dp From james at jwm-art.net Mon Mar 31 20:16:43 2014 From: james at jwm-art.net (James Morris) Date: Mon, 31 Mar 2014 20:16:43 +0000 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <5339C1C5.9010505@autostatic.com> References: <5338648C.6020405@autostatic.com> <20140331191957.63742863@Scrapyard.lan> <5339C1C5.9010505@autostatic.com> Message-ID: <20140331201643.22df9788@Scrapyard.lan> On Mon, 31 Mar 2014 21:28:05 +0200 Jeremy Jongepier wrote: > On 03/31/2014 09:19 PM, James Morris wrote: > > Where's the bass? > > > > regards, > > James. > > I consider this an insult towards Nekobi ;) > Sorry! How then do you consider I had no idea what Nekobi was from preliminary research, I had almost settled on concluding it was a reference to a Nekobi character to which this song was somehow dedicated until an image search revealed all! Not even in the AUR (searched for nekobi). James. > Jeremy > From sakrecoer at gmail.com Mon Mar 31 21:01:02 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Mon, 31 Mar 2014 23:01:02 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> <201403311003.26792.zotz@100jamz.com> <20140331194945.2d100826@debian> Message-ID: If I have to apply a "we" to this subject, then probably Bitwig spends very little time defining the boundaries of music. :) Else, all i can express are my dreams occurring when i envision what a community like ours could achieve in the best of the worlds. Have a delicious week ya'all! -- Set Hallstr?m AKA Sakrecoer http://sakrecoer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From gordonjcp at gjcp.net Mon Mar 31 21:03:28 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Mon, 31 Mar 2014 22:03:28 +0100 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <533786DA.4020208@gmail.com> <201403311003.26792.zotz@100jamz.com> <20140331194945.2d100826@debian> Message-ID: <20140331210328.GA5080@gjcp.net> On Mon, Mar 31, 2014 at 11:15:43PM +0400, Alexandre Prokoudine wrote: > 31 ????? 2014 ?. 22:50 ???????????? "Will Godfrey" < > willgodfrey at musically.me.uk> ???????: > > > Sorry. I don't recognise this scenario. > > > > I turn on the 'big switch' and computer, auxiliaries and amp power up in > > sequence. > > For a new composition I click on my yoshimi icon followed by the > rosegarden one. > > I select an instrument for yoshimi and play a few times then hit record on > > rosegarden. > > Switch both to track 2 and repeat. Repeat for 'n'. > > Save both the yoshimi parameter set and the rosegarden file. > > > > To work on an existing track I simply click on the yoshimi and rosegarden > files > > to pull up the programs with the files loaded and ready to go. > > I, however, recognize your scenario all too well. There's no night where I > do not go to sleep without praying to lord to deliver me from the dark ages > of linux audio where I needed to do extra jack connection plumbing and > manually restoring presets :) > > Alexandre This is exactly why I gave up using computers for music. Now my workflow is "turn on the S50 or W30 depending what mood I'm in, stick in a boot disk and wait 30 seconds, then get programming". Until someone comes up with a sequencer that works and runs on a PC, I'll stick with that. -- Gordonjcp MM0YEQ From len at ovenwerks.net Mon Mar 31 21:03:00 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 31 Mar 2014 14:03:00 -0700 (PDT) Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: <1396272125.568.38.camel@archlinux> References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <201403311509.50956.gerhard.zintel@web.de> <1396272125.568.38.camel@archlinux> Message-ID: On Mon, 31 Mar 2014, Ralf Mardorf wrote: > Music isn't a competition about smartness. I think that is what was being said. Music today seems to be no longer about communicating anything at all... merely soundiing somewhat pleasant, or showing off some vocal (or other) gymnastics seems to be most of it. Actually communicating what is in the artists heart is rare and too hard for a money making organization to quantify, so they have gone for what is quantifiable: Take a song that is already a hit, use a producer that we know makes us money, get someone with a strong voice who can hit all the notes and doesn't care too much what they sing so long as they get paid. "Music" making tools that help that process are going to be what sw makers are going to look to for their bread and butter. It may be possible to make actual music with those tools too... -- Len Ovens www.ovenwerks.net From louigi.verona at gmail.com Mon Mar 31 21:09:31 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Tue, 1 Apr 2014 