From p8rpp at aol.com Sat Feb 1 00:00:57 2014 From: p8rpp at aol.com (Peter P.) Date: Sat, 1 Feb 2014 01:00:57 +0100 Subject: [LAU] HDSP wrong firmware gets loaded (digi- vs multiface) Message-ID: <20140201000054.GB1804@aol.de> Hi list, hdsploader (via hotplug) on my box seems to load firmware at will. Sometimes it loads the Digiface firmware into my Multiface, sometimes it loads the correct one. I suppose I could simply delete the digiface firmware from the path, but what is it that makes it load the wrong one? correct one: kernel: [98033.160982] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] kernel: [98033.161012] pci 0000:05:00.0: no hotplug settings from platform kernel: [98033.161264] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) kernel: [98033.467674] Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations kernel: [98033.474630] snd_hdsp 0000:05:00.0: firmware: agent loaded multiface_firmware_rev11.bin into memory wrong one: kernel: [98146.327251] Hammerfall-DSP: loading firmware kernel: [98149.392459] Hammerfall-DSP: finished firmware loading kernel: [98243.384790] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] kernel: [98243.384824] pci 0000:05:00.0: no hotplug settings from platform kernel: [98243.385118] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) kernel: [98243.689751] snd_hdsp 0000:05:00.0: firmware: agent loaded digiface_firmware_rev11.bin into memory Does someone else have the same issue? best, P From nettings at stackingdwarves.net Sat Feb 1 14:36:31 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sat, 01 Feb 2014 15:36:31 +0100 Subject: [LAU] Strange change of sample rate upon connecting mixing desk In-Reply-To: <20140131235929.GA1804@aol.de> References: <20140131235929.GA1804@aol.de> Message-ID: <52ED066F.1060703@stackingdwarves.net> On 02/01/2014 12:59 AM, Peter P. wrote: > Dear list, > allow me to post a question here that I already sent to alsa-user, but > without any response so far, thank you! > > ----------8<----------- > > Hi all, now here is one of the weirdest problems I have come across in > the past years. Actually it is halting my current production even, but > I am primarily interested how this error is caused (and how it can be > solved): > > The sampling rate of my RME HDSP Card is set from "Internal 44.1k" to > "Internal 48k" upon connecting a Yamaha DM1000 mixing console via usb! > > Can anyone imagine what causes this strange behavior? http://xkcd.com/1316/ other than that, no idea. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From nettings at stackingdwarves.net Sat Feb 1 14:37:46 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sat, 01 Feb 2014 15:37:46 +0100 Subject: [LAU] HDSP wrong firmware gets loaded (digi- vs multiface) In-Reply-To: <20140201000054.GB1804@aol.de> References: <20140201000054.GB1804@aol.de> Message-ID: <52ED06BA.9000905@stackingdwarves.net> On 02/01/2014 01:00 AM, Peter P. wrote: > Hi list, > > hdsploader (via hotplug) on my box seems to load firmware at will. > Sometimes it loads the Digiface firmware into my Multiface, sometimes > it loads the correct one. > > I suppose I could simply delete the digiface firmware from the path, > but what is it that makes it load the wrong one? > > correct one: > kernel: [98033.160982] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] > kernel: [98033.161012] pci 0000:05:00.0: no hotplug settings from platform > kernel: [98033.161264] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) > kernel: [98033.467674] Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations > kernel: [98033.474630] snd_hdsp 0000:05:00.0: firmware: agent loaded multiface_firmware_rev11.bin into memory > > wrong one: > kernel: [98146.327251] Hammerfall-DSP: loading firmware > kernel: [98149.392459] Hammerfall-DSP: finished firmware loading > kernel: [98243.384790] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] > kernel: [98243.384824] pci 0000:05:00.0: no hotplug settings from platform > kernel: [98243.385118] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) > kernel: [98243.689751] snd_hdsp 0000:05:00.0: firmware: agent loaded digiface_firmware_rev11.bin into memory > > Does someone else have the same issue? not yet, but i haven't used the card in two weeks. which kernel, which distro? -- 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 Sat Feb 1 14:49:25 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Sat, 01 Feb 2014 15:49:25 +0100 Subject: [LAU] Strange change of sample rate upon connecting mixing desk In-Reply-To: <20140131235929.GA1804@aol.de> References: <20140131235929.GA1804@aol.de> Message-ID: <52ED0975.5020606@ladisch.de> Peter P. wrote: > The sampling rate of my RME HDSP Card is set from "Internal 44.1k" to > "Internal 48k" upon connecting a Yamaha DM1000 mixing console via usb! In theory, this should not happen. On possible reason could be that some tool gets run for the hot-plug event, tries to configure the new device, and somehow accesses the wrong one. Does running "alsactl restore" have the same effect? If yes, then this is just the last saved mixer setting. Regards, Clemens From nettings at stackingdwarves.net Sat Feb 1 14:51:42 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sat, 01 Feb 2014 15:51:42 +0100 Subject: [LAU] HDSP wrong firmware gets loaded (digi- vs multiface) In-Reply-To: <52ED06BA.9000905@stackingdwarves.net> References: <20140201000054.GB1804@aol.de> <52ED06BA.9000905@stackingdwarves.net> Message-ID: <52ED09FE.4050007@stackingdwarves.net> On 02/01/2014 03:37 PM, J?rn Nettingsmeier wrote: > On 02/01/2014 01:00 AM, Peter P. wrote: >> Hi list, >> >> hdsploader (via hotplug) on my box seems to load firmware at will. >> Sometimes it loads the Digiface firmware into my Multiface, sometimes >> it loads the correct one. >> >> Does someone else have the same issue? > > not yet, but i haven't used the card in two weeks. > > which kernel, which distro? can't reproduce that here. kernel is 3.13.0, 05:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 3c) [ 10.879698] ALSA hdsp.c:709 Hammerfall-DSP: iobox found after 0ms! [ 10.883045] ALSA hdsp.c:898 Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations [ 10.883157] ALSA hdsp.c:817 Hammerfall-DSP: Multiface found [ 10.883901] ALSA hdsp.c:737 Hammerfall-DSP: loading firmware ... [ 14.260572] ALSA hdsp.c:769 Hammerfall-DSP: finished firmware loading -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From p8rpp at aol.com Sat Feb 1 18:13:34 2014 From: p8rpp at aol.com (Peter P.) Date: Sat, 1 Feb 2014 19:13:34 +0100 Subject: [LAU] Strange change of sample rate upon connecting mixing desk In-Reply-To: <52ED066F.1060703@stackingdwarves.net> References: <20140131235929.GA1804@aol.de> <52ED066F.1060703@stackingdwarves.net> Message-ID: <20140201181334.GA9015@aol.de> * J?rn Nettingsmeier [2014-02-01 15:36]: > On 02/01/2014 12:59 AM, Peter P. wrote: > >Dear list, > >allow me to post a question here that I already sent to alsa-user, but > >without any response so far, thank you! > > > >----------8<----------- > > > >Hi all, now here is one of the weirdest problems I have come across in > >the past years. Actually it is halting my current production even, but > >I am primarily interested how this error is caused (and how it can be > >solved): > > > >The sampling rate of my RME HDSP Card is set from "Internal 44.1k" to > >"Internal 48k" upon connecting a Yamaha DM1000 mixing console via usb! > > > >Can anyone imagine what causes this strange behavior? > > http://xkcd.com/1316/ > > other than that, no idea. Thanks, that helped a _bit_ . I am wondering if loading the snd-usb-audio driver somehow sets the sampling rate to some default value. Well, I might move this one over to alsa-dev for a last resort. What's also strange is that this happens regardless of what Sr the DM1000 is set to internally. But then there is no usb audio interface present within that mixer anyway. best, P From p8rpp at aol.com Sat Feb 1 18:14:40 2014 From: p8rpp at aol.com (Peter P.) Date: Sat, 1 Feb 2014 19:14:40 +0100 Subject: [LAU] HDSP wrong firmware gets loaded (digi- vs multiface) In-Reply-To: <52ED06BA.9000905@stackingdwarves.net> References: <20140201000054.GB1804@aol.de> <52ED06BA.9000905@stackingdwarves.net> Message-ID: <20140201181439.GB9015@aol.de> * J?rn Nettingsmeier [2014-02-01 15:37]: > On 02/01/2014 01:00 AM, Peter P. wrote: > >Hi list, > > > >hdsploader (via hotplug) on my box seems to load firmware at will. > >Sometimes it loads the Digiface firmware into my Multiface, sometimes > >it loads the correct one. > > > >I suppose I could simply delete the digiface firmware from the path, > >but what is it that makes it load the wrong one? > > > >correct one: > >kernel: [98033.160982] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] > >kernel: [98033.161012] pci 0000:05:00.0: no hotplug settings from platform > >kernel: [98033.161264] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) > >kernel: [98033.467674] Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations > >kernel: [98033.474630] snd_hdsp 0000:05:00.0: firmware: agent loaded multiface_firmware_rev11.bin into memory > > > >wrong one: > >kernel: [98146.327251] Hammerfall-DSP: loading firmware > >kernel: [98149.392459] Hammerfall-DSP: finished firmware loading > >kernel: [98243.384790] pci 0000:05:00.0: BAR 0: assigned [mem 0xf0000000-0xf000ffff] > >kernel: [98243.384824] pci 0000:05:00.0: no hotplug settings from platform > >kernel: [98243.385118] snd_hdsp 0000:05:00.0: enabling device (0000 -> 0002) > >kernel: [98243.689751] snd_hdsp 0000:05:00.0: firmware: agent loaded digiface_firmware_rev11.bin into memory > > > >Does someone else have the same issue? > > not yet, but i haven't used the card in two weeks. > > which kernel, which distro? oh, rather conservative Debian testing under 3.2.0-4-rt-amd64 But the hdsploader code is really old anyway. best, P From p8rpp at aol.com Sat Feb 1 18:15:49 2014 From: p8rpp at aol.com (Peter P.) Date: Sat, 1 Feb 2014 19:15:49 +0100 Subject: [LAU] HDSP wrong firmware gets loaded (digi- vs multiface) In-Reply-To: <52ED09FE.4050007@stackingdwarves.net> References: <20140201000054.GB1804@aol.de> <52ED06BA.9000905@stackingdwarves.net> <52ED09FE.4050007@stackingdwarves.net> Message-ID: <20140201181549.GC9015@aol.de> * J?rn Nettingsmeier [2014-02-01 15:51]: > On 02/01/2014 03:37 PM, J?rn Nettingsmeier wrote: > >On 02/01/2014 01:00 AM, Peter P. wrote: > >>Hi list, > >> > >>hdsploader (via hotplug) on my box seems to load firmware at will. > >>Sometimes it loads the Digiface firmware into my Multiface, sometimes > >>it loads the correct one. > >> > > > >>Does someone else have the same issue? > > > >not yet, but i haven't used the card in two weeks. > > > >which kernel, which distro? > > can't reproduce that here. > kernel is 3.13.0, > > 05:00.0 Multimedia audio controller: Xilinx Corporation RME > Hammerfall DSP (rev 3c) > > [ 10.879698] ALSA hdsp.c:709 Hammerfall-DSP: iobox found after 0ms! > [ 10.883045] ALSA hdsp.c:898 Hammerfall-DSP: wait for FIFO status > <= 0 failed after 30 iterations > [ 10.883157] ALSA hdsp.c:817 Hammerfall-DSP: Multiface found > [ 10.883901] ALSA hdsp.c:737 Hammerfall-DSP: loading firmware > ... > [ 14.260572] ALSA hdsp.c:769 Hammerfall-DSP: finished firmware loading Yep, in 95% of the cases the correct one is loaded as well. It is mostly after not having used the card for a longer time, or switching components (multiface vs PCMCIA/ExpressCard) around. best, P From jonetsu at teksavvy.com Sat Feb 1 23:28:39 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Sat, 1 Feb 2014 18:28:39 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone Message-ID: <20140201182839.33507395@mistral> Hello, (Sorry for crossposting from Ardour list but I figured someone here could answer faster). I clicked on 'something' in Ardour 3.5.358 top bar and now it's gone. It was hiding behind the main window, so I clicked on the small up arrow on the right of the stand-alone bar (eg. detached window for these controls) and now, it's really gone from sight. I've browser the menu options but did not find where to re-activate it. Cannot record, sicne the record button is part of the bar. I have loaded the Axiom 25 control interface, but the record button of the Axiom does not enable recording (play and rewind works). How to make the top control bar visible again ? Thanks ! From james at jwm-art.net Sat Feb 1 23:41:50 2014 From: james at jwm-art.net (James Morris) Date: Sat, 1 Feb 2014 23:41:50 +0000 Subject: [LAU] Caps Plate2x2 differences between Arch/Debian Message-ID: <20140201234150.451b7a1b@Scrapyard.lan> Has anyone else come across this issue with the LADSPA Caps Plate plugin with the effect on Arch being very much reduced compared with the amount of effect provided by the same plugin in Debian? With the plugin tail and bandwidth set to maximum, damping at a minimum, and no dry effect, silence arrives in a flash. Its a problem I've noticed for several months... James. From jonetsu at teksavvy.com Sun Feb 2 02:23:20 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Sat, 1 Feb 2014 21:23:20 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: <20140201182839.33507395@mistral> References: <20140201182839.33507395@mistral> Message-ID: <20140201212320.2062b5bb@mistral> On Sat, 1 Feb 2014 18:28:39 -0500, "jonetsu at teksavvy.com" wrote : > I clicked on 'something' in Ardour 3.5.358 top bar and now it's > gone. It was hiding behind the main window, so I clicked on the small > up arrow on the right of the stand-alone bar (eg. detached window for > these controls) and now, it's really gone from sight. I've browser > the menu options but did not find where to re-activate it. Cannot > record, sicne the record button is part of the bar. I have loaded > the Axiom 25 control interface, but the record button of the Axiom > does not enable recording (play and rewind works). > > How to make the top control bar visible again ? I certainly do not understand why it is so friggeringly obscure to make the GUI bar with the recording button and other controls and clock visble again. The whole software is useless if the main recording button cannot be seen and activated. I don't know on what I clicked to make it go away, but the remaining application screen, or a menu option somewhere should provide a way to make it visible again. The control bar became a floating window on its own, but now it's completely gone from all desktops and the main application doe snot provide a mean to make it part of the main GUI window as it was before. Maybe I have to downgrade to the previous version... hopefully the sessions newly created are compatible with the previous version ! From paul at linuxaudiosystems.com Sun Feb 2 02:53:41 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sat, 1 Feb 2014 21:53:41 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: <20140201212320.2062b5bb@mistral> References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> Message-ID: On Sat, Feb 1, 2014 at 9:23 PM, jonetsu at teksavvy.com wrote: > On Sat, 1 Feb 2014 18:28:39 -0500, > "jonetsu at teksavvy.com" wrote : > > > I clicked on 'something' in Ardour 3.5.358 top bar and now it's > > gone. It was hiding behind the main window, so I clicked on the small > > up arrow on the right of the stand-alone bar (eg. detached window for > > these controls) and now, it's really gone from sight. I've browser > > the menu options but did not find where to re-activate it. Cannot > > record, sicne the record button is part of the bar. I have loaded > > the Axiom 25 control interface, but the record button of the Axiom > > does not enable recording (play and rewind works). > > > > How to make the top control bar visible again ? > > I certainly do not understand why it is so friggeringly obscure to make > the GUI bar with the recording button and other controls and clock > visble again. The whole software is useless if the main recording > button cannot be seen and activated. I don't know on what I clicked to > make it go away, but the remaining application screen, or a menu option > somewhere should provide a way to make it visible again. The control bar > became a floating window on its own, but now it's completely gone from > all desktops and the main application doe snot provide a mean to make > it part of the main GUI window as it was before. Maybe I have to > downgrade to the previous version... hopefully the sessions newly > created are compatible with the previous version ! > (1) ardour-users is the mailing list for this, as you've noted (2) we cannot accomodate ALL window managers and their behaviour Linux offers massive flexibility with things like window managers. This comes with a price. You've already figured out (based on the mail to ardour-users) what to click on. The rest - we can't control for this and keep every Linux user happy. What window manager do you use? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonetsu at teksavvy.com Sun Feb 2 03:00:27 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Sat, 1 Feb 2014 22:00:27 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> Message-ID: <20140201220027.4473c30f@mistral> On Sat, 1 Feb 2014 21:53:41 -0500, Paul Davis wrote : > (1) ardour-users is the mailing list for this, as you've noted Yes, and I did say I'm sorry for cross-posting :) > You've already figured out (based on the mail to ardour-users) what to > click on. The rest - we can't control for this and keep every Linux > user happy. I've learned that Shift-Space starts recording, which is nice. Controlling play/rewing from the Axiom is also nice. Having the record button on the Axiom working would be also nice. > What window manager do you use? KDE that comes with Linux Mint 14. From fons at linuxaudio.org Sun Feb 2 11:54:51 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 2 Feb 2014 11:54:51 +0000 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> Message-ID: <20140202115451.GA32667@linuxaudio.org> On Sat, Feb 01, 2014 at 09:53:41PM -0500, Paul Davis wrote: > Linux offers massive flexibility with things like window managers. This > comes with a price. > > You've already figured out (based on the mail to ardour-users) what to > click on. The rest - we can't control for this and keep every Linux user > happy. * Quit Ardour. * Open ~/.config/ardour3/ardour.rc in a text editor. * Search for "transport tornoff" and set it to "no". * Save. * Restart ardour. I'd agree with the OP that Ardour should provide a way to recover torn-off windows if they ever get lost off-screen or otherwise. 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 Sun Feb 2 17:02:14 2014 From: thomas at residuum.org (Thomas Mayer) Date: Sun, 02 Feb 2014 18:02:14 +0100 Subject: [LAU] [ANN] XSL stylesheets for converting specimen banks to petri-foo banks Message-ID: <52EE7A16.80601@residuum.org> Hi, I have just finished two XSL stylesheets for converting specimen banks to the save format used by petri-foo, so a user can easily switch to petri-foo: https://github.com/residuum/specimen2petri-foo There are two stylesheets, depending on your locale settings, i.e. if your locale uses comma or dot as decimal separator (if 1/2 is saved as 0.5 or 0,5 on your system). If there are any other decimal separators out there your copy s2p_comma.xsl and replace strings "0," and "1," to your settings. Do not do that with s2p_dot.xsl, because necessary version information for XML and XSL use strings like "1.0" etc. Standard sample rate is set to 44.1 kHz, if you want to change to another value, just edit the value in line 6. If you encounter any bugs, open an issue on Github or send me an email. Thanks, Thomas -- "We left all that stuff out. If there's an error, we have this routine called panic, and when it is called, the machine crashes, and you holler down the hall, 'Hey, reboot it.'" (Dennis Ritchie) http://www.residuum.org/ From paul at linuxaudiosystems.com Sun Feb 2 17:05:45 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 2 Feb 2014 12:05:45 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: <20140202115451.GA32667@linuxaudio.org> References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> <20140202115451.GA32667@linuxaudio.org> Message-ID: On Sun, Feb 2, 2014 at 6:54 AM, Fons Adriaensen wrote: > > I'd agree with the OP that Ardour should provide a way to recover > torn-off windows if they ever get lost off-screen or otherwise. > an excellent and seemingly obvious idea. i'll add it today. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rm at mh-freiburg.de Sun Feb 2 17:57:19 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Sun, 2 Feb 2014 18:57:19 +0100 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> <20140202115451.GA32667@linuxaudio.org> Message-ID: <20140202175146.M88988@mh-freiburg.de> On Sun, 2 Feb 2014 12:05:45 -0500, Paul Davis wrote > On Sun, Feb 2, 2014 at 6:54 AM, Fons Adriaensen wrote: > > I'd agree with the OP that Ardour should provide a way to recover > torn-off windows if they ever get lost off-screen or otherwise. > > > an excellent and seemingly obvious idea. i'll add it today. After reading Fons' mail I was about to reply that one could simply save the initial view and then later on restore it. But, to my astonishment, 'goto view ...' doesn't restore UI state. I have to admit that I never used views before, but wouldn't that be the mechanism to use? BTW, my recent Ardour checkout seems to fix a longstanding issue with starting Ardour in a maximized window (the ardour log window would constantly come in front of the main window which would then be put in front of the log etc.). Thanks for the fix. Cheers RalfD ? -- R. Mattes - Hochschule fuer Musik Freiburg rm at inm.mh-freiburg.de From paul at linuxaudiosystems.com Sun Feb 2 18:01:42 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 2 Feb 2014 13:01:42 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: <20140202175146.M88988@mh-freiburg.de> References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> <20140202115451.GA32667@linuxaudio.org> <20140202175146.M88988@mh-freiburg.de> Message-ID: On Sun, Feb 2, 2014 at 12:57 PM, R. Mattes wrote: > On Sun, 2 Feb 2014 12:05:45 -0500, Paul Davis wrote > > On Sun, Feb 2, 2014 at 6:54 AM, Fons Adriaensen > wrote: > > > > I'd agree with the OP that Ardour should provide a way to recover > > torn-off windows if they ever get lost off-screen or otherwise. > > > > > > an excellent and seemingly obvious idea. i'll add it today. > > After reading Fons' mail I was about to reply that one could > simply save the initial view and then later on restore it. > But, to my astonishment, 'goto view ...' doesn't restore UI state. > I have to admit that I never used views before, but wouldn't that > be the mechanism to use? > the definition of what a "view" is can be subject to quite a bit of variation. it does not (currently) include every aspect of window configuration but is mostly about track visibility, zoom level, timeline position and vertical scroll. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonetsu at teksavvy.com Sun Feb 2 22:00:28 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Sun, 2 Feb 2014 17:00:28 -0500 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> <20140202115451.GA32667@linuxaudio.org> Message-ID: <20140202170028.0f7cd0ae@mistral> On Sun, 2 Feb 2014 12:05:45 -0500, Paul Davis wrote : > On Sun, Feb 2, 2014 at 6:54 AM, Fons Adriaensen > wrote: > > I'd agree with the OP that Ardour should provide a way to recover > > torn-off windows if they ever get lost off-screen or otherwise. > an excellent and seemingly obvious idea. i'll add it today. There's a reset option for the Appearance, eg. color scheme. Maybe a reset that not only includes colors, but default positions of menus ? One last note on the problem: today I rebooted the machine and everything was OK. I haven't modified any config file. So it was some confusion (KDE, Ardour ?) going on. The way I'm setting up musical apps is like this: on one desktop I have Ardour main screen full screen (via menu option), on another desktop I have Ardour's mixer full screen (KDE full screen window option). Hot keys makes it easy to switch desktops (and also to Renoise which is full screen on another desktop, or to Zyn and QSynth on yet another desktop). One symptom of the software 'confusion' yesterday was that the add-a-track dialog box did not pop up on the current main Ardour desktop, but on the one with the full screen mixer. I'd guess Ardour does not specify where to popup dialog boxes, and for some reason KDE popped it up on another desktop. Anyways, back to normal now. I'll make darn sure not to hit that little arrow again !! :)) From brent at keycorner.org Sun Feb 2 22:47:19 2014 From: brent at keycorner.org (Brent Busby) Date: Sun, 2 Feb 2014 16:47:19 -0600 (CST) Subject: [LAU] question about Ardour tempo Message-ID: I have an Ardour project that was recorded with Ardour setup as a Jack slave, driven by Muse. It correctly tracks the tempo from Muse at 130bpm. However, it has ended up with a tempo timeline tag of 120bpm. Is that something hardcoded? Should I change it in the Ardour project to reflect the true tempo? Why didn't it notice that it was tracking faster than that? -- + 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 paul at linuxaudiosystems.com Sun Feb 2 23:02:52 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 2 Feb 2014 18:02:52 -0500 Subject: [LAU] question about Ardour tempo In-Reply-To: References: Message-ID: On Sun, Feb 2, 2014 at 5:47 PM, Brent Busby wrote: > I have an Ardour project that was recorded with Ardour setup as a Jack > slave, driven by Muse. It correctly tracks the tempo from Muse at 130bpm. > However, it has ended up with a tempo timeline tag of 120bpm. Is that > something hardcoded? Should I change it in the Ardour project to reflect > the true tempo? Why didn't it notice that it was tracking faster than that? > the timeline tempo markers are not updated in response to external tempo information. -------------- next part -------------- An HTML attachment was scrubbed... URL: From cbannister at slingshot.co.nz Mon Feb 3 00:19:13 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Mon, 3 Feb 2014 13:19:13 +1300 Subject: [LAU] Ardour: UI bar with record button/clock, etc is gone In-Reply-To: <20140201220027.4473c30f@mistral> References: <20140201182839.33507395@mistral> <20140201212320.2062b5bb@mistral> <20140201220027.4473c30f@mistral> Message-ID: <20140203001913.GE5980@tal> On Sat, Feb 01, 2014 at 10:00:27PM -0500, jonetsu at teksavvy.com wrote: > On Sat, 1 Feb 2014 21:53:41 -0500, > Paul Davis wrote : > > > (1) ardour-users is the mailing list for this, as you've noted > > Yes, and I did say I'm sorry for cross-posting :) JFTR, and sorry for being pedantic - but what you did was multipost http://en.wikipedia.org/wiki/Crossposting "Crossposting is the act of posting the same message to multiple information channels (forums, mailing lists, or newsgroups) in such a way that reading software can relate copies of this message on different information channels. Thus reading software is showing this message only once. This is distinct from multiposting, where copies of the message cannot be related." -- "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 phaselocker at gmail.com Mon Feb 3 15:17:31 2014 From: phaselocker at gmail.com (Lewis Pike) Date: Mon, 3 Feb 2014 10:17:31 -0500 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <20140129140546.GA4085@ordinator> References: <20140129024408.GA3961@ordinator> <52E8F06F.4080503@parisson.com> <20140129140546.GA4085@ordinator> Message-ID: <20140203151731.GA11566@ordinator> On Wed, Jan 29, 2014 at 09:05:46AM -0500, Lewis Pike wrote: > On Wed, Jan 29, 2014 at 01:13:35PM +0100, Guillaume Pellerin wrote: > > Hi Lewis, > > > > I'm a contributor of the driver for this card in the Linux kernel > > and it works very well for me. You should maybe tune your ALSA > > setup for this. > > > > First, what is your kernel version? > > > > Guillaume > > Guillaume! > > Your help with this is hugely appreciated! The fact that you're not > seeing the same issue gives me hope that a solution exists. > > My current kernel version is 3.12.7. I first attempted to use the > Fast Track Pro for audio capture three years ago, around version > 2.6.37 but I ran into the same problem. > > .lewis Hi Guillaume, I've been continuing to test the Fast Track Pro, trying to identify a solution to my problem, but unfortunately I haven't had any success thus far. I made a post on the alsa-user mailing list [1] but it hasn't generated any interest. As a contributor to the driver of for this device, can you recommend another forum where I might be able to find some help? Much obliged! .lewis [1] http://permalink.gmane.org/gmane.linux.alsa.user/38146 From federicogalland at gmail.com Mon Feb 3 20:38:24 2014 From: federicogalland at gmail.com (F Tux) Date: Mon, 3 Feb 2014 18:38:24 -0200 Subject: [LAU] rt-patch vs cgroups approach Message-ID: Hi all! In the past few months I've been fiddling with the kernel config to get the best latency out of my laptop (core2 t6400, 2GHz 1MB L2, 4GB of RAM). The best settings I've got so far are 128*3 with a sample rate of 48000Hz. All of this with the rt-patch and following the guide in the linux audio wiki. My question is regarding the method described here http://proaudio.tuxfamily.org/wiki/index.php?title=DAW_Digital_Audio_Workstation#Instructions_for_3.x_Kernels I've seen this described in many a forum, and people say they got better results with it than with the rt-patch. Do you have any experiences with the cgroups solution? Is it stable enough for live performance? I will be testing it some time in the future, but I thought it would be a good idea to fire up the discussion here and make it for a common profit. Thanks a lot to all of you. It's great to know there's a place for the really independent musician to learn and share for free with his fellow hackers around the globe. Bye From jeremy at autostatic.com Mon Feb 3 21:12:53 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 03 Feb 2014 22:12:53 +0100 Subject: [LAU] rt-patch vs cgroups approach In-Reply-To: References: Message-ID: <52F00655.3060808@autostatic.com> On 02/03/2014 09:38 PM, F Tux wrote: > Hi all! > > In the past few months I've been fiddling with the kernel config to > get the best latency out of my laptop (core2 t6400, 2GHz 1MB L2, 4GB > of RAM). > The best settings I've got so far are 128*3 with a sample rate of > 48000Hz. All of this with the rt-patch and following the guide in the > linux audio wiki. > > My question is regarding the method described here > http://proaudio.tuxfamily.org/wiki/index.php?title=DAW_Digital_Audio_Workstation#Instructions_for_3.x_Kernels > > I've seen this described in many a forum, and people say they got > better results with it than with the rt-patch. > > Do you have any experiences with the cgroups solution? Is it stable > enough for live performance? > > I will be testing it some time in the future, but I thought it would > be a good idea to fire up the discussion here and make it for a common > profit. > > Thanks a lot to all of you. It's great to know there's a place for the > really independent musician to learn and share for free with his > fellow hackers around the globe. > > Bye Hello, The information in the proaudio wiki seems wrong to me. cgroups do not yield the same result as using the RT patchset as the RT patchset targets a completely different goal. Also cgroups add an unnecessary layer of complexity. I've never had to use them (from what I've understood cgroups are disabled by default on Ubuntu and maybe also Debian) and that's also one of the reasons there's nothing about them in the linuxaudio.org wiki (of which I'm the main author). What I think that still works best is to use a RT kernel or low-latency (PREEMPT__LL) kernel, use rtirq and raise JACK's priority to a reasonable number. And not the default prio of 10 that figures in the proaudio wiki, another reason that makes me doubt if the author understands how setting rtprio works. But I'm open to a discussion about this as I don't have any hands-on experience with cgroups so I could be completely mistaken :) Best, 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 csanchezgs at gmail.com Mon Feb 3 21:34:47 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 3 Feb 2014 22:34:47 +0100 Subject: [LAU] new song with ardour and freesound.org In-Reply-To: <52EA11A7.5090206@thorstenwilms.com> References: <20140129210538.015aa290@debian> <52EA11A7.5090206@thorstenwilms.com> Message-ID: El 30/01/2014 09:47, "Thorsten Wilms" escribi?: > > On 01/29/2014 11:04 PM, Chris Bungue wrote: >> >> I have update the credits. Thank you very much >> for the hint. I hope everything is ok now. > > > NP. Good to see you took care of it quickly. Seems fine now. > > >> Usually I don't work with samples/loops and it was not ease to mix it. > > > Neither shows ;) > > > > -- > Thorsten Wilms > > thorwil's design for free software: > http://thorwil.wordpress.com/ > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Nice track, Chris. It gives you a nice move and there are some motif/ hook changes before it gets two much "loopy-repetitive". Thanks for sharing. -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Mon Feb 3 22:16:49 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Mon, 3 Feb 2014 17:16:49 -0500 Subject: [LAU] rt-patch vs cgroups approach In-Reply-To: References: Message-ID: On Mon, Feb 3, 2014 at 3:38 PM, F Tux wrote: > Hi all! > > In the past few months I've been fiddling with the kernel config to > get the best latency out of my laptop (core2 t6400, 2GHz 1MB L2, 4GB > of RAM). > The best settings I've got so far are 128*3 with a sample rate of > 48000Hz. All of this with the rt-patch and following the guide in the > linux audio wiki. > > My question is regarding the method described here > > http://proaudio.tuxfamily.org/wiki/index.php?title=DAW_Digital_Audio_Workstation#Instructions_for_3.x_Kernels > > I've seen this described in many a forum, and people say they got > better results with it than with the rt-patch. > > Do you have any experiences with the cgroups solution? Is it stable > enough for live performance? > > I will be testing it some time in the future, but I thought it would > be a good idea to fire up the discussion here and make it for a common > profit. > > Thanks a lot to all of you. It's great to know there's a place for the > really independent musician to learn and share for free with his > fellow hackers around the globe. > as jeremy hinted, cgroups and the RT patch are 100% orthogonal. The RT patch is way to improve scheduling latencies. The cgroup mechanism is a way to control access to realtime scheduling. -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Tue Feb 4 08:16:39 2014 From: atte at youmail.dk (Atte) Date: Tue, 04 Feb 2014 09:16:39 +0100 Subject: [LAU] New album out: Modlys/2013 - workflow In-Reply-To: <52C2DBE1.1030107@woh.rr.com> References: <52C2CF6F.6060105@youmail.dk> <52C2D5D5.9020907@gmail.com> <52C2D9F4.3070207@youmail.dk> <52C2DBE1.1030107@woh.rr.com> Message-ID: <52F0A1E7.4000507@youmail.dk> On 12/31/2013 03:59 PM, Dave Phillips wrote: > Even though I'm not a Renoise user - not even a heretical one - I would > love to read how you create your music with the software. Please > consider posting a description here or elsewhere, the more detailed the > better. Ok, as promised a few (?) words about "how I use renoise" and "my work flow". First a little bit about how the tracks started: After playing with paulstretch I was struck by the beauty of the sounds produced by it. Actually it's kind of surprising how just about everything sounds beautiful at 1/10 or 1/20 speed. So I started working on an album (entitled b?ndsl?jfe (tape loop)) where the textures would be comprised of stretched material, and with faster stuff like glitches, percussion and drums on top. I stretched all kinds of audio material, chopped it up (using the very handy "slice" feature in renoise) to have smaller building blocks (typically of one or two bars length), and arrange them into a different structure. Most of the time material from different stretched audio were layered to allow me to control the texture individually. Sometimes they were filtered, sometimes with a lfo controlled filter, sometimes ran through a mda vocoder/talkbox as carrier and some rhythmic thing like a percussion loop as modulator. I've been working on/off on this album for about two years, but never finished all the tracks (there were 19). On "2013" the following 5 tracks comes from that concept album: "engang", "maria bebudelse", "forfra", "boston undervej" and "13" (the later contains no stretched material, but was meant as a contrast from the rest of the songs). A few years ago I was on vacation on Gran Canaria. While on the plane, hanging around the pool and in general relaxing, the other members of the family were reading books or solving cross word puzzles, but I started 19 small sketches, mostly one or two bar loops. The idea was not to finish anything, just play around, some took only 15 minutes or less. I had plans for later finishing the best sketches for an album entitled "Las Palmas" (named after the airport on Gran Canaria). In general I went for instrumental tracks with a dance floor feel without the actual beat. Each sketch were named after something the happened just before or after the sketch was made. "de venter", "du kender titanium", "20", igen igen", "p? vej hjem" og "meyer" were selection from that concept album. The three meditations were done on commission as meditation tracks. About the workflow: I almost always start out by going for a mood that I like. Every sound contributes to the mood, but to me the bass drum is especially important. So I spend some time finding a bass drum that fits with the mood. I also like to truncate sounds like hihats or claps to make them more percussive. In general I think every sound or part of a song should be strong, so instead of just throwing random stuff in, I try to find thing that are not "just a sound", something with character. Sometimes (heavy) processing can make a dull sound interesting, but sometimes I just throw it away again if it's too blah. When I have a loop that I like with basis drums and some harmony I often start to think about the structure of the whole track, since I found it very difficult to break out of a loop later, if I work too much on it. In renoise this means that I would copy my pattern so the song takes about 4 minutes, and then start working on an alternate, contrasting part. I also found it important always to remember to leave space when adding parts. It's so easy to overdo each part, which leaves no room for later parts, so I try to imagine how a certain sound would fit with both what is there already and things that could be added later, sometimes I have a pretty clear idea of what those later things could be sometimes I don't. It's also nice to have things, be it sounds, effects, musical ideas that only show up once or maybe twice in the track, this is easier to work with when the structure is more or less in place. In renoise you can really "program" the music, by typing in the music on the computer keyboard. This is usually the fastest and most precise way to work, but the music tends to sound very sterile (to my ears) if done this way. I have a small LPK25 that's always hooked up and sits on my desktop, and I try to use it as much as possible (I also have a 88keys weighted yamaha that I hook up if I have the time). I mostly quantize things to have a tight groove, but some parts are non-quantized to add some organic sound the the music. Regarding the production: I mix things as I go, since to me that's part of the compositional process. Some things needs effects to work in the mood or in the sound. I rarely use reverb, mostly because the tend to give a washed out and often cheap sound to the track, but admittedly because I don't have that many great sounding reverbs. The ones I use the most are the TAL reverbs, I think their tails sounds the least metallic. Instead I use a lot of delay (especially renoise buildin Multitap Delay with filter on the delays) that in renoise are easy to sync to the tempo. I also use alot of lopass and hipass filters to shape the sound of things. For instance I like to do extreme hipass on hihats of things that should work like hihats, to really lift them above the rest of the track, leaving room in the midrange and lo hiend for other sounds. Prominent melodic sounds (electric pianos, piano, synth sounds, pads, lead sounds) often needs alot of EQ to bring out the body of the sound and make them blend with the mood. I like to spread out the work of a track over a long time. Work a little on it, then leave it alone for a while and come back to it. For this purpose I made a renoise hackish tool that renders a track to wav from the commandline. I then have a script "generate_mp3" that renders modified (make style) tracks to an mp3 folder directly on my server, this folder is synced to my android phone with the app "folder sync" so that I always have the latest versions of the tracks with me. So I spend alot of time listening to things away from the computer, for instance while running of driving the car. That works really well for me, if I listen to a track in renoise I tend to jump right in and change something I don't like, but just listening without the ability to change anything gives a much better overview, especially when working on a collection of tracks at the same time. Some final words regarding renoise: Brendan Jones commented "you made this with a tracker?"... Well to me renoise is just a tool. I worked with a lot of different programs, my first album was made with floss tools (ardour, muse, specemin, zynaddsubfx, phasex, ams, pd and more). I really believe that for the most part you could make your music in any tool. So why renoise? The most important thing for me is it's extreme stability, it just never crashes. There's nothing that sucks the motivation more out of me than being in flow with a song and having the program crash on me. I might loose work, but even if I do it really breaks the flow, and I really, really hate that. It's also really nice to just load a project and everything is exactly where you left it. I tried the session managers, but I never got into it. Also when using alot of tools, a bug in one program might break the whole chain. So you have to upgrade that piece of software, and now your project might not load. And renoise is very "rounded" compared to the floss tools IMHO. Everything is really well thought of and well tested. So many small things that mean working is simply faster, consistent keyboard shortcuts, fast switching of views, infinite undo on everything including fader moves, support for mouse scroll almost everywhere. And it doesn't hurt that renoise is really light on the CPU. There are things that are more challenging in a tracker, mostly working over the seemingly hard boundaries of patterns and accepting the messy look of realtime midi recordings in the pattern editor. I hope that gives an idea about the workflow. Although that was a lot of words, please feel free to ask if there's something specific you'd like to know about :-) -- Atte http://atte.dk http://modlys.dk From willgodfrey at musically.me.uk Tue Feb 4 09:24:40 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 4 Feb 2014 09:24:40 +0000 Subject: [LAU] New album out: Modlys/2013 - workflow In-Reply-To: <52F0A1E7.4000507@youmail.dk> References: <52C2CF6F.6060105@youmail.dk> <52C2D5D5.9020907@gmail.com> <52C2D9F4.3070207@youmail.dk> <52C2DBE1.1030107@woh.rr.com> <52F0A1E7.4000507@youmail.dk> Message-ID: <20140204092440.2de6cf6e@debian> On Tue, 04 Feb 2014 09:16:39 +0100 Atte wrote: > On 12/31/2013 03:59 PM, Dave Phillips wrote: > > > Even though I'm not a Renoise user - not even a heretical one - I would > > love to read how you create your music with the software. Please > > consider posting a description here or elsewhere, the more detailed the > > better. > > Ok, as promised a few (?) words about "how I use renoise" and "my work > flow". > > First a little bit about how the tracks started: > After playing with paulstretch I was struck by the beauty of the sounds > produced by it. Actually it's kind of surprising how just about > everything sounds beautiful at 1/10 or 1/20 speed. So I started working > on an album (entitled b?ndsl?jfe (tape loop)) where the textures would > be comprised of stretched material, and with faster stuff like glitches, > percussion and drums on top. I stretched all kinds of audio material, > chopped it up (using the very handy "slice" feature in renoise) to have > smaller building blocks (typically of one or two bars length), and > arrange them into a different structure. Most of the time material from > different stretched audio were layered to allow me to control the > texture individually. Sometimes they were filtered, sometimes with a lfo > controlled filter, sometimes ran through a mda vocoder/talkbox as > carrier and some rhythmic thing like a percussion loop as modulator. > I've been working on/off on this album for about two years, but never > finished all the tracks (there were 19). On "2013" the following 5 > tracks comes from that concept album: "engang", "maria bebudelse", > "forfra", "boston undervej" and "13" (the later contains no stretched > material, but was meant as a contrast from the rest of the songs). > > A few years ago I was on vacation on Gran Canaria. While on the plane, > hanging around the pool and in general relaxing, the other members of > the family were reading books or solving cross word puzzles, but I > started 19 small sketches, mostly one or two bar loops. The idea was not > to finish anything, just play around, some took only 15 minutes or less. > I had plans for later finishing the best sketches for an album entitled > "Las Palmas" (named after the airport on Gran Canaria). In general I > went for instrumental tracks with a dance floor feel without the actual > beat. Each sketch were named after something the happened just before or > after the sketch was made. "de venter", "du kender titanium", "20", igen > igen", "p? vej hjem" og "meyer" were selection from that concept album. > > The three meditations were done on commission as meditation tracks. > > About the workflow: > I almost always start out by going for a mood that I like. Every sound > contributes to the mood, but to me the bass drum is especially > important. So I spend some time finding a bass drum that fits with the > mood. I also like to truncate sounds like hihats or claps to make them > more percussive. In general I think every sound or part of a song should > be strong, so instead of just throwing random stuff in, I try to find > thing that are not "just a sound", something with character. Sometimes > (heavy) processing can make a dull sound interesting, but sometimes I > just throw it away again if it's too blah. When I have a loop that I > like with basis drums and some harmony I often start to think about the > structure of the whole track, since I found it very difficult to break > out of a loop later, if I work too much on it. In renoise this means > that I would copy my pattern so the song takes about 4 minutes, and then > start working on an alternate, contrasting part. I also found it > important always to remember to leave space when adding parts. It's so > easy to overdo each part, which leaves no room for later parts, so I try > to imagine how a certain sound would fit with both what is there already > and things that could be added later, sometimes I have a pretty clear > idea of what those later things could be sometimes I don't. It's also > nice to have things, be it sounds, effects, musical ideas that only show > up once or maybe twice in the track, this is easier to work with when > the structure is more or less in place. > > In renoise you can really "program" the music, by typing in the music on > the computer keyboard. This is usually the fastest and most precise way > to work, but the music tends to sound very sterile (to my ears) if done > this way. I have a small LPK25 that's always hooked up and sits on my > desktop, and I try to use it as much as possible (I also have a 88keys > weighted yamaha that I hook up if I have the time). I mostly quantize > things to have a tight groove, but some parts are non-quantized to add > some organic sound the the music. > > Regarding the production: I mix things as I go, since to me that's part > of the compositional process. Some things needs effects to work in the > mood or in the sound. I rarely use reverb, mostly because the tend to > give a washed out and often cheap sound to the track, but admittedly > because I don't have that many great sounding reverbs. The ones I use > the most are the TAL reverbs, I think their tails sounds the least > metallic. Instead I use a lot of delay (especially renoise buildin > Multitap Delay with filter on the delays) that in renoise are easy to > sync to the tempo. I also use alot of lopass and hipass filters to shape > the sound of things. For instance I like to do extreme hipass on hihats > of things that should work like hihats, to really lift them above the > rest of the track, leaving room in the midrange and lo hiend for other > sounds. Prominent melodic sounds (electric pianos, piano, synth sounds, > pads, lead sounds) often needs alot of EQ to bring out the body of the > sound and make them blend with the mood. > > I like to spread out the work of a track over a long time. Work a little > on it, then leave it alone for a while and come back to it. For this > purpose I made a renoise hackish tool that renders a track to wav from > the commandline. I then have a script "generate_mp3" that renders > modified (make style) tracks to an mp3 folder directly on my server, > this folder is synced to my android phone with the app "folder sync" so > that I always have the latest versions of the tracks with me. So I spend > alot of time listening to things away from the computer, for instance > while running of driving the car. That works really well for me, if I > listen to a track in renoise I tend to jump right in and change > something I don't like, but just listening without the ability to change > anything gives a much better overview, especially when working on a > collection of tracks at the same time. > > Some final words regarding renoise: Brendan Jones commented "you made > this with a tracker?"... Well to me renoise is just a tool. I worked > with a lot of different programs, my first album was made with floss > tools (ardour, muse, specemin, zynaddsubfx, phasex, ams, pd and more). I > really believe that for the most part you could make your music in any > tool. So why renoise? The most important thing for me is it's extreme > stability, it just never crashes. There's nothing that sucks the > motivation more out of me than being in flow with a song and having the > program crash on me. I might loose work, but even if I do it really > breaks the flow, and I really, really hate that. It's also really nice > to just load a project and everything is exactly where you left it. I > tried the session managers, but I never got into it. Also when using > alot of tools, a bug in one program might break the whole chain. So you > have to upgrade that piece of software, and now your project might not > load. And renoise is very "rounded" compared to the floss tools IMHO. > Everything is really well thought of and well tested. So many small > things that mean working is simply faster, consistent keyboard > shortcuts, fast switching of views, infinite undo on everything > including fader moves, support for mouse scroll almost everywhere. And > it doesn't hurt that renoise is really light on the CPU. There are > things that are more challenging in a tracker, mostly working over the > seemingly hard boundaries of patterns and accepting the messy look of > realtime midi recordings in the pattern editor. > > I hope that gives an idea about the workflow. Although that was a lot of > words, please feel free to ask if there's something specific you'd like > to know about :-) Wow! Thanks for that. It's fascinating to read how other musicians develop their ideas. -- 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 atte at youmail.dk Tue Feb 4 10:01:01 2014 From: atte at youmail.dk (Atte) Date: Tue, 04 Feb 2014 11:01:01 +0100 Subject: [LAU] New album out: Modlys/2013 - workflow In-Reply-To: <20140204092440.2de6cf6e@debian> References: <52C2CF6F.6060105@youmail.dk> <52C2D5D5.9020907@gmail.com> <52C2D9F4.3070207@youmail.dk> <52C2DBE1.1030107@woh.rr.com> <52F0A1E7.4000507@youmail.dk> <20140204092440.2de6cf6e@debian> Message-ID: <52F0BA5D.2090807@youmail.dk> On 02/04/2014 10:24 AM, Will Godfrey wrote: > Wow! > Thanks for that. It's fascinating to read how other musicians develop their > ideas. And it's nice to talk about other things than .jackrc, Makefiles and patching kernels sometimes :-) -- Atte http://atte.dk http://modlys.dk From mlang at delysid.org Tue Feb 4 13:31:22 2014 From: mlang at delysid.org (Mario Lang) Date: Tue, 04 Feb 2014 14:31:22 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> (raf's message of "Sat, 25 Jan 2014 15:31:52 +0100") References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> Message-ID: <87vbwvatsl.fsf@fx.delysid.org> raf writes: > Hello, > > you'l probably be happy to know the existence of three great tools : midish, linuxsampler and Nama. > 1) midish is a command line midi sequencer with a lot of great features > http://www.midish.org/ midish looks rather interesting. However, the manual.html basically just explains how to record data from an input device. Does latest midish support creating MIDI data from scratch, and if so, is there perhaps some examples on how to do that? -- CYa, ????? | Debian Developer .''`. | Get my public key via finger mlang/key at db.debian.org : :' : | 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44 `. `' `- From pshirkey at boosthardware.com Tue Feb 4 13:58:19 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Wed, 5 Feb 2014 00:58:19 +1100 (EST) Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <87vbwvatsl.fsf@fx.delysid.org> References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> Message-ID: <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> On Wed, February 5, 2014 12:31 am, Mario Lang wrote: > raf writes: > >> Hello, >> >> you'l probably be happy to know the existence of three great tools : >> midish, linuxsampler and Nama. >> 1) midish is a command line midi sequencer with a lot of great features >> http://www.midish.org/ > > midish looks rather interesting. However, the manual.html basically > just explains how to record data from an input device. Does latest > midish support creating MIDI data from scratch, and if so, is there > perhaps some examples on how to do that? > Check this section : http://www.midish.org/manual.html#ev You can compose note on/off events and save the sequence as a song or export the song to .mid There are also these two other options: alsaseq: http://pp.com.mx/python/alsaseq/project.html mididings: http://das.nasophon.de/mididings/ Teqqer also looks very promising. -- Patrick Shirkey Boost Hardware Ltd From mista.tapas at gmx.net Tue Feb 4 14:19:45 2014 From: mista.tapas at gmx.net (Florian Paul Schmidt) Date: Tue, 04 Feb 2014 15:19:45 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> Message-ID: <52F0F701.1000002@gmx.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 04.02.2014 14:58, Patrick Shirkey wrote: > Teqqer also looks very promising. Mario even commited some bugfixes/improvements to the teq library. But then seemingly lost interest ;D I'm still working on it. Expect a beta in about a month or so.. Flo -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.14 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iQEcBAEBAgAGBQJS8Pb+AAoJEA5f4Coltk8Zr8gH/3ovqZwjbrojMSvTNbes6vLp SsykYAOrQ01+iuiXufaL8jzXsmJHekGm8QKdr+4INKqY9c7DtpyYF4T28iIDkZf0 9Ev9TOaC5jiiQFoJ2jlZRN3Ic+KvifxpWTyMJSJg5n0xCeQt1eyzJQ5kBwCFhpmS 2ITpAPvsBOu2s6X99CQaCshyu6a7+R9k1ZUprCFGDrmjwP6292uBVb1RT+NrWYpc XvH1Wcm0Iks2zBxg+wUt2mjb/LzzFRT+SyVszbz93WBBgKc2d0Ymj4XIizrCMAab MCo2xAwdO7i8/2Gnf/kE71yxRqTU0Sdx3OCyJuF+gKVmahVxiiI+VtjJRJ8xwL0= =NjOn -----END PGP SIGNATURE----- From agus3985 at gmail.com Tue Feb 4 16:02:11 2014 From: agus3985 at gmail.com (=?ISO-8859-1?Q?Jos=E9_Agust=EDn_Terol_Sanchis?=) Date: Tue, 4 Feb 2014 17:02:11 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz Message-ID: Hello, I am thinking in set up a home studio for recording and mixing based on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but first, I would like to know the experience of someone who has already used it. I've taken a look at http://www.ffado.org/?q=devicesupport/list but I don't have clear if this device can be used at 96kHz, because currently it's marked as "experimental". I know other Saffire have "full support", but nowadays are discontinued. Does anybody used it for hours (and days) at this frequency? Also, I would be interested in experiences with similar interfaces that also work fine in linux. Thank you, Agus Terol -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Tue Feb 4 16:11:05 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 4 Feb 2014 11:11:05 -0500 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: Message-ID: On Tue, Feb 4, 2014 at 11:02 AM, Jos? Agust?n Terol Sanchis < agus3985 at gmail.com> wrote: > Hello, I am thinking in set up a home studio for recording and mixing > based on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but > first, I would like to know the experience of someone who has already used > it. > > I've taken a look at http://www.ffado.org/?q=devicesupport/list but I > don't have clear if this device can be used at 96kHz, because currently > it's marked as "experimental". I know other Saffire have "full support", > but nowadays are discontinued. Does anybody used it for hours (and days) at > this frequency? > > Also, I would be interested in experiences with similar interfaces that > also work fine in linux. > If you're building a *home studio*, is there some reason NOT to use a PCI-based card, rather than something that has to connect via an external port (firewire, USB, etc) ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From agus3985 at gmail.com Tue Feb 4 16:29:34 2014 From: agus3985 at gmail.com (=?ISO-8859-1?Q?Jos=E9_Agust=EDn_Terol_Sanchis?=) Date: Tue, 4 Feb 2014 17:29:34 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: Message-ID: Hi Paul, thanks for answering. Yes, indeed I have a reason to not use a PCI-based card. My idea is to set up the interface in a portable 19'' rack, in order to connect it to a laptop, and then make some recordings in the practice (out of the home studio). Regards, Agus Terol 2014-02-04 Paul Davis : > > > > On Tue, Feb 4, 2014 at 11:02 AM, Jos? Agust?n Terol Sanchis < > agus3985 at gmail.com> wrote: > >> Hello, I am thinking in set up a home studio for recording and mixing >> based on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but >> first, I would like to know the experience of someone who has already used >> it. >> >> I've taken a look at http://www.ffado.org/?q=devicesupport/list but I >> don't have clear if this device can be used at 96kHz, because currently >> it's marked as "experimental". I know other Saffire have "full support", >> but nowadays are discontinued. Does anybody used it for hours (and days) at >> this frequency? >> >> Also, I would be interested in experiences with similar interfaces that >> also work fine in linux. >> > > If you're building a *home studio*, is there some reason NOT to use a > PCI-based card, rather than something that has to connect via an external > port (firewire, USB, etc) ? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at parisson.com Tue Feb 4 17:29:28 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Tue, 04 Feb 2014 18:29:28 +0100 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <20140203151731.GA11566@ordinator> References: <20140129024408.GA3961@ordinator> <52E8F06F.4080503@parisson.com> <20140129140546.GA4085@ordinator> <20140203151731.GA11566@ordinator> Message-ID: <52F12378.7070700@parisson.com> Hi! On 03/02/2014 16:17, Lewis Pike wrote: > > Hi Guillaume, > > I've been continuing to test the Fast Track Pro, trying to identify a > solution to my problem, but unfortunately I haven't had any success > thus far. I made a post on the alsa-user mailing list [1] but it > hasn't generated any interest. As a contributor to the driver of for > this device, can you recommend another forum where I might be able to > find some help? Much obliged! > Sorry Lewis, I was AFK these days. Back now :) Can you try to add this line to /etc/modprobe.d/alsa-base.conf options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x03 enable=1 nrpacks=1 and then: $ sudo modprobe -r snd-usb-audio $ sudo modprobe snd-usb-audio un/replug the card and make your tests again. I must admit I've almost always use the FTP with JACK so I'm not completly aware of the behavior of the device without it. If this has no (good) effect, it could be something between you USB system and the kernel. I've discover recently that crackles could raise up on other cards and systems because of a dynamic USB ID allocation of the kernel. The pb disappeared when the option is switched off. So, let's continue if the first ALSA tweak doesn't work.. Regards, Guillaume From jeremy at autostatic.com Tue Feb 4 20:08:01 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 04 Feb 2014 21:08:01 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: Message-ID: <52F148A1.5000006@autostatic.com> On 02/04/2014 05:02 PM, Jos? Agust?n Terol Sanchis wrote: > Hello, I am thinking in set up a home studio for recording and mixing based > on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but first, I > would like to know the experience of someone who has already used it. > > I've taken a look at http://www.ffado.org/?q=devicesupport/list but I don't > have clear if this device can be used at 96kHz, because currently it's > marked as "experimental". I know other Saffire have "full support", but > nowadays are discontinued. Does anybody used it for hours (and days) at > this frequency? > > Also, I would be interested in experiences with similar interfaces that > also work fine in linux. > > Thank you, > Agus Terol Hello Agus, I can confirm the Saffire Pro 40 works at 96Khz. I barely use this sample rate though so I can't tell if it will run for hours or even days. I don't see any reason why it shouldn't. I can use my Saffire Pro 40 for hours at 48Khz without any hitch. I've occasionally forgot to switch the set-up off and the next day everything was still running fine. Best, 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 atte at youmail.dk Tue Feb 4 21:16:28 2014 From: atte at youmail.dk (Atte) Date: Tue, 04 Feb 2014 22:16:28 +0100 Subject: [LAU] New album out: Modlys/2013 - workflow In-Reply-To: References: <52C2CF6F.6060105@youmail.dk> <52C2D5D5.9020907@gmail.com> <52C2D9F4.3070207@youmail.dk> <52C2DBE1.1030107@woh.rr.com> <52F0A1E7.4000507@youmail.dk> Message-ID: <52F158AC.5020803@youmail.dk> On 02/04/2014 02:39 PM, James Mckernon wrote: > On Tue, Feb 4, 2014 at 8:16 AM, Atte wrote: > You use the word 'find'; do you have a large library of drum samples > you use for this? Or do you synthesize any of your own drum sounds > from scratch? And a similar question for the other instruments in your > songs - do you use softsynths, sample-based instruments, or a > combination? I have a quite large sample (flac) collection :-) atte at skagen:~/music/samples$ du -h | tail -n1 39G . Some are ancient and sampled on my roland s50 back in the day directly from vinyl that I personally hunted down in second hand record stores, most are collected from all over the internet. Others (mostly melodic non-drum instruments) are samples from the synths I have or have owned (wavestation, jv80, xp80, micron, m50) or generated with ams, zynaddsubfx or even the nativ synth in energyXT. The only synth I really use is Loomer Aspect, although I don't use it that much. The main reason why I prefer samples over synths are the "build-in" complexity with samples. To me music mostly build with synthesizers tends to lack depth. Don't get me wrong, I like Jarre and Kitaro, but I'm going for a more organic sound, my electronic heros are Squarepusher and chr15 + various underground stuff I stumble over from time to time. I really like to load a sample that might suggest a chord that I don't even know how is tuned and start messing with it by ear. I love to be surprised and get pushed to harmonies I wouldn't think of myself. >> I like to spread out the work of a track over a long time. Work a little on >> it, then leave it alone for a while and come back to it. For this purpose I >> made a renoise hackish tool that renders a track to wav from the >> commandline. > > No need to go into too much detail, but as a Renoise user, I'm curious > - roughly how did you achieve this? Something using Renoise's internal > Lua scripting? Yes, it's done in lua. Some things are hardcoded, it works in companion with a bash script. If this doesn't scare you. I can send you the stuff off-list... > I like the idea of listening to and evaluating things in another context. It's sooo important, at least for me. -- Atte http://atte.dk http://modlys.dk From atte at youmail.dk Tue Feb 4 21:23:15 2014 From: atte at youmail.dk (Atte) Date: Tue, 04 Feb 2014 22:23:15 +0100 Subject: [LAU] New album out: Modlys/2013 - workflow In-Reply-To: <52F0EE44.6010405@woh.rr.com> References: <52C2CF6F.6060105@youmail.dk> <52C2D5D5.9020907@gmail.com> <52C2D9F4.3070207@youmail.dk> <52C2DBE1.1030107@woh.rr.com> <52F0A1E7.4000507@youmail.dk> <52F0EE44.6010405@woh.rr.com> Message-ID: <52F15A43.6030301@youmail.dk> On 02/04/2014 02:42 PM, Dave Phillips wrote: > Very cool, thank you for this detailed explication. :) Some of your > methods are rather different from what I expected, you're definitely in > control of your tools, and you have fine musical sense. I'm not sure what you expected (but would like to hear it)... > I > use RubberBand for this sort of processing, though I have experimented > with paulstretch. I tested various stretching engines, and paulstretch just sounds so beautiful. Of course it's a little picky/tricky to work with, which is why I made a python wrapper script that makes it read/write all kinds of formats and makes it more pleasant to work with :-) > Best regards, and as always, thanks for the beautiful music, You're welcome, glad someone enjoys it! -- Atte http://atte.dk http://modlys.dk From poeticintensity at gmail.com Tue Feb 4 21:31:58 2014 From: poeticintensity at gmail.com (Jason Jones) Date: Tue, 4 Feb 2014 14:31:58 -0700 Subject: [LAU] Album produced in Linux Message-ID: Fusion between gospel and punk, this whole album was produced in Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) plugins. http://tinyboats.bandcamp.com/ Any thoughts or comments? --Jason www.artcitysound.com (Linux-based studio) -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonetsu at teksavvy.com Wed Feb 5 00:37:18 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Tue, 4 Feb 2014 19:37:18 -0500 Subject: [LAU] Album produced in Linux In-Reply-To: References: Message-ID: <20140204193718.395b253b@mistral> On Tue, 4 Feb 2014 14:31:58 -0700, Jason Jones wrote : > Fusion between gospel and punk, this whole album was produced in > Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) > plugins. > > http://tinyboats.bandcamp.com/ > > Any thoughts or comments? The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. Can play youtube all right. All enabled in No Script for this bandcamp page. Maybe I don;t understand the concept of bandcamp and the play button, even though the mouse cursor changes to a pointing hand when hovering above it, actually only works when someone has paid. I don't know. So, no musical comment ! From harryhaaren at gmail.com Wed Feb 5 00:44:13 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Wed, 5 Feb 2014 00:44:13 +0000 Subject: [LAU] Album produced in Linux In-Reply-To: <20140204193718.395b253b@mistral> References: <20140204193718.395b253b@mistral> Message-ID: On Wed, Feb 5, 2014 at 12:37 AM, jonetsu at teksavvy.com wrote > > The play button doe snto work. Works perfectly here: there must be something up with your flash setup. > So, no musical comment ! > I've only listend on little laptop speakers, but I'm liking Dead Planet Society, the cymbals are very bright and sparkely. Perhaps its over-emphasised by the horrid little speakers though! I'll listen to it on proper speakers, and report back then. Sounds good so far! Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonetsu at teksavvy.com Wed Feb 5 01:38:25 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Tue, 4 Feb 2014 20:38:25 -0500 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> Message-ID: <20140204203825.1dae1fbb@mistral> On Wed, 5 Feb 2014 00:44:13 +0000, Harry van Haaren wrote : > On Wed, Feb 5, 2014 at 12:37 AM, jonetsu at teksavvy.com > wrote > > > > The play button doe snto work. > > Works perfectly here: there must be something up with your flash > setup. Never did any setup. Youtube works nicely. Hence I think it's the bandcamp website that may be asking for 'something else'. Even with enabled cookies for this domain, it won't work. It's playing right now something from Soundcloud.com - works fine. I guess that if I put music online I'll be using Soundcloud as it seems most easiest to use, technically speaking :) From lsd at wootangent.net Wed Feb 5 06:23:29 2014 From: lsd at wootangent.net (Leigh Dyer) Date: Wed, 05 Feb 2014 17:23:29 +1100 Subject: [LAU] Album produced in Linux In-Reply-To: <20140204193718.395b253b@mistral> References: <20140204193718.395b253b@mistral> Message-ID: <52F1D8E1.9050500@wootangent.net> On 5/02/2014 11:37 am, jonetsu at teksavvy.com wrote: > On Tue, 4 Feb 2014 14:31:58 -0700, > Jason Jones wrote : > >> Fusion between gospel and punk, this whole album was produced in >> Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) >> plugins. >> >> http://tinyboats.bandcamp.com/ >> >> Any thoughts or comments? > > The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. Can > play youtube all right. All enabled in No Script for this bandcamp > page. For Firefox on Linux, you'll need to have actual Adobe Flash installed -- YMMV if you have one of the open-source flash plugins installed. If you have Flash installed, and you're not blocking any scripts, then everything should work. Otherwise, you can use a browser that supports HTML5 MP3 playback; right now on Linux, that means Chrome, or Chromium with "extra" ffmpeg codecs installed (on Ubuntu etc. that's the chromium-codecs-ffmpeg-extra package). Firefox has started to support MP3 on Windows and OS X using codecs supplied by the OS, and it looks like the Linux version may do this soon, too, using GStreamer, but it's not quite there yet. (Full disclosure -- I work for Bandcamp, though I'm replying from my personal address, since that's what I'm subscribed to the list with). Thanks Leigh From shanipribadi at gmx.net Wed Feb 5 06:46:10 2014 From: shanipribadi at gmx.net (Shani Hadiyanto Pribadi) Date: Wed, 5 Feb 2014 13:46:10 +0700 Subject: [LAU] Album produced in Linux In-Reply-To: <20140204203825.1dae1fbb@mistral> References: <20140204193718.395b253b@mistral> <20140204203825.1dae1fbb@mistral> Message-ID: On Wed, Feb 5, 2014 at 8:38 AM, jonetsu at teksavvy.com wrote: > Never did any setup. Youtube works nicely. Hence I think it's the > bandcamp website that may be asking for 'something else'. Even with > enabled cookies for this domain, it won't work. > > It's playing right now something from Soundcloud.com - works fine. > > I guess that if I put music online I'll be using Soundcloud as it seems > most easiest to use, technically speaking :) I'm using firefox without flash and bandcamp works for me as long as I allow bandcamp.com and bcbits.com (I think that's their cdn) in noscript. Firefox outputs to alsa loopback, zalsa_in connects to zita-mu1, zita-mu1 (with HP switch on) connects to hw out. The way I see it soundcloud and bandcamp serves different purposes. I feel that bandcamp is more suited for album releases, while soundcloud is for the occasional track releases. But there's no reason not to use both, some people might prefer bandcamp's personalized and cleaner page, some people might want to use soundcloud social features, like comments and followers. On Tue, 4 Feb 2014 14:31:58 -0700, Jason Jones wrote : > Fusion between gospel and punk, this whole album was produced in > Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) > plugins. > > http://tinyboats.bandcamp.com/ > > Any thoughts or comments? The songs are really nice. The mix is clear and the little details came out clearly. The louder songs are a bit tiring after a while, the bass and kick is a bit too much for me if listened on headphones (Focusrite HP60, I think it's a relabeled Superlux HD662, and it is a bit boomy), but it's good on earphones and tiny laptop speakers. Haven't tried to listen it on a proper monitor. I love the toms. Overall, they're superb!!! I'm curious though, did they go through maximizers/limiters? If they did, how hot did you push them for the louder songs? From gnome at hawaii.rr.com Wed Feb 5 07:00:28 2014 From: gnome at hawaii.rr.com (david) Date: Tue, 04 Feb 2014 21:00:28 -1000 Subject: [LAU] Album produced in Linux In-Reply-To: <52F1D8E1.9050500@wootangent.net> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> Message-ID: <52F1E18C.4070702@hawaii.rr.com> On 02/04/2014 08:23 PM, Leigh Dyer wrote: > On 5/02/2014 11:37 am, jonetsu at teksavvy.com wrote: >> On Tue, 4 Feb 2014 14:31:58 -0700, >> Jason Jones wrote : >> >>> Fusion between gospel and punk, this whole album was produced in >>> Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) >>> plugins. >>> >>> http://tinyboats.bandcamp.com/ >>> >>> Any thoughts or comments? I love it! Sometimes it seems to me that Linux audio is a few scattered specialist musical ghettos. Your album is definitely not stuck in a ghetto! Nicely done. Listening to it on stereo headphones through an external soundcard running into a stereo amp. Sometimes sounds a little on the muddy side, depending on the instrument mix. Just my personal opinion, but I think you guys have great commercial potential. You're good at writing songs, playing and singing. Any videos of you guys performing? >> The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. Can >> play youtube all right. All enabled in No Script for this bandcamp >> page. > > For Firefox on Linux, you'll need to have actual Adobe Flash installed > -- YMMV if you have one of the open-source flash plugins installed. If > you have Flash installed, and you're not blocking any scripts, then > everything should work. > > Otherwise, you can use a browser that supports HTML5 MP3 playback; right > now on Linux, that means Chrome, or Chromium with "extra" ffmpeg codecs > installed (on Ubuntu etc. that's the chromium-codecs-ffmpeg-extra > package). Firefox has started to support MP3 on Windows and OS X using > codecs supplied by the OS, and it looks like the Linux version may do > this soon, too, using GStreamer, but it's not quite there yet. > > (Full disclosure -- I work for Bandcamp, though I'm replying from my > personal address, since that's what I'm subscribed to the list with). > > Thanks > Leigh Firefox 24.2 on Linux with Adobe Flash installed. Works fine after it took me about two years to figure out how to make Adobe Flash use my external soundcard instead of the sucky internal sound ... Flash sucks. And HTML5 video is slow and klunky unless you have a lot of CPU and graphics adapter hardware to throw at it. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From cbannister at slingshot.co.nz Wed Feb 5 08:39:20 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Wed, 5 Feb 2014 21:39:20 +1300 Subject: [LAU] Album produced in Linux In-Reply-To: <52F1E18C.4070702@hawaii.rr.com> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <52F1E18C.4070702@hawaii.rr.com> Message-ID: <20140205083920.GB27317@tal> On Tue, Feb 04, 2014 at 09:00:28PM -1000, david wrote: > > I love it! Sometimes it seems to me that Linux audio is a few > scattered specialist musical ghettos. Your album is definitely not > stuck in a ghetto! What's wrong with "In The Ghetto" Elvis Presley did a good version. :) SCNR. -- "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 mlang at delysid.org Wed Feb 5 10:03:52 2014 From: mlang at delysid.org (Mario Lang) Date: Wed, 05 Feb 2014 11:03:52 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <52F0F701.1000002@gmx.net> (Florian Paul Schmidt's message of "Tue, 04 Feb 2014 15:19:45 +0100") References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> <52F0F701.1000002@gmx.net> Message-ID: <87vbwtvptj.fsf@fx.delysid.org> Florian Paul Schmidt writes: > On 04.02.2014 14:58, Patrick Shirkey wrote: >> Teqqer also looks very promising. > > Mario even commited some bugfixes/improvements to the teq library. But > then seemingly lost interest ;D Heh, no, I just went away for a few days :-) -- CYa, ????? | Debian Developer .''`. | Get my public key via finger mlang/key at db.debian.org : :' : | 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44 `. `' `- From mlang at delysid.org Wed Feb 5 10:13:23 2014 From: mlang at delysid.org (Mario Lang) Date: Wed, 05 Feb 2014 11:13:23 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> (Patrick Shirkey's message of "Wed, 5 Feb 2014 00:58:19 +1100 (EST)") References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> Message-ID: <87r47hvpdo.fsf@fx.delysid.org> "Patrick Shirkey" writes: > On Wed, February 5, 2014 12:31 am, Mario Lang wrote: >> raf writes: >> >>> Hello, >>> >>> you'l probably be happy to know the existence of three great tools : >>> midish, linuxsampler and Nama. >>> 1) midish is a command line midi sequencer with a lot of great features >>> http://www.midish.org/ >> >> midish looks rather interesting. However, the manual.html basically >> just explains how to record data from an input device. Does latest >> midish support creating MIDI data from scratch, and if so, is there >> perhaps some examples on how to do that? >> > > Check this section : > > http://www.midish.org/manual.html#ev > > You can compose note on/off events and save the sequence as a song or > export the song to .mid A simple example on how to actually do that would be appreciated. > There are also these two other options: > > alsaseq: http://pp.com.mx/python/alsaseq/project.html > > mididings: http://das.nasophon.de/mididings/ Hmm, interesting, thanks for the tip. -- CYa, ????? | Debian Developer .''`. | Get my public key via finger mlang/key at db.debian.org : :' : | 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44 `. `' `- From mista.tapas at gmx.net Wed Feb 5 10:26:23 2014 From: mista.tapas at gmx.net (Florian Paul Schmidt) Date: Wed, 05 Feb 2014 11:26:23 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <87vbwtvptj.fsf@fx.delysid.org> References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> <52F0F701.1000002@gmx.net> <87vbwtvptj.fsf@fx.delysid.org> Message-ID: <52F211CF.3080200@gmx.