01:09:31 +0400 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <201403311509.50956.gerhard.zintel@web.de> <1396272125.568.38.camel@archlinux> Message-ID: Gentlemen, this is going nowhere. There is no use debating what is music and what is not. Seriously. This is not what we can learn from Bitwig. On Tue, Apr 1, 2014 at 1:03 AM, Len Ovens wrote: > > On Mon, 31 Mar 2014, Ralf Mardorf wrote: > > Music isn't a competition about smartness. >> > > I think that is what was being said. Music today seems to be no longer > about communicating anything at all... merely soundiing somewhat pleasant, > or showing off some vocal (or other) gymnastics seems to be most of it. > Actually communicating what is in the artists heart is rare and too hard > for a money making organization to quantify, so they have gone for what is > quantifiable: Take a song that is already a hit, use a producer that we > know makes us money, get someone with a strong voice who can hit all the > notes and doesn't care too much what they sing so long as they get paid. > > "Music" making tools that help that process are going to be what sw makers > are going to look to for their bread and butter. It may be possible to make > actual music with those tools too... > > -- > Len Ovens > www.ovenwerks.net > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Mar 31 21:20:08 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 31 Mar 2014 23:20:08 +0200 Subject: [LAU] The Infinite Repeat - Cala Del Aceite In-Reply-To: <20140331201643.22df9788@Scrapyard.lan> References: <5338648C.6020405@autostatic.com> <20140331191957.63742863@Scrapyard.lan> <5339C1C5.9010505@autostatic.com> <20140331201643.22df9788@Scrapyard.lan> Message-ID: <5339DC08.1030200@autostatic.com> On 03/31/2014 10:16 PM, James Morris wrote: > On Mon, 31 Mar 2014 21:28:05 +0200 > Jeremy Jongepier wrote: > >> On 03/31/2014 09:19 PM, James Morris wrote: >>> Where's the bass? >>> >>> regards, >>> James. >> >> I consider this an insult towards Nekobi ;) >> > > Sorry! How then do you consider I had no idea what Nekobi was from > preliminary research, I had almost settled on concluding it was a > reference to a Nekobi character to which this song was somehow > dedicated until an image search revealed all! Not even in the AUR > (searched for nekobi). > > James. Hi James, Nekobi is a fork of Nekobee and part of the Distrho project. And it's in AUR: https://aur.archlinux.org/packages/distrho-lv2-git/ Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From list at nilsgey.de Mon Mar 31 21:21:36 2014 From: list at nilsgey.de (Nils) Date: Mon, 31 Mar 2014 23:21:36 +0200 Subject: [LAU] Bitwig: what we can learn from it In-Reply-To: References: <5336A119.3080502@gmail.com> <20140331103302.GA2641@gjcp.net> <20140331112645.GA26586@linuxaudio.org> <201403311509.50956.gerhard.zintel@web.de> <1396272125.568.38.camel@archlinux> Message-ID: <5339DC60.8090406@nilsgey.de> On 31.03.2014 23:09, Louigi Verona wrote: > Gentlemen, this is going nowhere. > > There is no use debating what is music and what is not. Seriously. > This is not what we can learn from Bitwig. +1000 Internets. From fons at linuxaudio.org Mon Mar 31 21:32:24 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 31 Mar 2014 21:32:24 +0000 Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <20140331192019.54d4e3ab@debian> References: <20140331092227.GA31837@linuxaudio.org> <1396266539.568.3.camel@archlinux> <20140331143323.GD21111@linuxaudio.org> <20140331192019.54d4e3ab@debian> Message-ID: <20140331213224.GA18937@linuxaudio.org> On Mon, Mar 31, 2014 at 07:20:19PM +0100, Will Godfrey wrote: > But were you born before midday? :) I don't remember (it's been some years...) Why would it matter ? Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From thomas at residuum.org Mon Mar 31 22:43:03 2014 From: thomas at residuum.org (Thomas Mayer) Date: Tue, 01 Apr 2014 00:43:03 +0200 Subject: [LAU] music from image metadata In-Reply-To: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> References: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> Message-ID: <5339EF77.80504@residuum.org> Hi, On 31.03.2014 11:11, Patrick Shirkey wrote: > Hi, > > Can anyone think of a way to automate the creation of a music track from > the metadata embedded in an image track? I haven't worked with metadata yet, but did some experiments in sonification. Here is what I would do: Analyse the format of EXIF data. What is actually encoded? What is varying from image to image, camera to camera? Try to get a fair sample size to see the data. Try to convert the data to numbers, that can be interpreted