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Sorry for the duplicate, Mario, I still don't have my mail program under control ;D On 05.02.2014 11:03, Mario Lang wrote: > Florian Paul Schmidt writes: > >> On 04.02.2014 14:58, Patrick Shirkey wrote: >>> Teqqer also looks very promising. >> >> Mario even commited some bugfixes/improvements to the teq >> library. But then seemingly lost interest ;D > > Heh, no, I just went away for a few days :-) > Ah cool :D I thought you didn't like me or my code anymore ;D Flo -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.14 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iQEcBAEBAgAGBQJS8hHNAAoJEA5f4Coltk8Zf4EIAIiFYODDXDtCfwgncOyIpg1E d0A2NgjFXO+p+5LK4l6cneTBPQqION4QDuIieHDNW0udrSuT+86m7puZGl/zRoce o+ea0LY/GEwRaSnS+hTkkSdWspM+vYzU4YZzkbGhhPPcgGmctHcjlYcRzEzJMyam PaHDvgNVguyYgNxJbvpQ1GcO1AQfDHRxb2CkzCIFNoqkNhpev1aAKdQUy68GuVds ke4ItFSPnqKxICvq6lZIJd6aLwQskln/juckAmXWGExRga92UnSLd14Mxkid3Twg gn8guzebyN4ato+56eYQfY855SPtodHmu1gFHbmgdaF9OUfBfEoWMp3tI1/hISc= =zOSz -----END PGP SIGNATURE----- From rmouneyres at gmail.com Wed Feb 5 10:41:13 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Wed, 5 Feb 2014 11:41:13 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: References: Message-ID: 2014-02-04, Jason Jones : > Fusion between gospel and punk, this whole album was produced in Linux, > using Harrison Mixbus and only Linux-native (LV2 / LADSPA) plugins. > > http://tinyboats.bandcamp.com/ > > Any thoughts or comments? > > --Jason > www.artcitysound.com (Linux-based studio) Yo ! some serious rock music made with linux ! You've made some great mixing here, i agree about the great toms, the reverb make them sound ala Phil Collins, and that's good. Just my personal taste, the hihat is a bit too loud and roomy. Many things i would say, but ony one i'm saying : great work ! Rapha?l From gerhard.zintel at web.de Wed Feb 5 12:28:03 2014 From: gerhard.zintel at web.de (Gerhard Zintel) Date: Wed, 5 Feb 2014 13:28:03 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: <52F1D8E1.9050500@wootangent.net> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> Message-ID: <201402051328.03349.gerhard.zintel@web.de> On Wednesday 05 February 2014, Leigh Dyer wrote: > On 5/02/2014 11:37 am, jonetsu at teksavvy.com wrote: > > On Tue, 4 Feb 2014 14:31:58 -0700, > >> http://tinyboats.bandcamp.com/ > >> > >> Any thoughts or comments? > > > > The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. Can > > play youtube all right. All enabled in No Script for this bandcamp > > page. > > For Firefox on Linux, you'll need to have actual Adobe Flash installed > -- YMMV if you have one of the open-source flash plugins installed. If > you have Flash installed, and you're not blocking any scripts, then > everything should work. I have the same problem with Firefox under Linux (Mate Olivia). My Flash Plugin is: Shockwave Flash 11,2,202,332. Any other recommendations? The interesting thing is that - as soon as I press the play button - I got a new Jack output client (alsa-jack.jackP.6913.2 with out_000 and out_001 but it is not connected to the system playbacks and I'm not able to connect by myself in Qjackctl. Frpm another hint: > I'm using firefox without flash and bandcamp works for me as long as I allow > bandcamp.com and bcbits.com (I think that's their cdn) in noscript. I don't know how to allow anything in noscripts. Any hint is highly appreciated. Gerhard From murks at tuxfamily.org Wed Feb 5 12:34:57 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Wed, 5 Feb 2014 13:34:57 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: <52F1D8E1.9050500@wootangent.net> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> Message-ID: <20140205133457.01279eb0@eeyore.mozart.uni-klu.ac.at> On Wed, 05 Feb 2014 17:23:29 +1100 Leigh Dyer wrote: > On 5/02/2014 11:37 am, jonetsu at teksavvy.com wrote: > > On Tue, 4 Feb 2014 14:31:58 -0700, > > Jason Jones wrote : > > > >> Fusion between gospel and punk, this whole album was produced in > >> Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) > >> plugins. > >> > >> http://tinyboats.bandcamp.com/ > >> > >> Any thoughts or comments? > > > > The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. > > Can play youtube all right. All enabled in No Script for this > > bandcamp page. > > For Firefox on Linux, you'll need to have actual Adobe Flash > installed -- YMMV if you have one of the open-source flash plugins > installed. If you have Flash installed, and you're not blocking any > scripts, then everything should work. > > Otherwise, you can use a browser that supports HTML5 MP3 playback; > right now on Linux, that means Chrome, or Chromium with "extra" > ffmpeg codecs installed (on Ubuntu etc. that's the > chromium-codecs-ffmpeg-extra package). Firefox has started to support > MP3 on Windows and OS X using codecs supplied by the OS, and it looks > like the Linux version may do this soon, too, using GStreamer, but > it's not quite there yet. > > (Full disclosure -- I work for Bandcamp, though I'm replying from my > personal address, since that's what I'm subscribed to the list with). > > Thanks > Leigh I too have issues, FF26 and FF27, flash installed. Flash playback typically works if I allow the necessary bits and pieces. I can get soundcloud, youtube and so on to work this way. I do allow the page to load stuff from bcbits.com and allow execution of scripts from bcbits.com and bandcamp.com. This should be enough, since the remaining scripts, from quantserve.com, google.com, google-analytics.com, facebook.com and facebook.net, are garbage in my book and should not be necessary to play music. Even if I allow those garbage scripts, it does not play. For me the play button does something, it gets replaced with some spinning circle animation and it seems to attempt to play each song in the playlist, without success. Regards, Philipp -- JID: murks at jit.si From tim at quitte.de Wed Feb 5 12:34:53 2014 From: tim at quitte.de (Tim Goetze) Date: Wed, 5 Feb 2014 13:34:53 +0100 (CET) Subject: [LAU] [ANN] CAPS 0.9.17 Message-ID: CAPS 0.9.17 =========== http://quitte.de/dsp/caps.html The latest release of CAPS, a collection of LADSPA plugins, contains two important bugfixes and minor sonic improvements. http://quitte.de/dsp/caps.html#CHANGES James Morris insisted that something was wrong with the PlateX2 stereo reverb, and indeed, it turned out that its damping parameter would not be read correctly. (This bug, it pains me to admit it, probably dates back to 0.9.0, possibly even earlier. In my defense, I only ever use the mono-to-stereo version which was not affected.) Ricardo Crudo isolated a potential segmentation fault triggered by the 'model' parameter selection of the CabinetIV plugin. Thanks guys! http://quitte.de/dsp/caps.html#Download Upgrading is recommended. Enjoy, Tim From ralf.mardorf at rocketmail.com Wed Feb 5 12:47:57 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Wed, 05 Feb 2014 13:47:57 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: <201402051328.03349.gerhard.zintel@web.de> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> Message-ID: <1391604477.701.77.camel@archlinux> On Wed, 2014-02-05 at 13:28 +0100, Gerhard Zintel wrote: > Any other recommendations? Don't install falshplayer and don't care about websites that need flashplayer, simply ignore those bad websites. If you really need to visit a site that requires flashplayer, anyway don't install flashplayer, but install google-chrome-stable. "Albums produced in Linux" and posted on websites that require proprietary Adobe stuff IMO are not really interesting. It could be considered as halfhearted. IMO it's more punk-rock to produce on what ever gear is available, but to publish the art by something that is accessible by everybody, that doesn't require non-open source codecs, non-open source software. That's just who I am. IMO people are free to produce on Linux and than they even could publish using DRM, just I won't care about those publications. Years ago I was pissed when a Linux musician posted on this website, since when I clicked on play a window popped up and abused me by sarcastically asking why I'm using the Internet, when I'm using "I don't like the Internet add-ons/settings". From murks at tuxfamily.org Wed Feb 5 12:50:18 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Wed, 5 Feb 2014 13:50:18 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: <20140205133457.01279eb0@eeyore.mozart.uni-klu.ac.at> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <20140205133457.01279eb0@eeyore.mozart.uni-klu.ac.at> Message-ID: <20140205135018.6c3f42b9@eeyore.mozart.uni-klu.ac.at> On Wed, 5 Feb 2014 13:34:57 +0100 Philipp ?berbacher wrote: > On Wed, 05 Feb 2014 17:23:29 +1100 > Leigh Dyer wrote: > > > On 5/02/2014 11:37 am, jonetsu at teksavvy.com wrote: > > > On Tue, 4 Feb 2014 14:31:58 -0700, > > > Jason Jones wrote : > > > > > >> Fusion between gospel and punk, this whole album was produced in > > >> Linux, using Harrison Mixbus and only Linux-native (LV2 / LADSPA) > > >> plugins. > > >> > > >> http://tinyboats.bandcamp.com/ > > >> > > >> Any thoughts or comments? > > > > > > The play button doe snto work. Firefox 22.0, Linux Mint 64 bits. > > > Can play youtube all right. All enabled in No Script for this > > > bandcamp page. > > > > For Firefox on Linux, you'll need to have actual Adobe Flash > > installed -- YMMV if you have one of the open-source flash plugins > > installed. If you have Flash installed, and you're not blocking any > > scripts, then everything should work. > > > > Otherwise, you can use a browser that supports HTML5 MP3 playback; > > right now on Linux, that means Chrome, or Chromium with "extra" > > ffmpeg codecs installed (on Ubuntu etc. that's the > > chromium-codecs-ffmpeg-extra package). Firefox has started to > > support MP3 on Windows and OS X using codecs supplied by the OS, > > and it looks like the Linux version may do this soon, too, using > > GStreamer, but it's not quite there yet. > > > > (Full disclosure -- I work for Bandcamp, though I'm replying from > > my personal address, since that's what I'm subscribed to the list > > with). > > > > Thanks > > Leigh > > I too have issues, FF26 and FF27, flash installed. Flash playback > typically works if I allow the necessary bits and pieces. I can get > soundcloud, youtube and so on to work this way. > > I do allow the page to load stuff from bcbits.com and allow execution > of scripts from bcbits.com and bandcamp.com. > This should be enough, since the remaining scripts, from > quantserve.com, google.com, google-analytics.com, facebook.com and > facebook.net, are garbage in my book and should not be necessary to > play music. Even if I allow those garbage scripts, it does not play. > > For me the play button does something, it gets replaced with some > spinning circle animation and it seems to attempt to play each song in > the playlist, without success. > > Regards, > Philipp Oh, I doubt it matters, but it works with neither of the two flash-implementations. flashplugin-11.2.202.336 shumway 0.8.11 Regards, Philipp -- JID: murks at jit.si From arve.barsnes at gmail.com Wed Feb 5 12:51:30 2014 From: arve.barsnes at gmail.com (Arve Barsnes) Date: Wed, 5 Feb 2014 13:51:30 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: <1391604477.701.77.camel@archlinux> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: On 5 February 2014 13:47, Ralf Mardorf wrote: > "Albums produced in Linux" and posted on websites that require > proprietary Adobe stuff IMO are not really interesting. It could be > considered as halfhearted. IMO it's more punk-rock to produce on what > ever gear is available, but to publish the art by something that is > accessible by everybody, that doesn't require non-open source codecs, > non-open source software. > Unfortunately bandcamp is hands down the best place to put your music, and I know of no alternative that even comes close. Let's hope firefox fixes its mp3 support sooner rather than later so you can also use this page. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandre.prokoudine at gmail.com Wed Feb 5 13:02:12 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Wed, 5 Feb 2014 17:02:12 +0400 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: On Wed, Feb 5, 2014 at 4:51 PM, Arve Barsnes wrote: > Unfortunately bandcamp is hands down the best place to put your music, and I > know of no alternative that even comes close. Are there any major issues with SoundCloud? Alexandre From ralf.mardorf at rocketmail.com Wed Feb 5 13:14:56 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Wed, 05 Feb 2014 14:14:56 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: <1391606096.701.94.camel@archlinux> On Wed, 2014-02-05 at 13:51 +0100, Arve Barsnes wrote: > On 5 February 2014 13:47, Ralf Mardorf > wrote: > "Albums produced in Linux" and posted on websites that require > proprietary Adobe stuff IMO are not really interesting. It > could be > considered as halfhearted. IMO it's more punk-rock to produce > on what ever gear is available, but to publish the art by > something that is accessible by everybody, that doesn't > require non-open source codecs, non-open source software. > Unfortunately bandcamp is hands down the best place to put your music, > and I know of no alternative that even comes close. Let's hope firefox > fixes its mp3 support sooner rather than later so you can also use > this page. No, I won't. They e.g. use Google Analytics. If they want to do marketing research or for what ever this should be "good" for, why don't they use e.g. Pwik Analytics? I'm not the punk-rock pope. People should do whatever they like, but I guess it's unlikely that people from my generation, born in 1966, IOW 80s punk-rock children, will care about rock'n'roll released on such platforms. JFTR I'm not a fan of google-chrome-stable, but it IMO is a better choice than flashplayer, just in case there should be the need to visit a site that requires flashplayer, perhaps a government agency or something like that. By having Chrome installed and not flashplayer, other web browsers aren't affected by a flashplayer install. Again, I don't judge people using what ever they want, people just should stop wining about offended data protection, capitalism, Babylon, if they participate on their own choice, while there absolutely is no need to do it. Assumed there shouldn't be a "good" platform to share music. Is there really the need to share music by the Internet? Regards, Ralf From alexandre.prokoudine at gmail.com Wed Feb 5 13:24:25 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Wed, 5 Feb 2014 17:24:25 +0400 Subject: [LAU] Album produced in Linux In-Reply-To: <1391606096.701.94.camel@archlinux> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> <1391606096.701.94.camel@archlinux> Message-ID: On Wed, Feb 5, 2014 at 5:14 PM, Ralf Mardorf wrote: > No, I won't. They e.g. use Google Analytics. If they want to do > marketing research or for what ever this should be "good" for, why don't > they use e.g. Pwik Analytics? With its stone-age IP-based metrics, lack of demography data, as well as a total zero of on-page analytics? Are you friggin kidding me? :) Piwik is simply lightyears behind GA. Alexandre From ralf.mardorf at rocketmail.com Wed Feb 5 13:39:15 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Wed, 05 Feb 2014 14:39:15 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> <1391606096.701.94.camel@archlinux> Message-ID: <1391607555.701.101.camel@archlinux> On Wed, 2014-02-05 at 17:24 +0400, Alexandre Prokoudine wrote: > On Wed, Feb 5, 2014 at 5:14 PM, Ralf Mardorf wrote: > > > No, I won't. They e.g. use Google Analytics. If they want to do > > marketing research or for what ever this should be "good" for, why don't > > they use e.g. Pwik Analytics? > > With its stone-age IP-based metrics, lack of demography data, as well > as a total zero of on-page analytics? Are you friggin kidding me? :) > Piwik is simply lightyears behind GA. For what usage does bandcamp need all this and why do they share the data with Google, IOW with all kinds of well known and completely unknown third parties? Why should a rock'n'roll/punk-rock band being interested to participate? Why should listeners being interested to participate? I'm not interested to participate. I'm willing to turn a blind eye, if somebody does use Piwik instead of Google. From arve.barsnes at gmail.com Wed Feb 5 14:02:43 2014 From: arve.barsnes at gmail.com (Arve Barsnes) Date: Wed, 5 Feb 2014 15:02:43 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: On 5 February 2014 14:02, Alexandre Prokoudine < alexandre.prokoudine at gmail.com> wrote: > On Wed, Feb 5, 2014 at 4:51 PM, Arve Barsnes wrote: > > > Unfortunately bandcamp is hands down the best place to put your music, > and I > > know of no alternative that even comes close. > > Are there any major issues with SoundCloud? > > Besides the whole site seemingly being geared towards single songs, I really think the whole design is a mess. And contrary to the single song angle that many use the site for, if you play a song, it automatically starts playing the next song in that user's account, which I personally find extremely annoying. The last time I tried using it, there were two technical barriers (that might not be there any more, I haven't checked), namely a ridiculous low limit on file size, and a need to convert any file to 44.1/16 before uploading. Showstopper, and I have not bothered with the site since. On 5 February 2014 14:14, Ralf Mardorf wrote: >No, I won't. They e.g. use Google Analytics. If they want to do >marketing research or for what ever this should be "good" for, why don't >they use e.g. Pwik Analytics? I don't see any reason why that would stop you, if you have so much against it you should have it blocked (as I have). It is not necessary to use the site. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandre.prokoudine at gmail.com Wed Feb 5 14:13:05 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Wed, 5 Feb 2014 18:13:05 +0400 Subject: [LAU] Album produced in Linux In-Reply-To: <1391607555.701.101.camel@archlinux> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> <1391606096.701.94.camel@archlinux> <1391607555.701.101.camel@archlinux> Message-ID: On Wed, Feb 5, 2014 at 5:39 PM, Ralf Mardorf wrote: >> With its stone-age IP-based metrics, lack of demography data, as well >> as a total zero of on-page analytics? Are you friggin kidding me? :) >> Piwik is simply lightyears behind GA. > > For what usage does bandcamp need all this For instance, on-page analytics has _everything_ to do with maintaining _any_ website worth caring about. It helps understanding and visualizing, how users think, what they desire, and how they get it (or how they fail to get it). I'm sorry (actually, no, I'm not), but the market doesn't work like that: people don't choose a worse tool for the job _en masse_. I do like the fact that someone's trying to make a decent free/open source web analytics tool, but so far I can't get all the actionable data I need from Piwik, and I don't expect maintainers of much larger websites to have lower demands. I also don't expect Piwik to ever get certain features that are only possible when there's either a centralized entity with a single database behind it, or large federated networks (and I mean _really_ large). > Why should a rock'n'roll/punk-rock band being > interested to participate? Why should listeners being interested to > participate? Another question: why should they care? Alexandre From alexandre.prokoudine at gmail.com Wed Feb 5 14:17:34 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Wed, 5 Feb 2014 18:17:34 +0400 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: On Wed, Feb 5, 2014 at 6:02 PM, Arve Barsnes wrote: >> Are there any major issues with SoundCloud? >> > Besides the whole site seemingly being geared towards single songs, I really > think the whole design is a mess. And contrary to the single song angle that > many use the site for, if you play a song, it automatically starts playing > the next song in that user's account, which I personally find extremely > annoying. Doesn't it annoy you (to any degree) that bandcamp tends to always start playing an album from the second song? > The last time I tried using it, there were two technical barriers (that > might not be there any more, I haven't checked), namely a ridiculous low > limit on file size, and a need to convert any file to 44.1/16 before > uploading. Showstopper, and I have not bothered with the site since. OK, that's understandable. Alexandre From arve.barsnes at gmail.com Wed Feb 5 14:21:08 2014 From: arve.barsnes at gmail.com (Arve Barsnes) Date: Wed, 5 Feb 2014 15:21:08 +0100 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> Message-ID: On 5 February 2014 15:17, Alexandre Prokoudine < alexandre.prokoudine at gmail.com> wrote: > Doesn't it annoy you (to any degree) that bandcamp tends to always > start playing an album from the second song? > Yes, it's very annoying, but it's not bandcamp's fault. When you upload an album, the uploader chooses which song should be the "featured" song, that is the one that will play when you press the play button at the top. I guess some people don't think their album opener is good enough to get people to keep listening? -------------- next part -------------- An HTML attachment was scrubbed... URL: From matthew.garman at gmail.com Wed Feb 5 15:36:37 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Wed, 5 Feb 2014 09:36:37 -0600 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? Message-ID: <20140205153636.GA29740@septictank.raw-sewage.fake> I have a collection of FLAC files, all ripped from my CD collection What I would like to do is run an analysis across all the music to determine how the bass/lower frequencies are generally mixed. For example, how much content below (for example) 150 Hz is on the left channel versus the right channel? I'm not sure if "histogram" is the right word, but in my mind what I'd like to see, per-channel, is something like this: 150--125 Hz: x samples 125--100 Hz: y samples 100--80 Hz: z samples ... Then I can look at the two channels of a song, and if the histograms are approximately the same, I can assume the bass was mixed equally to both channels. I am a programmer, and thought it would be easy to quickly hack something up that would do this, but I have no experience with signal processing, and as I started reading about this, I quickly got in over my head! So I was hoping there might already exist a tool that has this functionality. Note that I don't need any kind of graphical output, as this needs to be wrapped up in some kind of batch processing script---I have about 11,000 files to analyze! The motivation for this is: I have a hardware DAC (digital audio converter) in one part of my house, and a subwoofer in another. There is a single coax run between the DAC and subwoofer, so I can only send one channel. If the overwhelming majority of my music has the bass mixed equally, sending only one channel isn't a problem. But if I choose the "L" channel to send to the sub, and much music has the bass mixed only to the "R" channel, then I won't be able to hear the low frequencies. I want to find out how often this might happen. Thanks, Matt From murks at tuxfamily.org Wed Feb 5 15:46:50 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Wed, 5 Feb 2014 16:46:50 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: <20140205164650.54b41db6@eeyore.mozart.uni-klu.ac.at> On Wed, 5 Feb 2014 09:36:37 -0600 Matt Garman wrote: > > I have a collection of FLAC files, all ripped from my CD collection > What I would like to do is run an analysis across all the music to > determine how the bass/lower frequencies are generally mixed. For > example, how much content below (for example) 150 Hz is on the left > channel versus the right channel? > > I'm not sure if "histogram" is the right word, but in my mind what > I'd like to see, per-channel, is something like this: > > 150--125 Hz: x samples > 125--100 Hz: y samples > 100--80 Hz: z samples > ... > > Then I can look at the two channels of a song, and if the histograms > are approximately the same, I can assume the bass was mixed equally > to both channels. > > I am a programmer, and thought it would be easy to quickly hack > something up that would do this, but I have no experience with > signal processing, and as I started reading about this, I quickly > got in over my head! So I was hoping there might already exist a > tool that has this functionality. One thing to note: According to my understanding you won't get accurate sample counts here. To do frequency analysis you'll need to look at a bunch of samples at a time. Typically you'll use some sort of Fourier transform. So your granularity will be limited by the window size I guess. I don't think this is very important in this case though, but maybe it gets you on the track. > Note that I don't need any kind of graphical output, as this needs > to be wrapped up in some kind of batch processing script---I have > about 11,000 files to analyze! > > The motivation for this is: I have a hardware DAC (digital audio > converter) in one part of my house, and a subwoofer in another. > There is a single coax run between the DAC and subwoofer, so I can > only send one channel. If the overwhelming majority of my music has > the bass mixed equally, sending only one channel isn't a problem. > But if I choose the "L" channel to send to the sub, and much music > has the bass mixed only to the "R" channel, then I won't be able to > hear the low frequencies. I want to find out how often this might > happen. > > Thanks, > Matt Can you mix the bass channel to mono? That would be the simplest solution. Regards, Philipp From jh at brainiac.com Wed Feb 5 16:28:15 2014 From: jh at brainiac.com (Joe Hartley) Date: Wed, 5 Feb 2014 11:28:15 -0500 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: <20140205112815.beddd16f5ebbc3252368f1fe@brainiac.com> On Wed, 5 Feb 2014 09:36:37 -0600 Matt Garman wrote: > The motivation for this is: I have a hardware DAC (digital audio > converter) in one part of my house, and a subwoofer in another. > There is a single coax run between the DAC and subwoofer, so I can > only send one channel. If the overwhelming majority of my music has > the bass mixed equally, sending only one channel isn't a problem. > But if I choose the "L" channel to send to the sub, and much music > has the bass mixed only to the "R" channel, then I won't be able to > hear the low frequencies. I want to find out how often this might > happen. Bass is almost always mixed to the center as low frequencies are omnidirectional. If you only send the L or R channel, you'll lose some of the signal strength (about 6dB if I remember correctly) and the subwoofer won't have the "oomph" it should, but since you'll need an amp after the DAC anyway you should be able to make that up easily. That said, if you're listening to old Beatles stereo mixes, the bass will only be on one channel (left, usually) and if you've sent the right to the DAC, there's not much for the sub. I'd try and convert the signal to mono for the sub feed. -- ====================================================================== 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 Wed Feb 5 17:20:03 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Wed, 05 Feb 2014 18:20:03 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205112815.beddd16f5ebbc3252368f1fe@brainiac.com> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140205112815.beddd16f5ebbc3252368f1fe@brainiac.com> Message-ID: <1391620803.701.119.camel@archlinux> On Wed, 2014-02-05 at 11:28 -0500, Joe Hartley wrote: > Bass is almost always mixed to the center > > That said, if you're listening to old Beatles stereo mixes, the bass > will only be on one channel Correct and not only the bass has got deep frequencies and not only the Beatles mixed that hard left/right for the older stereo recordings, but when I read your first sentence, I was thinking of the Beatles too, to disagree with that claim. Btw. those old Beatles recordings aren't bad mixes, many of them are very good. Right now I'm listening to the record single Eleanor Rigby by head phones, backside of Yellow Submarine. It's even pleasant to listen to such a hard left/right mix on headphones, quite psychedelic what they did to the vocal mix by separating it. I read that all Beatles recordings were remixed by compressing them to the loudness war, if so, then for sure the balanced hard left/right stereo mix won't be balanced and psychedelic anymore, since an important part of those mixings is the dynamic range. From poeticintensity at gmail.com Wed Feb 5 18:15:12 2014 From: poeticintensity at gmail.com (Jason Jones) Date: Wed, 5 Feb 2014 11:15:12 -0700 Subject: [LAU] Album produced in Linux In-Reply-To: <52F1E18C.4070702@hawaii.rr.com> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <52F1E18C.4070702@hawaii.rr.com> Message-ID: > > > I love it! Sometimes it seems to me that Linux audio is a few scattered > specialist musical ghettos. Your album is definitely not stuck in a ghetto! > > Nicely done. Listening to it on stereo headphones through an external > soundcard running into a stereo amp. Sometimes sounds a little on the muddy > side, depending on the instrument mix. > > Just my personal opinion, but I think you guys have great commercial > potential. You're good at writing songs, playing and singing. Any videos of > you guys performing? > > Thank you so much, David! The album started in October of 2012, so it's been in the works for over a year. I appreciate your kind words, and sadly, the band is composed of only two people. We don't have enough bandmates to perform these songs live yet, so no videos yet, but we're working on it. Thanks again! --Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From poeticintensity at gmail.com Wed Feb 5 18:24:14 2014 From: poeticintensity at gmail.com (Jason Jones) Date: Wed, 5 Feb 2014 11:24:14 -0700 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <20140204203825.1dae1fbb@mistral> Message-ID: > > > The songs are really nice. The mix is clear and the little details > came out clearly. > The louder songs are a bit tiring after a while, the bass and kick is > a bit too much for me > if listened on headphones (Focusrite HP60, I think it's a relabeled > Superlux HD662, and it is a bit boomy), > but it's good on earphones and tiny laptop speakers. Haven't tried to > listen it on a proper monitor. > I love the toms. > Overall, they're superb!!! > > I'm curious though, did they go through maximizers/limiters? > If they did, how hot did you push them for the louder songs? > _______________________________________________ > Thank you! When I mastered the songs, I used the LinuxDSP's Black-EQ (10-band parametric), and a multi-band compressor. No limiters, but I did push it up to levels that can compete with radio-ready mixes. Anyone know of any Linux plugins to get the overall loudness of a mix? I haven't found one yet. Anyway, thank you for the kind words! --Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From poeticintensity at gmail.com Wed Feb 5 18:25:39 2014 From: poeticintensity at gmail.com (Jason Jones) Date: Wed, 5 Feb 2014 11:25:39 -0700 Subject: [LAU] Album produced in Linux In-Reply-To: References: Message-ID: > > > Yo ! some serious rock music made with linux ! > You've made some great mixing here, i agree about the great toms, the > reverb make them sound ala Phil Collins, and that's good. > Just my personal taste, the hihat is a bit too loud and roomy. Many > things i would say, but ony one i'm saying : great work ! > > Rapha?l > Thank you, Raphael! It's good to hear that people like the songs and mix after spending a long time on them. Thank you! --Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From brent at keycorner.org Wed Feb 5 21:47:35 2014 From: brent at keycorner.org (Brent Busby) Date: Wed, 5 Feb 2014 15:47:35 -0600 (CST) Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: On Wed, 5 Feb 2014, Matt Garman wrote: > I have a collection of FLAC files, all ripped from my CD collection > What I would like to do is run an analysis across all the music to > determine how the bass/lower frequencies are generally mixed. For > example, how much content below (for example) 150 Hz is on the left > channel versus the right channel? > > I'm not sure if "histogram" is the right word, but in my mind what I'd > like to see, per-channel, is something like this: > > 150--125 Hz: x samples > 125--100 Hz: y samples > 100--80 Hz: z samples > ... > > Then I can look at the two channels of a song, and if the histograms > are approximately the same, I can assume the bass was mixed equally to > both channels. > > I am a programmer, and thought it would be easy to quickly hack > something up that would do this, but I have no experience with signal > processing, and as I started reading about this, I quickly got in over > my head! So I was hoping there might already exist a tool that has > this functionality. [...] This is a subject which, as someone getting more and more immersed in recording, I find very interesting: Everyone pretty much seems to agree that a perfect loudspeaker is physically impossible. We also have mastering engineers because even if there were perfect loudspeakers, most people wouldn't use them, thus emerges the black art of creating a mix that will sound good on "most of what most people are using," whatever that means. If you did try to perfect your speakers' EQ curve, waterfall plot, etc., you might not even like the result, since people actually like coloration in their system when they listen for enjoyment (instead of mixing). Basically, this means that if one were to look at all of the listeners and recording engineers as a whole, they are following eachothers guidance like a bunch of lost travelers driving in the fog, following eachother's car tail lights: The speaker manufacturers don't have a standard. They're making speakers that sound good with most of the music out there. The recording engineers don't have a standard. They're mixing and mastering to sound good on most of the speakers. The listeners don't have a standard. They're EQ'ing their systems to make the music the engineers made with reference to the speakers people are using sound good on their speakers (which were made to sound good with most of the music). Do you notice a pattern here? :) Despite all of this, some sort of consensus has emerged from the fog, otherwise mastering would not even be possible, not even for old guys who "know kung fu" in this black art. The speaker companies, listeners, and engineers must have, however accidentally, reached a sort of meta-standard for what things should sound like if you want them to translate across audio reproduction systems. Maybe they weren't trying to...maybe they don't even realize it...but if they hadn't, mastering would be impossible. What you're talking about, as far as batch analysis of commercially produced music, surely would turn up some interesting information, if it was done properly by someone who knows audio analysis way better than I do. Context is everything with those sorts of measurements, so it would have to be done by someone who knows what to measure (not me). But I'm thinking if it was done right, you might get a peek at the parameters of the golden "meta-standard" that has emerged from all this tail light chasing that's going on, as far as what mixes are really aiming for in realy quantifiable numbers, whether the engineers doing them know it consciously or not. -- + 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 james at jwm-art.net Wed Feb 5 22:02:58 2014 From: james at jwm-art.net (James Morris) Date: Wed, 5 Feb 2014 22:02:58 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205112815.beddd16f5ebbc3252368f1fe@brainiac.com> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140205112815.beddd16f5ebbc3252368f1fe@brainiac.com> Message-ID: <20140205220258.71ba895e@Scrapyard.lan> On Wed, 5 Feb 2014 11:28:15 -0500 Joe Hartley wrote: > > That said, if you're listening to old Beatles stereo mixes, the bass > will only be on one channel (left, usually) and if you've sent the > right to the DAC, there's not much for the sub. I'd try and convert > the signal to mono for the sub feed. > This suprised me as I had heard that for vinyl it is particularly important to have mono bass to prevent the needle jumping :-) ie http://www.customrecords.com/prepare_music_for_vinyl_record.html From djdualcore at gmail.com Wed Feb 5 22:39:25 2014 From: djdualcore at gmail.com (Neil) Date: Wed, 5 Feb 2014 16:39:25 -0600 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: <52F148A1.5000006@autostatic.com> References: <52F148A1.5000006@autostatic.com> Message-ID: If you decide to go with a rack-mounted PC with internal slots things like this become an option. http://www.m-audio.com/products/en_us/Delta1010.html I totally understand this isn't useful with an existing laptop. On Tue, Feb 4, 2014 at 2:08 PM, Jeremy Jongepier wrote: > On 02/04/2014 05:02 PM, Jos? Agust?n Terol Sanchis wrote: > > Hello, I am thinking in set up a home studio for recording and mixing > based > > on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but first, I > > would like to know the experience of someone who has already used it. > > > > I've taken a look at http://www.ffado.org/?q=devicesupport/list but I > don't > > have clear if this device can be used at 96kHz, because currently it's > > marked as "experimental". I know other Saffire have "full support", but > > nowadays are discontinued. Does anybody used it for hours (and days) at > > this frequency? > > > > Also, I would be interested in experiences with similar interfaces that > > also work fine in linux. > > > > Thank you, > > Agus Terol > > Hello Agus, > > I can confirm the Saffire Pro 40 works at 96Khz. I barely use this > sample rate though so I can't tell if it will run for hours or even > days. I don't see any reason why it shouldn't. I can use my Saffire Pro > 40 for hours at 48Khz without any hitch. I've occasionally forgot to > switch the set-up off and the next day everything was still running fine. > > Best, > > Jeremy > > > > _______________________________________________ > 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 zettberlin at linuxuse.de Wed Feb 5 23:46:22 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Thu, 06 Feb 2014 00:46:22 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> Message-ID: <52F2CD4E.3030809@linuxuse.de> Sorry for getting into this so late.... Am 22.01.2014 22:14, schrieb Alexandre Prokoudine: > On Thu, Jan 23, 2014 at 12:52 AM, R. Mattes wrote: > >>> 1. Last time I checked, Novation, M-Audio, Roland etc. had no >>> OSC keyboards, just regular MIDI ones :) >> >> It would at least be a way to overcome the lack of LV2/LADSPA >> support > > Which, as pointed out earlier, isn't necessarily such a big deal. > My point is, most of us haven't had a go at beta versions of Bitwig > yet, hence there's no knowing, how good/bad the built-in plugins > are. They are quite usable and complete as in "every basic thing is available". The synths are too simple to replace big LV2-plugins such as Calf Organ but they sound pretty OK, EQ/Dynamics are solid and sound pretty good, as far as I remember there is a usable reverb but not a IR-convolver, this would be the only thing I'd really miss. best regards HZN > > No LV2 support in v1.0 had been publicly known for a long, long > time. And people were/are still ready to pay for the app. LADSPA? > Haven't used those for ages. > > Alexandre _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > From jonetsu at teksavvy.com Thu Feb 6 00:13:25 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Wed, 5 Feb 2014 19:13:25 -0500 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <20140204203825.1dae1fbb@mistral> Message-ID: <20140205191325.4cd72c3d@mistral> On Wed, 5 Feb 2014 13:46:10 +0700, Shani Hadiyanto Pribadi wrote : > I'm using firefox without flash and bandcamp works for me as long as > I allow bandcamp.com and bcbits.com (I think that's their cdn) in > noscript. Firefox outputs to alsa loopback, zalsa_in connects to > zita-mu1, zita-mu1 (with HP switch on) connects to hw out. That was it: allowing bcbits.com. Now plays all right. - thanks. > The way I see it soundcloud and bandcamp serves different purposes. > I feel that bandcamp is more suited for album releases, while > soundcloud is for the occasional track releases. > But there's no reason not to use both, some people might prefer > bandcamp's personalized and cleaner page, > some people might want to use soundcloud social features, like > comments and followers. I do not know much either of them, so I have some listening to do. The number of people making music ! From phaselocker at gmail.com Thu Feb 6 02:05:45 2014 From: phaselocker at gmail.com (Lewis Pike) Date: Wed, 5 Feb 2014 21:05:45 -0500 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <52F12378.7070700@parisson.com> References: <20140129024408.GA3961@ordinator> <52E8F06F.4080503@parisson.com> <20140129140546.GA4085@ordinator> <20140203151731.GA11566@ordinator> <52F12378.7070700@parisson.com> Message-ID: <20140206020545.GA13658@ordinator> Hi Guillaume, On Tue, Feb 04, 2014 at 06:29:28PM +0100, Guillaume Pellerin wrote: > Sorry Lewis, I was AFK these days. Back now :) > > Can you try to add this line to /etc/modprobe.d/alsa-base.conf > > options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x03 enable=1 nrpacks=1 > > and then: > $ sudo modprobe -r snd-usb-audio > $ sudo modprobe snd-usb-audio I added your suggested kernel module option line and rebooted my system for good measure. $ cat /proc/asound/card0/stream0 # now gives: M-Audio FastTrack Pro at usb-0000:00:1d.0-2, full speed : USB Audio Playback: Status: Stop Interface 2 Altset 2 Format: S24_3BE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 2 Altset 5 Format: S24_3BE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 8000 - 48000 (continuous) $ cat /proc/asound/card0/stream1 # now gives: M-Audio FastTrack Pro at usb-0000:00:1d.0-2, full speed : USB Audio #1 Playback: Status: Stop Interface 3 Altset 2 Format: S24_3BE Channels: 2 Endpoint: 4 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 3 Altset 5 Format: S24_3BE Channels: 2 Endpoint: 4 OUT (ADAPTIVE) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 4 Altset 2 Format: S24_3BE Channels: 2 Endpoint: 5 IN (SYNC) Rates: 8000 - 48000 (continuous) I've also updated my arecord test command to account for the the S24_3BE format: arecord -t raw -f S24_3BE -c2 -r 48000 -D hw:0,1 -vv | aplay -f S24_3BE -c2 -r 48000 -D hw:0,0 Unfortunately, I am still getting bad captures in about ~10% of my test cases. I have noticed a slightly different problem in this configuration: about half of the bad captures are not just a distorted signal, but instead very loud pure white noise. My ears are still ringing! > un/replug the card and make your tests again. > > I must admit I've almost always use the FTP with JACK so I'm not > completly aware of the behavior of the device without it. > > If this has no (good) effect, it could be something between you USB > system and the kernel. I've discover recently that crackles could > raise up on other cards and systems because of a dynamic USB ID > allocation of the kernel. The pb disappeared when the option is > switched off. > > So, let's continue if the first ALSA tweak doesn't work.. > > Regards, > Guillaume > 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 # CONFIG_USB_OTG is not set CONFIG_USB_MON=m CONFIG_USB_WUSB=m CONFIG_USB_WUSB_CBAF=m # CONFIG_USB_WUSB_CBAF_DEBUG is not set # -------------------------------------------- Again, your help on this one is much appreciated! .lewis From simonzwise at gmail.com Thu Feb 6 03:37:56 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 06 Feb 2014 14:37:56 +1100 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <201402051328.03349.gerhard.zintel@web.de> <1391604477.701.77.camel@archlinux> <1391606096.701.94.camel@archlinux> <1391607555.701.101.camel@archlinux> Message-ID: <52F30394.2050906@gmail.com> On 06/02/14 01:13, Alexandre Prokoudine wrote: > I'm sorry (actually, no, I'm not), but the market doesn't work like > that: people don't choose a worse tool for the job _en masse_. ... > > Another question: why should they care? > > Alexandre I care because I want continued access to non-centralise (peer to peer) distribution networks, for information and for cultural works. So I choose to work in a way that prefers these channels. That choice is related to the kind of society I want to live in, as such it is a deeply political choice. Huge centralised server networks certainly make it easier for the user to consume what is offered, and perhaps make distribution of approved works easier for a broad range of conforming artists. News Limited is a very convenient source of news and information, very dominant here but consistently full of bullshit. Information and cultural works are increasingly blocked from centralised server networks for all kinds of reasons, some of which certainly contradict my idea of appropriate content. Most people don't seem to care much how that information, music and other content is filtered or for what purposes. Here (in Australia) the level of secrecy with regard to public policy and decision making and the extent to which mainstream media is filled almost exclusively with vacuous nonsense is getting worse and worse. We seem to believe it is the right way to go ... at least when it comes to actually choosing via elections (voting is compulsory and preferential here, we have well over 90% voting with plenty of choice available). If this is to be the easy, mainstream way I will choose the more difficult, fringe path. My tastes are frugal, I'm not young and I don't have dependants so I'll probably escape the harshest effects of steady shift in social structures here. Most of us seem willing to embrace it, and so we will, but I can at least keep a little bit independent on a personal level and argue against it whenever I can. Simon From gnome at hawaii.rr.com Thu Feb 6 05:53:18 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 05 Feb 2014 19:53:18 -1000 Subject: [LAU] Album produced in Linux In-Reply-To: References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <52F1E18C.4070702@hawaii.rr.com> Message-ID: <52F3234E.9020205@hawaii.rr.com> On 02/05/2014 08:15 AM, Jason Jones wrote: > > I love it! Sometimes it seems to me that Linux audio is a few > scattered specialist musical ghettos. Your album is definitely not > stuck in a ghetto! > > Nicely done. Listening to it on stereo headphones through an > external soundcard running into a stereo amp. Sometimes sounds a > little on the muddy side, depending on the instrument mix. > > Just my personal opinion, but I think you guys have great commercial > potential. You're good at writing songs, playing and singing. Any > videos of you guys performing? > > Thank you so much, David! The album started in October of 2012, so it's > been in the works for over a year. I appreciate your kind words, and > sadly, the band is composed of only two people. We don't have enough > bandmates to perform these songs live yet, so no videos yet, but we're > working on it. Jason - Well, I look forward to your future success and hope you can find some bandmates for live performances. And you could embark on a project to make a music video of one or more of your songs. Don't need bandmates for that, you just film each of you playing your instruments and make the vid from that. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Thu Feb 6 05:55:49 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 05 Feb 2014 19:55:49 -1000 Subject: [LAU] Album produced in Linux In-Reply-To: <20140205083920.GB27317@tal> References: <20140204193718.395b253b@mistral> <52F1D8E1.9050500@wootangent.net> <52F1E18C.4070702@hawaii.rr.com> <20140205083920.GB27317@tal> Message-ID: <52F323E5.2050804@hawaii.rr.com> On 02/04/2014 10:39 PM, Chris Bannister wrote: > On Tue, Feb 04, 2014 at 09:00:28PM -1000, david wrote: >> >> I love it! Sometimes it seems to me that Linux audio is a few >> scattered specialist musical ghettos. Your album is definitely not >> stuck in a ghetto! > > What's wrong with "In The Ghetto" Elvis Presley did a good version. :) > SCNR. Yes, he did a good performance of it. Apparently he didn't use Linux to make it, though. But that's a slightly-different use of the word; mine's the older meaning without the pejoratives that have come to dominance now. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From alex at caoua.org Thu Feb 6 07:50:59 2014 From: alex at caoua.org (Alexandre Ratchov) Date: Thu, 6 Feb 2014 08:50:59 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <87r47hvpdo.fsf@fx.delysid.org> References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> <87r47hvpdo.fsf@fx.delysid.org> Message-ID: <20140206075059.GF2392@moule.localdomain> On Wed, Feb 05, 2014 at 11:13:23AM +0100, Mario Lang wrote: > "Patrick Shirkey" writes: > > > On Wed, February 5, 2014 12:31 am, Mario Lang wrote: > >> raf writes: > >> > >>> Hello, > >>> > >>> you'l probably be happy to know the existence of three great tools : > >>> midish, linuxsampler and Nama. > >>> 1) midish is a command line midi sequencer with a lot of great features > >>> http://www.midish.org/ > >> > >> midish looks rather interesting. However, the manual.html basically > >> just explains how to record data from an input device. Does latest > >> midish support creating MIDI data from scratch, and if so, is there > >> perhaps some examples on how to do that? > >> > > > > Check this section : > > > > http://www.midish.org/manual.html#ev > > > > You can compose note on/off events and save the sequence as a song or > > export the song to .mid > > A simple example on how to actually do that would be appreciated. Hi, It's kinda painful, as the tool was designed to work with an input device. You could create a track and add events one by one, ex: onew piano {0 0} tnew mytrack taddev 1 0 0 {non piano 64 90} taddev 1 1 0 {noff piano 64 0} see: http://www.midish.org/manual.html#func_taddev http://www.midish.org/manual.html#ev_ev you'll get warnings about unterminated notes and/or other anomalies, that you can ignore until all events are added. Once you're done, you could run tcheck to fix any anomaly, just in case. To make the process less painful, you could define routines to make certain things automatic, depending on your needs. HTH -- Alexandre From agus3985 at gmail.com Thu Feb 6 08:29:52 2014 From: agus3985 at gmail.com (=?ISO-8859-1?Q?Jos=E9_Agust=EDn_Terol_Sanchis?=) Date: Thu, 6 Feb 2014 09:29:52 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: <52F148A1.5000006@autostatic.com> Message-ID: Thank you for your answers and suggestions. BTW, Jeremy, I've been taking a look at the Saffire 40 user manual, and I have a doubt. When phantom power is turned on (for example, at 5-8 inputs), does this affect to input lines in the same segment (5-8)? I mean, it has a connector XLR/TRS combo, so, I wonder if the +48V only affects the XLR plug, or it will also power the TRS connection (that could damage some line-level stuff connected at TRS). Well, I think this question is more or less applicable to those interfaces with this kind of input combos. Regards, Agus Terol 2014-02-05 23:39 GMT+01:00 Neil : > > If you decide to go with a rack-mounted PC with internal slots things like > this become an option. > > http://www.m-audio.com/products/en_us/Delta1010.html > > I totally understand this isn't useful with an existing laptop. > > > On Tue, Feb 4, 2014 at 2:08 PM, Jeremy Jongepier wrote: > >> On 02/04/2014 05:02 PM, Jos? Agust?n Terol Sanchis wrote: >> > Hello, I am thinking in set up a home studio for recording and mixing >> based >> > on Linux. I have in mind to buy a Focusrite Saffire PRO 40, but first, I >> > would like to know the experience of someone who has already used it. >> > >> > I've taken a look at http://www.ffado.org/?q=devicesupport/list but I >> don't >> > have clear if this device can be used at 96kHz, because currently it's >> > marked as "experimental". I know other Saffire have "full support", but >> > nowadays are discontinued. Does anybody used it for hours (and days) at >> > this frequency? >> > >> > Also, I would be interested in experiences with similar interfaces that >> > also work fine in linux. >> > >> > Thank you, >> > Agus Terol >> >> Hello Agus, >> >> I can confirm the Saffire Pro 40 works at 96Khz. I barely use this >> sample rate though so I can't tell if it will run for hours or even >> days. I don't see any reason why it shouldn't. I can use my Saffire Pro >> 40 for hours at 48Khz without any hitch. I've occasionally forgot to >> switch the set-up off and the next day everything was still running fine. >> >> Best, >> >> Jeremy >> >> >> >> _______________________________________________ >> 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. > > _______________________________________________ > 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 unaudio at gmail.com Thu Feb 6 08:41:33 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Thu, 6 Feb 2014 10:41:33 +0200 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: On Wed, Feb 5, 2014 at 5:36 PM, Matt Garman wrote: > > I have a collection of FLAC files, all ripped from my CD collection > What I would like to do is run an analysis across all the music to > determine how the bass/lower frequencies are generally mixed. For > example, how much content below (for example) 150 Hz is on the left > channel versus the right channel? > > I'm not sure if "histogram" is the right word, but in my mind what > I'd like to see, per-channel, is something like this: > > 150--125 Hz: x samples > 125--100 Hz: y samples > 100--80 Hz: z samples > ... > > Then I can look at the two channels of a song, and if the histograms > are approximately the same, I can assume the bass was mixed equally > to both channels. > > I am a programmer, and thought it would be easy to quickly hack > something up that would do this, but I have no experience with > signal processing, and as I started reading about this, I quickly > got in over my head! So I was hoping there might already exist a > tool that has this functionality. > > Note that I don't need any kind of graphical output, as this needs > to be wrapped up in some kind of batch processing script---I have > about 11,000 files to analyze! > > The motivation for this is: I have a hardware DAC (digital audio > converter) in one part of my house, and a subwoofer in another. > There is a single coax run between the DAC and subwoofer, so I can > only send one channel. If the overwhelming majority of my music has > the bass mixed equally, sending only one channel isn't a problem. > But if I choose the "L" channel to send to the sub, and much music > has the bass mixed only to the "R" channel, then I won't be able to > hear the low frequencies. I want to find out how often this might > happen. > > Thanks, > Matt > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user One of the first rules of mixing you learn is to mix bass to the center. It has to do with the fact that bass carries the most energy and it makes sense to have both speakers share that load or some other crap like that. I don't think such an analysis is needed - an overwhelming amount of records will have bass at the center. Also when mixing for vinyl if bass is not centered it will make the needle jump out of the groove. From rosea.grammostola at gmail.com Thu Feb 6 10:02:11 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Thu, 6 Feb 2014 11:02:11 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F2CD4E.3030809@linuxuse.de> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: Isn't it a disadvantage of such all-in-one-daws that all music produced with it sounds pretty similar to each other, everybody uses the same samples and plugins. An creative advantage of modular linuxaudio could be that you've a high level of creativity and uniqueness in your music. On Thu, Feb 6, 2014 at 12:46 AM, Hartmut Noack wrote: > Sorry for getting into this so late.... > Am 22.01.2014 22:14, schrieb Alexandre Prokoudine: > > On Thu, Jan 23, 2014 at 12:52 AM, R. Mattes wrote: > > > >>> 1. Last time I checked, Novation, M-Audio, Roland etc. had no > >>> OSC keyboards, just regular MIDI ones :) > >> > >> It would at least be a way to overcome the lack of LV2/LADSPA > >> support > > > > Which, as pointed out earlier, isn't necessarily such a big deal. > > My point is, most of us haven't had a go at beta versions of Bitwig > > yet, hence there's no knowing, how good/bad the built-in plugins > > are. > > They are quite usable and complete as in "every basic thing is available". > > The synths are too simple to replace big LV2-plugins such as Calf > Organ but they sound pretty OK, EQ/Dynamics are solid and sound pretty > good, as far as I remember there is a usable reverb but not a > IR-convolver, this would be the only thing I'd really miss. > > best regards > > HZN > > > > > No LV2 support in v1.0 had been publicly known for a long, long > > time. And people were/are still ready to pay for the app. LADSPA? > > Haven't used those for ages. > > > > Alexandre _______________________________________________ > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.santamauro at gmail.com Thu Feb 6 10:08:31 2014 From: david.santamauro at gmail.com (David Santamauro) Date: Thu, 06 Feb 2014 05:08:31 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F35F1F.9020300@gmail.com> On 02/06/2014 05:02 AM, rosea grammostola wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. An creative advantage of modular linuxaudio could > be that you've a high level of creativity and uniqueness in your music. Can't the same be said about the 12 tones of a musical scale? You can think of rhythm and interval juxtaposition as knobs on the plugins. Yes, the fundamental blocks are finite but the method of building them into a cohesive whole are infinite. David From david.santamauro at gmail.com Thu Feb 6 10:08:47 2014 From: david.santamauro at gmail.com (David Santamauro) Date: Thu, 06 Feb 2014 05:08:47 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F35F2F.1020102@gmail.com> On 02/06/2014 05:02 AM, rosea grammostola wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. An creative advantage of modular linuxaudio could > be that you've a high level of creativity and uniqueness in your music. Can't the same be said about the 12 tones of a musical scale? You can think of rhythm and interval juxtaposition as knobs on the plugins. Yes, the fundamental blocks are finite but the method of building them into a cohesive whole are infinite. David From alexandre.prokoudine at gmail.com Thu Feb 6 10:12:49 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Thu, 6 Feb 2014 14:12:49 +0400 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: On Thu, Feb 6, 2014 at 2:02 PM, rosea grammostola wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced with > it sounds pretty similar to each other, everybody uses the same samples and > plugins. An creative advantage of modular linuxaudio could be that you've a > high level of creativity and uniqueness in your music. I'm trying to make sense of your question, but I'm miserably failing :) Are you saying that e.g. if I run ZynAdddSubFX as an LV2 plugin inside a DAW as opposed to running it as a standalone app, then all of a sudden I'm on a lower level of creativity? How much more creative do I get when I use Hydrogen for drum section instead of a MIDI track inside A3 with DrMr plugin attached to it? How does it affect uniqueness, exactly? Alexandre From fons at linuxaudio.org Thu Feb 6 10:17:28 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 6 Feb 2014 10:17:28 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: <20140206101728.GA13289@linuxaudio.org> On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: > One of the first rules of mixing you learn is to mix bass to the > center. There is no such rule. > Also when mixing for vinyl if bass is not centered it will make > the needle jump out of the groove. No, it won't. 