as notes, frequency, duration, volume. If not all parameters can be set, then use some sane defaults. My guess is, that there is to little data to really interpret in EXIF, and that data is to disparate to create a melody: To take the example from Wikipedia (http://en.wikipedia.org/wiki/Exchangeable_image_file_format#Example), how many notes can you create from the data to create a melody using an algorithm, that you can explain to users in just a few sentences? If not all data can be made to create a melody, then create one note from each image and use series of images, a slideshow of sorts. You could combine that with converting the images to sound, interpreting the y-axis as frequency and x-axis as time, similar to "Sheet music" by Johannes Kreidler: http://www.youtube.com/watch?v=vdbpJmsaNAw An example for Twitter sonification is included in my Pd extension PuREST JSON: http://ix.residuum.org/pd/purest_json.html Here's an example, that uses the returned data from a search, including a description on the algorithm to generate the sound: https://soundcloud.com/residuum/twitter-sonification Have fun, Thomas -- "From the perspective of communication analysis, government is not an instrument of law and order, but of law and disorder." (Gracchus Gruad in: Robert Shea & Robert A. Wilson, The Golden Apple) http://www.residuum.org/ From thomas at residuum.org Mon Mar 31 22:49:08 2014 From: thomas at residuum.org (Thomas Mayer) Date: Tue, 01 Apr 2014 00:49:08 +0200 Subject: [LAU] music from image metadata In-Reply-To: <5339EF77.80504@residuum.org> References: <55034.86.107.254.57.1396257066.squirrel@boosthardware.com> <5339EF77.80504@residuum.org> Message-ID: <5339F0E4.3000407@residuum.org> Hi, On 01.04.2014 00:43, Thomas Mayer wrote: > Hi, > > On 31.03.2014 11:11, Patrick Shirkey wrote: >> Hi, >> >> Can anyone think of a way to automate the creation of a music track from >> the metadata embedded in an image track? > > I haven't worked with metadata yet, but did some experiments in > sonification. Here is what I would do: > > Analyse the format of EXIF data. What is actually encoded? What is > varying from image to image, camera to camera? Try to get a fair sample > size to see the data. > > Try to convert the data to numbers, that can be interpreted as notes, > frequency, duration, volume. If not all parameters can be set, then use > some sane defaults. Some more information: Have a look at ExifTool to get the data: http://petapixel.com/2012/05/25/hack-your-exif-data-from-the-command-line-five-fun-uses-for-exiftool/ And here are some examples for ExifTool usage: http://petapixel.com/2012/05/25/hack-your-exif-data-from-the-command-line-five-fun-uses-for-exiftool/ Have fun, Thomas -- "As long as people kept worrying that the machines were taking over, they wouldn't notice what was really happening. Which was that the programmers were taking over." (Robert Anton Wilson - The Homing Pidgeons) http://www.residuum.org/ From len at ovenwerks.net Mon Mar 31 23:03:37 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 31 Mar 2014 16:03:37 -0700 (PDT) Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: References: Message-ID: On Mon, 31 Mar 2014, Moshe Werner wrote: > I can't really talk about high end desks, but in my midrange Soundtracs > Topaz desk (which has a quite nice EQ)? > it is certainly serial. I've attached a pdf of the EQ section schematics. > Would be interesting to know how others did it. Ah yes, makes sense. The parallel circuits I have seen are simple low/mid/high controls with no q or frequency control. One network between opamps or in the feedback line. Parts count probably had more to do with design than quality. Thankyou. -- Len Ovens www.ovenwerks.net From len at ovenwerks.net Mon Mar 31 23:11:29 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 31 Mar 2014 16:11:29 -0700 (PDT) Subject: [LAU] Filter arrangement - was- Bitwig: what we can learn from it In-Reply-To: <53391B21.9070509@stackingdwarves.net> References: <53391B21.9070509@stackingdwarves.net> Message-ID: On Mon, 31 Mar 2014, J?rn Nettingsmeier wrote: > analog parallel filters can have some benefits wrt signal-to-noise ratio, but > you need to get the phase alignment right. for digital, it's mostly an > unnecessary complicaton, The s/n angle makes sense. I have a box with 8 parametric eqs in it, a signal gets pretty noisy when going one into another. (cheap box... noise noticable even on the first filter) I have used it on grungy sounds like guitar that already had some noise... but never since I have had digital stuff to play with. -- Len Ovens www.ovenwerks.net