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 atte at youmail.dk Thu Feb 6 10:38:41 2014 From: atte at youmail.dk (Atte) Date: Thu, 06 Feb 2014 11:38:41 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F36631.6000001@youmail.dk> On 02/06/2014 11:02 AM, rosea grammostola wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. They do? I don't... I truly believe that a tool is just a tool. It's the person handling the software that makes the decisions. Sure one decision could be "I wanna sound *exactly* like " or maybe the person doesn't have any (personal/unique) musical ideas in the first place. > An creative advantage of modular linuxaudio could > be that you've a high level of creativity and uniqueness in your music. And a creative disadvantage could be that your software might be unstable and limited in what you can do with it? -- Atte http://atte.dk http://modlys.dk From robin at gareus.org Thu Feb 6 11:07:46 2014 From: robin at gareus.org (Robin Gareus) Date: Thu, 06 Feb 2014 12:07:46 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: <52F36D02.9030608@gareus.org> On 02/05/2014 04:36 PM, Matt Garman wrote: > I'm not sure if "histogram" is the right word, but in my mind what > I'd like to see, per-channel, is something like this: > > 150--125 Hz: x samples 125--100 Hz: y samples 100--80 Hz: z > samples ... a low-pass or band-pass filter, followed by an RMS meter can do what you want. Though ideally you'll look at the stereo-phase correlation (after filtering). [..] > I am a programmer, and thought it would be easy to quickly hack > something up that would do this, but I have no experience with > signal processing, and as I started reading about this, I quickly > got in over my head! So I was hoping there might already exist a > tool that has this functionality. There are various GUI tools and Plugins for audio-analysis. But that's no fun for batch-analysis of > 10K audio-files. I don't think a commandline tool exists. You might be able to hack something together jalv.console (http://dev.drobilla.net/ticket/943) or vamp-simple-host or maybe ecasound. But it'll probably be easier to whip something up from scratch; some simplified pseudo-code: foreach audio-sample as inL, inR { /* 1st order low pass filter */ left += w * (inL - left); right += w * (inR - right); /* calc RMS */ rmsL = left * left; rmsR = right * right; if (rmsL > threshold_squared) ++L_above; if (rmsR > threshold_squared) ++R_above; } w ~ omega, is the filter constant. For -3dB at freq: w = 1.0 - e^(-2.0 * ? * freq / SampleRate); see also https://en.wikipedia.org/wiki/Low-pass_filter This should get you started at least, there's plenty of example code and literature around for more advanced filters. > The motivation for this is: I have a hardware DAC (digital audio > converter) in one part of my house, and a subwoofer in another. > There is a single coax run between the DAC and subwoofer, so I can > only send one channel. If you only have one subwoofer, just downmix to mono before sending the audio there. Cheers! robin From paul at linuxaudiosystems.com Thu Feb 6 12:39:00 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Thu, 6 Feb 2014 07:39:00 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: On Thu, Feb 6, 2014 at 5:02 AM, rosea grammostola < rosea.grammostola at gmail.com> wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. An creative advantage of modular linuxaudio could be > that you've a high level of creativity and uniqueness in your music. > that must explain the remarkable uniformity of music produced with proprietary tools such as ProTools and Logic. are you serious? -------------- next part -------------- An HTML attachment was scrubbed... URL: From linux at alextone.info Thu Feb 6 12:44:17 2014 From: linux at alextone.info (Alex) Date: Thu, 06 Feb 2014 13:44:17 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F383A1.7030109@alextone.info> On 02/06/2014 01:39 PM, Paul Davis wrote: > > > > On Thu, Feb 6, 2014 at 5:02 AM, rosea grammostola > > wrote: > > Isn't it a disadvantage of such all-in-one-daws that all music > produced with it sounds pretty similar to each other, everybody > uses the same samples and plugins. An creative advantage of > modular linuxaudio could be that you've a high level of creativity > and uniqueness in your music. > > > that must explain the remarkable uniformity of music produced with > proprietary tools such as ProTools and Logic. > > are you serious? > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Aaah, that "industry standard" sound........ From alexandre.prokoudine at gmail.com Thu Feb 6 12:53:54 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Thu, 6 Feb 2014 16:53:54 +0400 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: On Thu, Feb 6, 2014 at 4:39 PM, Paul Davis wrote: > that must explain the remarkable uniformity of music produced with > proprietary tools such as ProTools and Logic. This uniformity cannot possibly be true! Every DAW does channels summing differently in a unique, distinctive way! Everybody knows that :-P Alexandre From paul at linuxaudiosystems.com Thu Feb 6 13:00:57 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Thu, 6 Feb 2014 08:00:57 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F383A1.7030109@alextone.info> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <52F383A1.7030109@alextone.info> Message-ID: On Thu, Feb 6, 2014 at 7:44 AM, Alex wrote: > Aaah, that "industry standard" sound........ > yes, the same one that makes it impossible to differentiate a recording off the bach cello suites from the latest pop, or either of them from some chilled out scandanavian jazz. in all fairness, i think that the comment was targetting music "made" with "plugins" and "samples", but it still seems completely absurd given the amazing range of plugins available on non-linux platforms, and the amazing possibilities that any good sample library offers. -------------- next part -------------- An HTML attachment was scrubbed... URL: From lau at kudla.org Thu Feb 6 13:26:23 2014 From: lau at kudla.org (Rob) Date: Thu, 06 Feb 2014 08:26:23 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F38D7F.3010100@kudla.org> On 02/06/2014 05:02 AM, rosea grammostola wrote: > Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. An creative advantage of modular linuxaudio could be > that you've a high level of creativity and uniqueness in your music. My primary sequencer is LMMS, which is like the poster child for all-in-one composition tools. I may have used a few of their drum samples in the past, but everything else is from my own recordings or other sources. I agree that music created by different composers in a particular electronic music tool tends to sound the same, but I think that's because most people use electronic music tools to create electronic dance music, which, indeed, all sounds more or less the same for the most part by design, with a few dozen basic beats and common bass lines and the rest is just flair. Rob From unaudio at gmail.com Thu Feb 6 13:54:26 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Thu, 6 Feb 2014 15:54:26 +0200 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206101728.GA13289@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> Message-ID: On Thu, Feb 6, 2014 at 12:17 PM, Fons Adriaensen wrote: > On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: > >> One of the first rules of mixing you learn is to mix bass to the >> center. > > There is no such rule. > >> Also when mixing for vinyl if bass is not centered it will make >> the needle jump out of the groove. > > No, it won't. > > 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 http://www.resoundsound.com/mixing-for-vinyl-dont-fall-for-these-traps/ "Make the bass mono when mixing for vinyl. Always and absolutely. With bass I don't only mean the bassline. I mean all low frequencies - the bassline, the low end of your drums, percussion, any bassy effects, etc. No panning, no stereo effects. Make it mono. With stereo bass content the needle has to do big vertical movements which easily results in skips. Also the record will have to be cut quieter." I'm sure you know better, just saying that this is what everyone else is saying. From fons at linuxaudio.org Thu Feb 6 14:59:24 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 6 Feb 2014 14:59:24 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> Message-ID: <20140206145924.GB13289@linuxaudio.org> On Thu, Feb 06, 2014 at 03:54:26PM +0200, Vytautas Jancauskas wrote: > On Thu, Feb 6, 2014 at 12:17 PM, Fons Adriaensen wrote: > > On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: > > > >> One of the first rules of mixing you learn is to mix bass to the > >> center. > > > > There is no such rule. > > > >> Also when mixing for vinyl if bass is not centered it will make > >> the needle jump out of the groove. > > > > No, it won't. > http://www.resoundsound.com/mixing-for-vinyl-dont-fall-for-these-traps/ > > "Make the bass mono when mixing for vinyl. Always and absolutely. With > bass I don't only mean the bassline. I mean all low frequencies - the > bassline, the low end of your drums, percussion, any bassy effects, > etc. No panning, no stereo effects. Make it mono. A DJ talking about the type of music he works with, probably rather heavy on bass. > With stereo bass content the needle has to do big vertical movements > which easily results in skips. Also the record will have to be cut > quieter." That last sentence says it all. The only reason why bass is usually centered (both in the vinyl area and today) is that insane quest for 'loudness'. If the needle skips that means the record was cut badly, or the player has a problem, or both. Compliance of the needle + cartridge will be the same in horizontal and vertical directions, as should be the resonance frequencies. And a good player will have more damping vertically than horizontally. > I'm sure you know better, just saying that this is what everyone > else is saying. Ask 'everyone' if a sailboat can go faster than the wind. Nine out ten will say no. Which is wrong, it's being done all the time. 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 danstowell+lxau at gmail.com Thu Feb 6 16:15:48 2014 From: danstowell+lxau at gmail.com (Dan S) Date: Thu, 6 Feb 2014 16:15:48 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206145924.GB13289@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> Message-ID: 2014-02-06 Fons Adriaensen : > On Thu, Feb 06, 2014 at 03:54:26PM +0200, Vytautas Jancauskas wrote: >> On Thu, Feb 6, 2014 at 12:17 PM, Fons Adriaensen wrote: >> > On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: >> > >> >> One of the first rules of mixing you learn is to mix bass to the >> >> center. >> > >> > There is no such rule. >> > >> >> Also when mixing for vinyl if bass is not centered it will make >> >> the needle jump out of the groove. >> > >> > No, it won't. > >> http://www.resoundsound.com/mixing-for-vinyl-dont-fall-for-these-traps/ >> >> "Make the bass mono when mixing for vinyl. Always and absolutely. With >> bass I don't only mean the bassline. I mean all low frequencies - the >> bassline, the low end of your drums, percussion, any bassy effects, >> etc. No panning, no stereo effects. Make it mono. > > A DJ talking about the type of music he works with, probably rather > heavy on bass. Hi all, At the Audio Engineering Society conference last week in London, there was a great talk presenting an analysis of all the number one chart singles since the 1950s.* I wish the data was online because it demonstrates very clearly the point about bass and centre-panning: in the 1950s things were mono; in the 1960s there was plenty of experimentation with stereo; from the 1970s onwards, a very strong convention emerged in the production of these tracks, where the <100Hz part of the mix was extremely consistently centre-panned, while in the upper ranges things get panned around plenty. This pattern doesn't appear to have changed as we entered the CD/MP3 era. My point here is not that the "rule" Fons denies is not an unbreakable rule but it's an extremely strong convention, empirically demonstrated in this pop dataset at the least. So yes it's a "rule" in the colloquial sense, and not just in bass music. I have no idea if the jump-the-needle argument is plausible or urban myth - I had always thought it was motivated by perceptual considerations, not by the medium. Dan * Pestana & Reiss (2014). ftp://ftp.idmt.fraunhofer.de/aes53_abstracts.html#S2-2 From csanchezgs at gmail.com Thu Feb 6 16:27:15 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 6 Feb 2014 17:27:15 +0100 Subject: [LAU] [OT] RedPitaya, a HW laboratory in your pocket Message-ID: Don't know if anybody knows about this: http://www.redpitaya.com/ FWIW, Regards. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Thu Feb 6 16:34:23 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 6 Feb 2014 17:34:23 +0100 Subject: [LAU] [OT] Helium: The first supercapacitor-powered portable speaker Message-ID: Another FWIW, interesting and OS-HW crowdfunded project: https://www.crowdsupply.com/blueshift/helium At the end of the page there are another interesting projects including some related to FX. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Thu Feb 6 16:40:29 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 6 Feb 2014 17:40:29 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone Message-ID: Yet another FWIW, interesting and OS-HW project, https://phonebloks.com/en/goals Sorry, this is the last dear folks. I just thought you'd like. Regards -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From arnold at arnoldarts.de Thu Feb 6 17:05:50 2014 From: arnold at arnoldarts.de (Arnold Krille) Date: Thu, 6 Feb 2014 18:05:50 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: <52F148A1.5000006@autostatic.com> Message-ID: <20140206180550.67ddb26f@orinoco> Am Thu, 6 Feb 2014 09:29:52 +0100 schrieb Jos? Agust?n Terol Sanchis : > Thank you for your answers and suggestions. > > BTW, Jeremy, I've been taking a look at the Saffire 40 user manual, > and I have a doubt. When phantom power is turned on (for example, at > 5-8 inputs), does this affect to input lines in the same segment > (5-8)? I mean, it has a connector XLR/TRS combo, so, I wonder if the > +48V only affects the XLR plug, or it will also power the TRS > connection (that could damage some line-level stuff connected at > TRS). Well, I think this question is more or less applicable to those > interfaces with this kind of input combos. Applying phantom-power to the line-inputs would be against the specs. So, no, no sane vendor does that. And while the TRS and XLR share the socket, the contacts are different as the amplification-stages are different... - Arnold -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 230 bytes Desc: not available URL: From gheskett at wdtv.com Thu Feb 6 17:13:25 2014 From: gheskett at wdtv.com (Gene Heskett) Date: Thu, 6 Feb 2014 12:13:25 -0500 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> Message-ID: <201402061213.26048.gheskett@wdtv.com> On Thursday 06 February 2014 11:22:43 Vytautas Jancauskas did opine: > On Thu, Feb 6, 2014 at 12:17 PM, Fons Adriaensen wrote: > > On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: > >> One of the first rules of mixing you learn is to mix bass to the > >> center. > > > > There is no such rule. > > > >> Also when mixing for vinyl if bass is not centered it will make > >> the needle jump out of the groove. > > > > No, it won't. > > > > 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 > > http://www.resoundsound.com/mixing-for-vinyl-dont-fall-for-these-traps/ > > "Make the bass mono when mixing for vinyl. Always and absolutely. With > bass I don't only mean the bassline. I mean all low frequencies - the > bassline, the low end of your drums, percussion, any bassy effects, > etc. No panning, no stereo effects. Make it mono. > > With stereo bass content the needle has to do big vertical movements > which easily results in skips. Also the record will have to be cut > quieter." > > I'm sure you know better, just saying that this is what everyone else is > saying. This may be an artifact of the std vinyl playing gear at radio stations, used historically where the records were played live on the air by local dj's. The majority of the stations I'd had experience with usually were equipt with decent turntables and arms, but because it was indestructable, had the old Pickering/Stanton D500 cartridge fitted. Those were so stiff they ate records with brand new needles in them. I was the CE at a small county seat CA radio, am-fm op in NE CA for a couple years, went in just as Olivia-Newton John's "Physical" was climbing the charts. Figuring it would last several months near the top of the playlist, they ordered 10 copies because they were destroying those 45's from cue burn & skipping at about 1 a week. Checking the arms, I found them set for about 12 grams, and that the needles were a good 10,000 hours old, resembling a lathe tool under the scope. I could not see the needle move when I pulled a finger across it. I knew the Shure RE15 was a good cartridge, so I hied myself over to the local Radio Shack, who had just one (small town shack) under their own label for about $57. Cecil (owner) had a cow about it. I said listen to it. 4 days later I did the same, readjusting the arms for 1.5 grams, with his blessing to the other TT & arm. They were able to cut the number of records being burned up to zero. We had 8 copies of "Physical" that went to the morgue in the garage, never touched by a needle. Pet Clarks "Downtown" same story. Cue burns and skips simply never happened again. We had one big fellow who was doing the evening shift that had arms & hands that weren't used to handling an arm that weighed so little, he was around 6'6" & 325 lbs and he managed to destroy a needle by rotating it in the elastomer mount, I took some tweezers and twisted it back vertical. Still in service and looking good under the microscope 2 years later when I headed east. That $115 change had probably saved them 10 grand in records not bought in that 2 years. Radio stations, because of the expected high play counts, have to buy specially licensed records at 3 or 4 times what we pay at Wallies, and still have to keep play records for ACAP/BMI & pay those fees. So it can be a major ongoing expense. However with that cartridge in that old viscous damped Grey arm, could have been quite capable of playing one on Emery Cooks 78 rpm lp's of an earthquake, in real time or sped up 16x as some cuts on the record were. It also could repeatedly play the Mercury recording of the 1812 overture without damaging either the Harkness Tower bells or the cannon shots. Play that record once with a lessor cartridge and the bells turned into fingernails on a blackboard. Even if it was a mono recording, that record was and is a technological tour de force. Those cannon shots? Recorded by an Altec M21B mic, located about 6" below the muzzle of the cannon. That mic was capable of around 135 volts peak to peak output, admittedly with some 2nd harmonic distortion, but it did NOT clip. I do not think its equal has ever been made since. That cartridge tracked at 1.5 grams, .75 grams on the needle, and .75 grams on the carbon fiber brush that rode slightly inboard to clean the record before the needle got there. Does anyone sell something like that today, 35 years later? 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 Thu Feb 6 17:20:37 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 6 Feb 2014 17:20:37 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> Message-ID: <20140206172037.GD13289@linuxaudio.org> On Thu, Feb 06, 2014 at 04:15:48PM +0000, Dan S wrote: > My point here is not that the "rule" Fons denies is not an unbreakable > rule but it's an extremely strong convention, empirically demonstrated > in this pop dataset at the least. So yes it's a "rule" in the > colloquial sense, and not just in bass music. It is certainly a very strong convention. Apart from loudness there may be another reason why. In typical 'chart' music the bass is responsible for a large part of the signal level. It's easy to test this, take your favourite track and send it via a switchable highpass (200 Hz) to a meter that shows both RMS and peak (e.g. a K-meter). With the filter in place, the RMS level wil drop quite a bit (between 5 and 10 dB). The peak level will change much less. That means that when you put the bass off-center, your level meters may well show 'out of balance', in particular if they have a small range as e.g. a VU, during the entire song. Many sound engineers probably dislike that, even if it does no harm at all. 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 gordonjcp at gjcp.net Thu Feb 6 18:12:13 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Thu, 6 Feb 2014 18:12:13 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> Message-ID: <20140206181213.GA8541@gjcp.net> On Thu, Feb 06, 2014 at 04:15:48PM +0000, Dan S wrote: > > My point here is not that the "rule" Fons denies is not an unbreakable > rule but it's an extremely strong convention, empirically demonstrated > in this pop dataset at the least. So yes it's a "rule" in the > colloquial sense, and not just in bass music. I have no idea if the > jump-the-needle argument is plausible or urban myth - I had always > thought it was motivated by perceptual considerations, not by the > medium. I always thought it was because a) low frequencies are hard to localise, so panning them wildly left and right won't really do much and b) most stereos are fairly light on bass so panning low frequencies to the centre ensures that both channels are drive fairly equally giving the greatest sound pressure level possible. I suspect having the bassy sounds far off-centre would be uncomfortable to listen to, because we're so used to having bass and vocals panned to the middle. Experiment, but consider that we do things a conventional way because it works - imagine reading a book where paragraphs alternated between left to right and right to left. -- Gordonjcp MM0YEQ From nettings at stackingdwarves.net Thu Feb 6 18:46:58 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Thu, 06 Feb 2014 19:46:58 +0100 Subject: [LAU] Focusrite Saffire PRO 40 at 96kHz In-Reply-To: References: <52F148A1.5000006@autostatic.com> Message-ID: <52F3D8A2.7090701@stackingdwarves.net> On 02/06/2014 09:29 AM, Jos? Agust?n Terol Sanchis wrote: > Thank you for your answers and suggestions. > > BTW, Jeremy, I've been taking a look at the Saffire 40 user manual, and > I have a doubt. When phantom power is turned on (for example, at 5-8 > inputs), does this affect to input lines in the same segment (5-8)? I > mean, it has a connector XLR/TRS combo, so, I wonder if the +48V only > affects the XLR plug, or it will also power the TRS connection (that > could damage some line-level stuff connected at TRS). Well, I think this > question is more or less applicable to those interfaces with this kind > of input combos. even the simplest neutrik jack/xlr combo socket is actually a 6-pin connector, with separate pins for xlr 1-3, tip, ring, and a common sleeve/xlr chassis ground. so unless the manufacturer has been designing while drunk, the p48 should not affect the line inpute. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From jh at brainiac.com Thu Feb 6 19:02:01 2014 From: jh at brainiac.com (Joe Hartley) Date: Thu, 6 Feb 2014 14:02:01 -0500 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206181213.GA8541@gjcp.net> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> Message-ID: <20140206140201.3b8ffe00eb7a04bd03bea3b1@brainiac.com> On Thu, 6 Feb 2014 18:12:13 +0000 Gordon JC Pearce wrote: > I always thought it was because a) low frequencies are hard to localise, > so panning them wildly left and right won't really do much and b) most > stereos are fairly light on bass so panning low frequencies to the centre > ensures that both channels are drive fairly equally giving the greatest > sound pressure level possible. A good summary. > I suspect having the bassy sounds far off-centre would be uncomfortable > to listen to, because we're so used to having bass and vocals panned to > the middle. It's not uncomfortable at all, though it can be odd. I'm currently listening to a stereo mix of Rubber Soul on a pair of near-field monitors and it's pretty drastic. Switching over to a stereo in the living room the separation's noticable but doesn't feel nearly as drastic. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From gnome at hawaii.rr.com Thu Feb 6 19:16:34 2014 From: gnome at hawaii.rr.com (david) Date: Thu, 06 Feb 2014 09:16:34 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F3DF92.2070701@hawaii.rr.com> On 02/06/2014 02:39 AM, Paul Davis wrote: > On Thu, Feb 6, 2014 at 5:02 AM, rosea grammostola wrote: > > Isn't it a disadvantage of such all-in-one-daws that all music > produced with it sounds pretty similar to each other, everybody uses > the same samples and plugins. An creative advantage of modular > linuxaudio could be that you've a high level of creativity and > uniqueness in your music. > > that must explain the remarkable uniformity of music produced with > proprietary tools such as ProTools and Logic. > > are you serious? Or "professional" has come to mean "it sounds like this"? Like a "professional" singer sounds like Madonna or a "professional" guitarist sounds like Eddie Van Halen? I think it's independent of the DAW they use. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From ralf.mardorf at rocketmail.com Thu Feb 6 20:13:28 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Thu, 06 Feb 2014 21:13:28 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206172037.GD13289@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> Message-ID: <1391717608.690.16.camel@archlinux> On Wed, 2014-02-05 at 22:02 +0000, James Morris wrote: > On Wed, 5 Feb 2014 11:28:15 -0500 > Joe Hartley wrote: > > > > That said, if you're listening to old Beatles stereo mixes, the bass > > will only be on one channel (left, usually) and if you've sent the > > right to the DAC, there's not much for the sub. I'd try and convert > > the signal to mono for the sub feed. > > > > This suprised me as I had heard that for vinyl it is particularly > important to have mono bass to prevent the needle jumping :-) > > ie > http://www.customrecords.com/prepare_music_for_vinyl_record.html And if you gouge holes in a CD no data gets lost ;). JFTR a Beatles bass does sound natural, however, even a "Bring the noise"-"Bass! How low can you go?"-unnatural-bass mixed to one channel only wouldn't make the needle jump, the needle will jump if you didn't adjust the counterweight correctly and this btw. isn't the only thing you can adjust for a good record player. I'm not speaking about audiophile record players, I'm talking about good record players, e.g. the well known DJ model, but even some HiFi players are very good. No CD is able to hold a candle to a record played on a good record player. On Thu, 2014-02-06 at 10:41 +0200, Vytautas Jancauskas wrote: > One of the first rules of mixing you learn is to mix bass to the > center. Where did you learn and for what studios did you work? This is complete nonsense. On Thu, 2014-02-06 at 10:17 +0000, Fons Adriaensen wrote: > On Thu, Feb 06, 2014 at 10:41:33AM +0200, Vytautas Jancauskas wrote: > > > One of the first rules of mixing you learn is to mix bass to the > > center. > > There is no such rule. +1 > > Also when mixing for vinyl if bass is not centered it will make > > the needle jump out of the groove. > > No, it won't. +1 On Thu, 2014-02-06 at 15:54 +0200, Vytautas Jancauskas wrote: > just saying that this is what everyone else is saying. I don't know anybody who claims this bass nonsense. From ralf.mardorf at rocketmail.com Thu Feb 6 20:32:39 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Thu, 06 Feb 2014 21:32:39 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F3DF92.2070701@hawaii.rr.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <52F3DF92.2070701@hawaii.rr.com> Message-ID: <1391718759.690.21.camel@archlinux> On Thu, 2014-02-06 at 09:16 -1000, david wrote: > On 02/06/2014 02:39 AM, Paul Davis wrote: > > > On Thu, Feb 6, 2014 at 5:02 AM, rosea grammostola wrote: > > > > Isn't it a disadvantage of such all-in-one-daws that all music > > produced with it sounds pretty similar to each other, everybody uses > > the same samples and plugins. An creative advantage of modular > > linuxaudio could be that you've a high level of creativity and > > uniqueness in your music. > > > > that must explain the remarkable uniformity of music produced with > > proprietary tools such as ProTools and Logic. > > > > are you serious? > > Or "professional" has come to mean "it sounds like this"? Like a > "professional" singer sounds like Madonna or a "professional" guitarist > sounds like Eddie Van Halen? > > I think it's independent of the DAW they use. That much in the charts does sound equal is caused by such nonsense: On Thu, 2014-02-06 at 10:41 +0200, Vytautas Jancauskas wrote: > One of the first rules of mixing you learn is to mix bass to the > center. If DJs don't hire good audio engineers, but mix their crap on their own. The used recording gear, DAW or analog gear has less to do with it, just the fashion what real or virtual instruments are used have got influence, but there are for sure more proprietary virtual instruments available, than Linux instruments. From alf at mellomrommet.no Thu Feb 6 21:21:27 2014 From: alf at mellomrommet.no (Alf Haakon Lund) Date: Thu, 06 Feb 2014 22:21:27 +0100 Subject: [LAU] [OT] Helium: The first supercapacitor-powered portable speaker In-Reply-To: References: Message-ID: <52F3FCD7.2030609@mellomrommet.no> On 06. feb. 2014 17:34, Carlos sanchiavedraz wrote: > Another FWIW, interesting and OS-HW crowdfunded project: > https://www.crowdsupply.com/blueshift/helium > > At the end of the page there are another interesting projects > including some related to FX. > Thx for interesting links! Alf From tim at quitte.de Thu Feb 6 22:22:29 2014 From: tim at quitte.de (Tim Goetze) Date: Thu, 6 Feb 2014 23:22:29 +0100 (CET) Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206172037.GD13289@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> Message-ID: [Fons Adriaensen] >That means that when you put the bass off-center, your level >meters may well show 'out of balance', in particular if they >have a small range as e.g. a VU, during the entire song. >Many sound engineers probably dislike that, even if it does >no harm at all. They might dislike it because it implies the energy budget of the transmission isn't exploited maximally. :) Cheers, Tim From fons at linuxaudio.org Thu Feb 6 22:34:12 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 6 Feb 2014 22:34:12 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> Message-ID: <20140206223412.GB29528@linuxaudio.org> On Thu, Feb 06, 2014 at 11:22:29PM +0100, Tim Goetze wrote: > [Fons Adriaensen] > >That means that when you put the bass off-center, your level > >meters may well show 'out of balance', in particular if they > >have a small range as e.g. a VU, during the entire song. > >Many sound engineers probably dislike that, even if it does > >no harm at all. > > They might dislike it because it implies the energy budget of the > transmission isn't exploited maximally. :) Which leads to the question why they would care about the energy budget if the information budget is used in such a suboptimal way :-) 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 Thu Feb 6 23:09:51 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 6 Feb 2014 23:09:51 +0000 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: References: Message-ID: <20140206230951.GC29528@linuxaudio.org> On Thu, Feb 06, 2014 at 05:40:29PM +0100, Carlos sanchiavedraz wrote: > Yet another FWIW, interesting and OS-HW project, > https://phonebloks.com/en/goals Three very interesting projects ! Thanks ! -- 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 dragonseel at gmail.com Thu Feb 6 23:39:00 2014 From: dragonseel at gmail.com (Dragonseel) Date: Thu, 6 Feb 2014 15:39:00 -0800 (PST) Subject: [LAU] Steinberg UR22 on ArchLinux Message-ID: <1391729940186-89280.post@n7.nabble.com> Hello, I am owner of a Steinberg UR22 and user of ArchLinux with Kernel Version /3.12.9-2-ARCH/. I use ALSA version 1.0.27-2 Now my Linux does not recognize the UR22 as a sound device. It is recognized as a USB-Device. This is a similar situation to a older Thread. In that thread a solution was found by adding a section to a file named *quirks-table.h* but unfortunately there is no such file in ArchLinux, and apparently it is supposed do be that way. Is there an alternative way to do that fix? Or is there something else I miss? Thank you and greetings, Lukas -- View this message in context: http://linux-audio.4202.n7.nabble.com/Steinberg-UR22-on-ArchLinux-tp89280.html Sent from the linux-audio-user mailing list archive at Nabble.com. From ralf.mardorf at rocketmail.com Thu Feb 6 23:57:09 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 00:57:09 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206223412.GB29528@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <20140206223412.GB29528@linuxaudio.org> Message-ID: <1391731029.673.37.camel@archlinux> On Thu, 2014-02-06 at 22:34 +0000, Fons Adriaensen wrote: > On Thu, Feb 06, 2014 at 11:22:29PM +0100, Tim Goetze wrote: > > [Fons Adriaensen] > > >That means that when you put the bass off-center, your level > > >meters may well show 'out of balance', in particular if they > > >have a small range as e.g. a VU, during the entire song. > > >Many sound engineers probably dislike that, even if it does > > >no harm at all. > > > > They might dislike it because it implies the energy budget of the > > transmission isn't exploited maximally. :) > > Which leads to the question why they would care about the energy > budget if the information budget is used in such a suboptimal way > :-) Unfortunately people don't listen, they seem to watch music by meters and measuring instrument. There also is an issue with the gear, people care about technical specifications such as S/N ratio, klirr factor, linear frequency response. Those specifications are important for audio production, but usually a biased sound is wanted for listening to a record, CD, tape. Nobody is able to notice a less good S/N ratio or klirr factor when just listening, when not producing. Instead of using a nice biased record player, they prefer to listen to a CD player with better technical specification, but then they add insane equalizing and cheap reverbs to the output of those clean, cold CD players and kill the sound completely or they completely don't care and consume music where ever they are by earplugs or mobile phone speakers. Most of the times people seems to listen to music, just to have some background noise. Less people seem to listen to music in a way they would read a book. The gauge for the quality of the compositions nowadays seems to be tits and shaking asses. "How is the bass mixed? Per-channel frequency analysis? Histogram?" Who cares? How does it sound? Doesn't the faculty of hearing and individuell liking count anymore? I often get the impression that people do not want to learn how to listen and mix music anymore, but they want to have a norm and measuring instruments that does show when the mix does fit to the norm. Music is related to individuell liking and creativity, mixing is related to individuell liking and creativity. Sure, imitating a style is ok and good for learning. Is there an analysis tool, a meter to measure if the composition, coloration and the stroke of the brush of a painting does fit to a painting norm? There only is craftsmanship, e.g. different theories of colours, e.g. different positioning of the microphone techniques etc., but there is no limitation, it's good to know much about it, but also ok to break with it. Regarding to audio, measurement engineering is needed to maintain the equipment, but not to do a mix. From fons at linuxaudio.org Fri Feb 7 00:31:01 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 00:31:01 +0000 Subject: [LAU] Steinberg UR22 on ArchLinux In-Reply-To: <1391729940186-89280.post@n7.nabble.com> References: <1391729940186-89280.post@n7.nabble.com> Message-ID: <20140207003101.GD29528@linuxaudio.org> On Thu, Feb 06, 2014 at 03:39:00PM -0800, Dragonseel wrote: > Now my Linux does not recognize the UR22 as a sound device. > It is recognized as a USB-Device. This is a similar situation to a older > Thread. In that thread a solution was found by adding a section to a file > named *quirks-table.h* but unfortunately there is no such file in ArchLinux, > and apparently it is supposed do be that way. Is there an alternative way to > do that fix? Or is there something else I miss? There is no such file in a normal Archlinux installation because this is a *source* file, part of the ALSA drivers. You can * Get the ALSA sources and patch, compile and install them. See on how to do this cleanly (without bypassing package management). I wouldn't recommend this unless you are a familiar with creating packages from source. Or * Make sure the issue is reported to the ALSA developers, and wait for the next release which Archlinux will adopt quite fast. 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 cbannister at slingshot.co.nz Fri Feb 7 01:06:33 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Fri, 7 Feb 2014 14:06:33 +1300 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <52F36D02.9030608@gareus.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <52F36D02.9030608@gareus.org> Message-ID: <20140207010633.GA19394@tal> On Thu, Feb 06, 2014 at 12:07:46PM +0100, Robin Gareus wrote: > no fun for batch-analysis of > 10K audio-files. > I don't think a commandline tool exists. You might be able to hack > something together jalv.console (http://dev.drobilla.net/ticket/943) > or vamp-simple-host or maybe ecasound. There is libaudio-ecasound-perl http://search.cpan.org/dist/Audio-Ecasound/ I've never used it, but does reading this: (http://search.cpan.org/src/BOWMANBS/Audio-Ecasound-1.01/README) help? -- "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 rtg at aapsc.com Fri Feb 7 04:56:00 2014 From: rtg at aapsc.com (Rick Green) Date: Thu, 6 Feb 2014 23:56:00 -0500 (EST) Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140206181213.GA8541@gjcp.net> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> Message-ID: On Thu, 6 Feb 2014, Gordon JC Pearce wrote: > I always thought it was because a) low frequencies are hard to localise, > so panning them wildly left and right won't really do much... Many home sound systems today simply sum the bass energy from both channels into a single subwoofer anyway. It doesn't matter where in the room the subwoofer is located, because our ears are too close together to detect a phase difference at those frequencies. I do live sound in a venue with an installed sound system. We have two large subwoofers, fed in phase from a single source. The problem is that the two boxes are hung from the ceiling 30 feet apart. Two point sources that widely spaced give us interference fringes all across the room. You can walk slowly across the room and hear the bass come in and disappear with every step. I simply turn off the amp for one of the subs, and no one is the wiser, and everyone is happier. So defy convention if you like, and mix the bass wherever you like, but please forgive those of us with real-world transducers, where more often than not it's summed to mono before presentation anyway! I had heard the story about phonograph needles jumping out of the track. It made sense to me. On a stereo phonograph record, in-phase energy (L+R) is represented as side-to-side motion of the needle, and out-of-phase energy(L-R) is up and down motion. It can also be thought of as the left channel is recorded on one side of the groove(at a 45deg angle), and the right channel on the other side. It seemed plausible that a kick drum panned all the way to one side or the other might send the needle into the adjacent groove. -- Rick Green We, the People of the United States of America, reject the U.S. Supreme Court's Citizens United ruling, and move to amend our Constitution to firmly establish that money is not speech, and that human beings, not corporations, are persons entitled to constitutional rights. http://www.MoveToAmend.org From gnome at hawaii.rr.com Fri Feb 7 05:46:11 2014 From: gnome at hawaii.rr.com (david) Date: Thu, 06 Feb 2014 19:46:11 -1000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> Message-ID: <52F47323.3060301@hawaii.rr.com> On 02/06/2014 06:56 PM, Rick Green wrote: > On Thu, 6 Feb 2014, Gordon JC Pearce wrote: > >> I always thought it was because a) low frequencies are hard to >> localise, so panning them wildly left and right won't really do much... > > Many home sound systems today simply sum the bass energy from both > channels into a single subwoofer anyway. It doesn't matter where in the > room the subwoofer is located, because our ears are too close together > to detect a phase difference at those frequencies. I think our home stereo is up so that the amplifier's speaker output goes to the subwoofer, then the subwoofer splits the output to each stereo speaker while filtering the highs out of the subwoofer signal and the lows out of the speaker outputs. Our receiver doesn't have a separate subwoofer output like newer ones do, I guess. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From unaudio at gmail.com Fri Feb 7 07:17:51 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Fri, 7 Feb 2014 09:17:51 +0200 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <1391717608.690.16.camel@archlinux> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <1391717608.690.16.camel@archlinux> Message-ID: > And if you gouge holes in a CD no data gets lost ;). JFTR a Beatles bass > does sound natural, however, even a "Bring the noise"-"Bass! How low can > you go?"-unnatural-bass mixed to one channel only wouldn't make the > needle jump, the needle will jump if you didn't adjust the counterweight > correctly and this btw. isn't the only thing you can adjust for a good > record player. I'm not speaking about audiophile record players, I'm > talking about good record players, e.g. the well known DJ model, but > even some HiFi players are very good. No CD is able to hold a candle to > a record played on a good record player. I'm not going to pretend I understood anything of this. > I don't know anybody who claims this bass nonsense. I gave you a link, but if you look around there are plenty of sources that claim this, for example here is what soundonsound says: "There are few 'rules' in music production, but panning bass isn't far off. It is usually a good idea to pan the bass and kick to the centre. Partly this is historical (the limitations of vinyl) but, more importantly, it shares the bass energy equally between the two stereo speakers. It is also important because the listener will not always be in the sweet spot, and given that the bass is so critical to the mix, you want them to hear it wherever they are in relation to the speakers (this applies to dance music as much as any other -- you want all the clubbers to feel the same bass groove). " http://www.soundonsound.com/sos/apr07/articles/betterbass.htm From louigi.verona at gmail.com Fri Feb 7 08:04:39 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Fri, 7 Feb 2014 12:04:39 +0400 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: "Isn't it a disadvantage of such all-in-one-daws that all music produced with it sounds pretty similar to each other, everybody uses the same samples and plugins. An creative advantage of modular linuxaudio could be that you've a high level of creativity and uniqueness in your music." I do not agree with the premise that because everybody uses the same plugins it all sounds similar. As for samples, all-in-one DAW does not force you to use any samples, rarely does a DAW come with decent samples anyway. And same plugins can be used in a very different way. Also, it is funny you say that. Doesn't every person here use same stuff - a couple of main synths and a limited set of plugins? -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From simonzwise at gmail.com Fri Feb 7 08:16:36 2014 From: simonzwise at gmail.com (Simon Wise) Date: Fri, 07 Feb 2014 19:16:36 +1100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F49664.1040501@gmail.com> On 07/02/14 19:04, Louigi Verona wrote: > "Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same > samples and plugins. An creative advantage of modular linuxaudio could be > that you've a high level of creativity and uniqueness in your music." > > I do not agree with the premise that because everybody uses the same > plugins it all sounds similar. As for samples, all-in-one DAW does not > force you to use any samples, rarely does a DAW come with decent samples > anyway. And same plugins can be used in a very different way. > > Also, it is funny you say that. Doesn't every person here use same stuff - > a couple of main synths and a limited set of plugins? or maybe some record a cello ... again, very different sounds indeed using the same 'limited' configuration of wood, strings, bow and mic ... depending very much on the musician using it. Simon From bouncingcats at gmail.com Fri Feb 7 08:53:10 2014 From: bouncingcats at gmail.com (David) Date: Fri, 7 Feb 2014 19:53:10 +1100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <1391717608.690.16.camel@archlinux> Message-ID: On 7 February 2014 18:17, Vytautas Jancauskas wrote: [snip] > you want all the clubbers to feel the same bass groove). " For anyone who never had the opportunity of some school physics experiments: If there is more than one speaker stack in the room reproducing the same signal, it is impossible, due to the physical phenomenon of interference occurring whenever the distance from the listener to each speaker is not identical (ie not on the centre line). See http://en.wikipedia.org/wiki/Interference_%28wave_propagation%29 and specifically the http://en.wikipedia.org/wiki/File:Two_sources_interference.gif there. Also here http://en.wikipedia.org/wiki/Ripple_tank#Interference And even if there is only one speaker stack, reflections from the walls and ceilings will interfere. Walking around the room and listening usually confirms this, and is always worth doing. The only way to avoid interference is to have one speaker in free space, or to have one speaker and 100% absorbent walls. See http://en.wikipedia.org/wiki/Anechoic_chamber This is why rooms often sound better when they are full of people. From rmouneyres at gmail.com Fri Feb 7 09:22:53 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Fri, 7 Feb 2014 10:22:53 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <1391731029.673.37.camel@archlinux> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <20140206223412.GB29528@linuxaudio.org> <1391731029.673.37.camel@archlinux> Message-ID: >Most of the times > people seems to listen to music, just to have some background noise. > Less people seem to listen to music in a way they would read a book. I totally agree with that. >The gauge for the quality of the compositions nowadays seems to be tits and > shaking asses. Well, i agree on this, but only for the mass music production heard on the media mainly. There are a lots of great bands with creative music. you will probably never hear about until you've found them, and obviously if you are looking for them. From zettberlin at linuxuse.de Fri Feb 7 09:43:51 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Fri, 07 Feb 2014 10:43:51 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <52F4AAD7.1060001@linuxuse.de> Am 06.02.2014 11:02, schrieb rosea grammostola: > Isn't it a disadvantage of such all-in-one-daws that all music > produced with it sounds pretty similar to each other, everybody > uses the same samples and plugins. An creative advantage of modular > linuxaudio could be that you've a high level of creativity and > uniqueness in your music. Tools do have an influence but it is up to the musician, to evaluate and master that influence in relation to his/her musical vision. And of course: if your musical vision is music with complex polyrythmic structure then a DAW that is optimized to do the olde 4onthefloor will be harder to use to get there. Anyway, I think you refer to the tendency of some hobbyists to actually *follow* lines that software draws by optimization for certain styles, sounds, structures etc. Some people love it, to use templates and resets and since some of those presets and templates are actually excellent, many releases sound similar. This is stupid but not the fault of the software. It is entirely up to those musicians. The only exception would be software, that is so primitive, that it excluselively allows to switch some presets. Regarding Bitwig and its built-ins, that is not the case. They ship some presets but you can alter them to make the plugins sound the same as freaked out as any obscure DIY-hardware or PD-patch would offer. When I talked to the devs at Bitwig I had the very strong impression, that one of their goals is to provide a system with infinite possibilities. Especially the concept for automation is so advanced, that it seems quite hard to handle for my tastes, when you really use all opportunities it has to offer. And the plugin-section is built for modularity, this does not work to the full yet but once it has matured it is actually a complete modular synth in the style of AMS. best regards HZN > > > On Thu, Feb 6, 2014 at 12:46 AM, Hartmut Noack > wrote: > >> Sorry for getting into this so late.... Am 22.01.2014 22:14, >> schrieb Alexandre Prokoudine: >>> On Thu, Jan 23, 2014 at 12:52 AM, R. Mattes wrote: >>> >>>>> 1. Last time I checked, Novation, M-Audio, Roland etc. had >>>>> no OSC keyboards, just regular MIDI ones :) >>>> >>>> It would at least be a way to overcome the lack of >>>> LV2/LADSPA support >>> >>> Which, as pointed out earlier, isn't necessarily such a big >>> deal. My point is, most of us haven't had a go at beta versions >>> of Bitwig yet, hence there's no knowing, how good/bad the >>> built-in plugins are. >> >> They are quite usable and complete as in "every basic thing is >> available". >> >> The synths are too simple to replace big LV2-plugins such as >> Calf Organ but they sound pretty OK, EQ/Dynamics are solid and >> sound pretty good, as far as I remember there is a usable reverb >> but not a IR-convolver, this would be the only thing I'd really >> miss. >> >> best regards >> >> HZN >> >>> >>> No LV2 support in v1.0 had been publicly known for a long, >>> long time. And people were/are still ready to pay for the app. >>> LADSPA? Haven't used those for ages. >>> >>> Alexandre _______________________________________________ >>> 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 linux at alextone.info Fri Feb 7 10:09:23 2014 From: linux at alextone.info (Alex) Date: Fri, 07 Feb 2014 11:09:23 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F49664.1040501@gmail.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <52F49664.1040501@gmail.com> Message-ID: <52F4B0D3.2020506@alextone.info> The idea that daws sound the "same" has been refuted fairly well by previous commenters. I'll add that using sample libs is arguably the most likely to sound similar if users record with them in their natural state, and not work at developing a unique sound with the same base material. Example. My main working libs set is the Sonic Implants Orchestral collection (I have others, but this is the meat and bones i work with daily), which i've owned since it was first released, in gig format. (And linuxsampler is my closest friend and ally in the linux audio world for playing them) In their "factory" state, the orchestral instruments and sections are recorded in situ, meaning they're recorded with a built in stage presence. At first glance, it would seem that every muso using the SI collection is going to get the same end result, i.e. the oboe always sounds like this, the flute has a slightly sharp top C, etc. Sounds ok, until the muso realizes that his oboe masterpiece will have exactly the same sonic signature as the next muso, unless he does something to change that. This goes for notes as well, where a detached G4 oboe note for example, will always start and stop exactly the same. (in general) And this is where the notion of OOTB falls flat. The sample libs are an instrument, in effect, and just like a live player or players, they need to be learnt as an instrument, with many hours practising not just playing technique (orchestration should be a prerequisite for this), but manipulation skills, where tools and plugins like EQ, and IR space, give the user a chance to uniquely stamp his style on the standard set. Manipulating the base set is the real skill set required, to have your own signature, and is essential to avoid the "same" sound as many others who don't perceive these skills are important. As a further quick example, i have 2 flutes, one of a different tone than the other. I will mix them in an orchestral project, alternately and together, but i will also, where it is merited, and usually when the flute line is exposed in some way, add a little mid and take a little off the top in EQ to differentiate the base samples from the rest of the project. So to imprinted presence. Earlier sample libs manufacturers feted this as a feature, but it wasn't too lonf after that that users assumed they were trapped by this, and manufacturers started taking a different approach, where IR samples were added separately to sample sets, with a text or data file loaded into popular samplers, enabling users to more directly control the direction and strength of ER, and tutti IR. Nevertheless, imprinted samples can be manipulated to quite some degree to minimise or change the imprinted IR, albeit with quite a bit of effort, often by panning and doubling sections to different stage locations with an addition IR, narrowing both fields to minimise the imprinted effect. This sounds totally at odds with assumed perspectives of "traditional" IR, ER and EQ, but hey, if it works... As a former orchestral player, and with a formal music education, the knowledge i gained was invaluable in writing score, putting together orchestral projects, etc. But when using "factory" sample libs on a computer, a completely new set of skills was required to project that unique sonic image. I know most of you are electronica based, and by preference, but imho, the same principles apply. ZynaddsubFX as an example, comes with preset banks, and if one is experienced in using this fine app, you can almost figure out which preset the user has applied, if "nothing" is done to the preset, the same as factory sample libs. (The same is true for sample libs. "He's using SI, or VSL, or Project Sam", etc) It's the users job to be the muso, and consider the preset, or sample lib as the START point, from which lots can be done to develop one's own sound. It's not the end result, unless the user chooses that, by any stretch of the imagination, and in the hands of users who are willing to go on from the factory standard, it's possible to depart almost completely from the notion of this is AMSynth, this is how it "sounds...." I would hope that this short note provides another view for users, and possibly devs, with the mindset of a synth or sample set being the "finished" product. This may seem like stating the obvious, but i've heard many pieces over quite a few years from home users and professionals alike, where they write something assuming the result to be the finished article based on factory settings, and then wonder why it sounds the same as so many others, using the same tools, without a perceivable difference in sonic style. As a final comment, in the process of writing this, i respectfully suggest devs building plugins, VIs, and hosts consider this as well. The more tools you add (and make the process of using them as quick and efficient as possible, the easier the process of integrating manipulative tools like midi integration with controls you've built (for just one example), and the more hosts provide the mechanisms and framework that enables the users to depart even further from "factory" beginnings, the greater chance the user has to enjoy a much greater creative opportunity to express him or her self, in creating his or her unique sonic signature. Alex. From ralf.mardorf at rocketmail.com Fri Feb 7 12:45:49 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 13:45:49 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <1391717608.690.16.camel@archlinux> Message-ID: <1391777149.667.13.camel@archlinux> On Fri, 2014-02-07 at 09:17 +0200, Vytautas Jancauskas wrote: > given that the bass is so critical to the mix, > you want them to hear it wherever they are in relation to the speakers > (this applies to dance music as much as any other -- you want all the > clubbers to feel the same bass groove This is written by somebody who has got absolutely no knowledge about mixing. Perhaps this person knows how to mix dance/club music, but the claim "this applies [...] as much as any other" is nonsense. I read much about sub woofers on this list. I hope people are aware that a bass has got some high frequencies too, so a bass can be located and JFTR it's also possible to listen to music by headphones. Some people like to listen to music that is drawing a room and that has got a wide dynamic range. Relation to the speakers does matter for stereo, so do they completely mix in mono for dance/club music? No, they are mixing stereo, it's just stupid music made by people who aren't musicians and the main instrument are a bass line and a kick in unison to the bass line. If such an dance/club audio engineer isn't aware hat there are thousands of different music styles and that some music styles are really composed and/or improvised and not just a bass line + a synced kick and that even for many kinds of dance music, the rhythm group often is very subdued in the background, then such a wrong claim still is a wrong claim. From fons at linuxaudio.org Fri Feb 7 12:46:39 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 12:46:39 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <1391717608.690.16.camel@archlinux> Message-ID: <20140207124639.GA19601@linuxaudio.org> On Fri, Feb 07, 2014 at 09:17:51AM +0200, Vytautas Jancauskas wrote: > "There are few 'rules' in music production, but panning bass isn't far > off. It is usually a good idea to pan the bass and kick to the centre. > Partly this is historical (the limitations of vinyl) but, more > importantly, it shares the bass energy equally between the two stereo > speakers. This is a valid argument, in case getting the maximum energy matters. > It is also important because the listener will not always be > in the sweet spot, and given that the bass is so critical to the mix, > you want them to hear it wherever they are in relation to the speakers > (this applies to dance music as much as any other -- you want all the > clubbers to feel the same bass groove). " If this matters it means you are not using the two speakers as a stereo system, but just to distribute the sound evenly over a large space. In that case the whole concept of panning (i.e. creating a wide sound stage) becomes invalid and *everything*, not just the bass, should be mono. 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 Fri Feb 7 12:47:18 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 13:47:18 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <1391777238.667.14.camel@archlinux> On Fri, 2014-02-07 at 12:04 +0400, Louigi Verona wrote: > Also, it is funny you say that. Doesn't every person here use same > stuff - a couple of main synths and a limited set of plugins? No! I play guitars and own 19" effects and hardware synth. From ralf.mardorf at rocketmail.com Fri Feb 7 12:56:26 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 13:56:26 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <20140206223412.GB29528@linuxaudio.org> <1391731029.673.37.camel@archlinux> Message-ID: <1391777786.667.20.camel@archlinux> On Fri, 2014-02-07 at 10:22 +0100, Rapha?l Mouneyres wrote: > >Most of the times > > people seems to listen to music, just to have some background noise. > > Less people seem to listen to music in a way they would read a book. > > I totally agree with that. > > >The gauge for the quality of the compositions nowadays seems to be tits and > > shaking asses. > > Well, i agree on this, but only for the mass music production heard on > the media mainly. > There are a lots of great bands with creative music. you will probably > never hear about until you've found them, and obviously if you are > looking for them. Even dance floor music from the charts sometimes isn't rapped but sung and doesn't show naked women for the videos. Randomly I heard a song and saw the video of a song that seems to be on top of the charts, I suspect the title is "Happy", at least the refrain is "Happy". Indeed, the title is "Happy" https://en.wikipedia.org/wiki/Happy_(Pharrell_Williams_song) I wouldn't buy this record and I don't want to listen to it, since it's loudness mixed to death, but it at least isn't such crap as most of the other popular music is. From fons at linuxaudio.org Fri Feb 7 13:04:05 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 13:04:05 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> Message-ID: <20140207130405.GB19601@linuxaudio.org> On Thu, Feb 06, 2014 at 11:56:00PM -0500, Rick Green wrote: > I had heard the story about phonograph needles jumping out of the > track. It made sense to me. On a stereo phonograph record, in-phase > energy (L+R) is represented as side-to-side motion of the needle, > and out-of-phase energy(L-R) is up and down motion. It can also be > thought of as the left channel is recorded on one side of the > groove(at a 45deg angle), and the right channel on the other side. > It seemed plausible that a kick drum panned all the way to one side > or the other might send the needle into the adjacent groove. It shouldn't unless the player has a problem. The cantilever of which the needle is the end has an elastic mount in the cartridge. Together with the mass of the arm and cartridge this forms a spring + mass system, which has a resonance frequency. Above that frequency the arm will not follow the groove, the needle moves relative to the cartridge, and this produces the signal. Below the resonance frequency the whole arm will start to follow the modulation of the groove. On a well-designed system, the resonance frequency should be below the audio range, but still high enough to enable the arm to follow any warping of the disk. If the resonance frequency is in the audio range, then a strong groove modulation below or at that frequency can make the needle jump out. But this means bad design, or the wrong combination of arm and cartridge. 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 unaudio at gmail.com Fri Feb 7 13:16:38 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Fri, 7 Feb 2014 15:16:38 +0200 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140207130405.GB19601@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> Message-ID: > On a well-designed system, the resonance frequency should > be below the audio range, but still high enough to enable > the arm to follow any warping of the disk. Not everyone has a well-designed system. You have to mix for what most of the people who will buy the recording use. From fons at linuxaudio.org Fri Feb 7 13:27:59 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 13:27:59 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> Message-ID: <20140207132759.GD19601@linuxaudio.org> On Fri, Feb 07, 2014 at 03:16:38PM +0200, Vytautas Jancauskas wrote: > > On a well-designed system, the resonance frequency should > > be below the audio range, but still high enough to enable > > the arm to follow any warping of the disk. > > Not everyone has a well-designed system. You have to mix for what most > of the people who will buy the recording use. If that matters it includes using conservative levels instead of going for maximum loudness. 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 Fri Feb 7 13:59:33 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 14:59:33 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140207132759.GD19601@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207132759.GD19601@linuxaudio.org> Message-ID: <1391781573.2060.14.camel@archlinux> On Fri, 2014-02-07 at 13:27 +0000, Fons Adriaensen wrote: > On Fri, Feb 07, 2014 at 03:16:38PM +0200, Vytautas Jancauskas wrote: > > > On a well-designed system, the resonance frequency should > > > be below the audio range, but still high enough to enable > > > the arm to follow any warping of the disk. > > > > Not everyone has a well-designed system. You have to mix for what most > > of the people who will buy the recording use. > > If that matters it includes using conservative levels instead > of going for maximum loudness. +1 and btw. I own a record player that is good enough and you likely will get it at Ebay for 80,-?, I remove the full automatic crap thingy, since this is what could break after some years of intensive usage and now I've got a cheap thingy not as good as a MkII, because if you stop my record player, the ramp-up time is a little bit longer, but good enough for listening. If you aren't DJing it doesn't matter and btw. using a slipmat you don't need to stop a record player, so there would be no ramp-up time and you even could use such an elCheapo record player as mine for DJing. Since my old needle is broken since several years ago, I use an disgusting Audio-Technica cartridge, because I can't pay for a needle that's needed for my good cartridge. A good cartridge or a needle only for such a cartridge is very, very expensive. But a CD player, DAT recorder of high quality is expensive too. IOW even today a good record player with an odd, but still usable cartridge, still isn't more expensive as digital consumer gear. If the needle should jump out of the groove, consider to remove the counterweight and hot-glue a coin to the cartridge ;). From rtg at aapsc.com Fri Feb 7 13:58:49 2014 From: rtg at aapsc.com (Rick Green) Date: Fri, 7 Feb 2014 08:58:49 -0500 (EST) Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140207130405.GB19601@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> Message-ID: On Fri, 7 Feb 2014, Fons Adriaensen wrote: > The cantilever of which the needle is the end has an elastic > mount in the cartridge. Together with the mass of the arm > and cartridge this forms a spring + mass system, which has > a resonance frequency. > > Above that frequency the arm will not follow the groove, > the needle moves relative to the cartridge, and this > produces the signal. Below the resonance frequency the > whole arm will start to follow the modulation of the groove. > > On a well-designed system, the resonance frequency should > be below the audio range, but still high enough to enable > the arm to follow any warping of the disk. > > If the resonance frequency is in the audio range, then > a strong groove modulation below or at that frequency can > make the needle jump out. But this means bad design, or > the wrong combination of arm and cartridge. > Thanks again, Fons. I can always count on you for an absolutely clear explanation of a complex phenomenon. An off-center hole in the record at 45RPM would induce a .75hz noise in the signal. (I don't think any stereo recordings were made in the 78RPM format) A severely warped record might give you three or four waves per revolution, so 6hz max? If the player were designed for audible signals at 20hz and above, it would seem that would be a wide enough target for mechanical resonance of the tone arm. But engineering always demands compromise, and the need to keep the stylus pressure low, and the length of the tone arm short enough to fit in living-room consoles, may force a higher tonearm resonance. I'm not a mechanical engineer, but just from observation of instruments such as organ pipes, Marimba keys, and grand piano bass strings, I imagine achieving a resonance <20hz in a physical structure of ~30cm length would be a non-trivial exercise. -- Rick Green We, the People of the United States of America, reject the U.S. Supreme Court's Citizens United ruling, and move to amend our Constitution to firmly establish that money is not speech, and that human beings, not corporations, are persons entitled to constitutional rights. http://www.MoveToAmend.org From ralf.mardorf at rocketmail.com Fri Feb 7 14:25:00 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 15:25:00 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140207132759.GD19601@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207132759.GD19601@linuxaudio.org> Message-ID: <1391783100.2060.22.camel@archlinux> On Fri, 2014-02-07 at 14:59 +0100, I joked: > If the needle should jump out of the groove, consider to remove the > counterweight and hot-glue a coin to the cartridge ;). I'm serious now :D. Buy records from secondhand record shops instead of remastered vinyl and regarding to new recordings, only buy records from people who composed and mix music, after they learned how to do this and who care about dynamic and room. If you prefer "music" that was made by people on speed and/or MDMA reflect about euthanasia. SICR From fons at linuxaudio.org Fri Feb 7 14:39:48 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 14:39:48 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> Message-ID: <20140207143948.GE19601@linuxaudio.org> On Fri, Feb 07, 2014 at 08:58:49AM -0500, Rick Green wrote: > I'm not a mechanical engineer, but just from observation > of instruments such as organ pipes, Marimba keys, and grand piano > bass strings, I imagine achieving a resonance <20hz in a physical > structure of ~30cm length would be a non-trivial exercise. We're not talking about resonance in the tonearm itself (which can also be a problem, but quite a different one), but the resonance of the combination of the mass of the tonearm and the compliance of the cartridge. See for example 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 idragosani at gmail.com Fri Feb 7 14:41:38 2014 From: idragosani at gmail.com (Brett McCoy) Date: Fri, 7 Feb 2014 09:41:38 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: On Fri, Feb 7, 2014 at 3:04 AM, Louigi Verona wrote: > "Isn't it a disadvantage of such all-in-one-daws that all music produced > with it sounds pretty similar to each other, everybody uses the same samples > and plugins. An creative advantage of modular linuxaudio could be that > you've a high level of creativity and uniqueness in your music." > > I do not agree with the premise that because everybody uses the same plugins > it all sounds similar. As for samples, all-in-one DAW does not force you to > use any samples, rarely does a DAW come with decent samples anyway. And same > plugins can be used in a very different way. That'd be like saying everyone who plays a Gibson SG sounds the same or everyone who composes for a string quartet sounds the same. -- 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 ralf.mardorf at rocketmail.com Fri Feb 7 15:15:36 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 16:15:36 +0100 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140207143948.GE19601@linuxaudio.org> References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207143948.GE19601@linuxaudio.org> Message-ID: <1391786136.2060.27.camel@archlinux> On Fri, 2014-02-07 at 14:39 +0000, Fons Adriaensen wrote: [snip] Something good, but anyway I need to role my eyes. Yes, what you want is a MK II with an Ortofon cartridge, but there are nearly equal solutions with less good record players and less good cartridges available that are much more than just good enough for listening! From ralf.mardorf at rocketmail.com Fri Feb 7 15:27:18 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 16:27:18 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> Message-ID: <1391786838.2060.38.camel@archlinux> On Fri, 2014-02-07 at 09:41 -0500, Brett McCoy wrote: > That'd be like saying everyone who plays a Gibson SG sounds the same > or everyone who composes for a string quartet sounds the same. ;D +1 Alban Berg Quartett ;) I'm a chronic depressive German, so nothing is able to hold a candle for my taste, than Schubert played by the Alban Berg Quartett playing Schubert :p and I really like 80s-no-future-punk-rock :D. Aaargh, I own a nice single coil Ibanez guitar, but I love to play the SG's of friends :). The fingerboard of a SG is close to a classical guitar, the mechanics are bad, but the "rest" is amazing. It's my favourite guitar, just because I grow up as a musician by imitating Jimi, I bought this single coil Ibanez. Today I would buy a SG :). From jeremy at autostatic.com Fri Feb 7 15:54:44 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 07 Feb 2014 16:54:44 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: <1391786838.2060.38.camel@archlinux> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> Message-ID: <52F501C4.70601@autostatic.com> On 02/07/2014 04:27 PM, Ralf Mardorf wrote: > Aaargh, I own a nice single coil Ibanez guitar, but I love to play the > SG's of friends :). The fingerboard of a SG is close to a classical > guitar, the mechanics are bad, but the "rest" is amazing. It's my > favourite guitar, just because I grow up as a musician by imitating > Jimi, I bought this single coil Ibanez. Today I would buy a SG :). As the owner of a 1970 SG I can say that the fingerboard doesn't feel at all like the one from a classical guitar. Also the mechanics of my SG are in pretty good shape for a guitar of 44 years old. Just to make clear that not every SG has the same fingerboard or bad mechanics. And then I'm not talking about how different they can sound. My SG has P90's but I've owned one with humbuckers too which sounded completely different. And then I almost forget the player. Robby Krieger sounds completely different than say Tony Iommi or Angus Young. On-topic, I think the same way about DAW's. I consider generalizing a vice. Best, 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 csanchezgs at gmail.com Fri Feb 7 16:51:10 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 7 Feb 2014 17:51:10 +0100 Subject: [LAU] [OT] Helium: The first supercapacitor-powered portable speaker In-Reply-To: <52F3FCD7.2030609@mellomrommet.no> References: <52F3FCD7.2030609@mellomrommet.no> Message-ID: 2014-02-06 Alf Haakon Lund : > > On 06. feb. 2014 17:34, Carlos sanchiavedraz wrote: >> >> Another FWIW, interesting and OS-HW crowdfunded project: >> https://www.crowdsupply.com/blueshift/helium >> >> At the end of the page there are another interesting projects >> including some related to FX. >> > > Thx for interesting links! > > Alf > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I see myself with a bike/motorbike, a guitar, a MIDI pedalboard controlling looper and FX, the system I'm developing for headless stuff like Raspberry and that kind of speakers... What a mobile rig! -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Fri Feb 7 17:05:24 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 7 Feb 2014 18:05:24 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <20140206230951.GC29528@linuxaudio.org> References: <20140206230951.GC29528@linuxaudio.org> Message-ID: 2014-02-07 Fons Adriaensen : > On Thu, Feb 06, 2014 at 05:40:29PM +0100, Carlos sanchiavedraz wrote: >> Yet another FWIW, interesting and OS-HW project, >> https://phonebloks.com/en/goals > > Three very interesting projects ! Thanks ! > > -- > 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 Hey dear Fons! Yes, that's really interesting. It seems that Motorola was going for that, or bought the company, but Google bought Motorola some time ago, and recently they've sell it to Lenovo but first keeping all Motorola patents (of course). Scary business. Let's see, but I'd love it would run on some kind of Linux to achieve a touchable portable FX Box and Audio workstation. I found another interesting project but related to an open HW and SW robot, but didn't want to flood the list ;). P.S: nice photos of your labs on the Zth's friday interview #16, are you the one with the violin? -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From ralf.mardorf at rocketmail.com Fri Feb 7 17:20:17 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 18:20:17 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] Message-ID: <1391793617.2060.50.camel@archlinux> -------- Forwarded Message -------- From: Ralf Mardorf To: linux-audio-user Subject: Re: [LAU] Bitwig at long last...? Date: Fri, 07 Feb 2014 18:19:21 +0100 Mailer: Evolution 3.10.3 On Fri, 2014-02-07 at 16:54 +0100, Jeremy Jongepier wrote: > On 02/07/2014 04:27 PM, Ralf Mardorf wrote: > > Aaargh, I own a nice single coil Ibanez guitar, but I love to play the > > SG's of friends :). The fingerboard of a SG is close to a classical > > guitar, the mechanics are bad, but the "rest" is amazing. It's my > > favourite guitar, just because I grow up as a musician by imitating > > Jimi, I bought this single coil Ibanez. Today I would buy a SG :). > > As the owner of a 1970 SG I can say that the fingerboard doesn't feel at > all like the one from a classical guitar. Also the mechanics of my SG > are in pretty good shape for a guitar of 44 years old. Just to make > clear that not every SG has the same fingerboard or bad mechanics. And > then I'm not talking about how different they can sound. My SG has P90's > but I've owned one with humbuckers too which sounded completely > different. And then I almost forget the player. Robby Krieger sounds > completely different than say Tony Iommi or Angus Young. > On-topic, I think the same way about DAW's. I consider generalizing a vice. :) No comment ;), no, a comment, I suspect you're aware about what I'm talking about ;). The fingerboard is more like a classical guitar, than a Stratocaster or Stratocaster alike guitar and all SG humbuckers from different ages have a special unique sound _and_ a Schaller M6 Mini or derivative definitively is better than an original (Conclusion or what ever is the name) opened SG mechanic. From fons at linuxaudio.org Fri Feb 7 17:46:46 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 17:46:46 +0000 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: References: <20140206230951.GC29528@linuxaudio.org> Message-ID: <20140207174646.GF19601@linuxaudio.org> On Fri, Feb 07, 2014 at 06:05:24PM +0100, Carlos sanchiavedraz wrote: > P.S: nice photos of your labs on the Zth's friday interview #16, are > you the one with the violin? No, I'm the one sitting at the computer in the back. The violins are Guarneri, some of the most expensive instruments ever. 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 idragosani at gmail.com Fri Feb 7 17:47:54 2014 From: idragosani at gmail.com (Brett McCoy) Date: Fri, 7 Feb 2014 12:47:54 -0500 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] In-Reply-To: <1391793617.2060.50.camel@archlinux> References: <1391793617.2060.50.camel@archlinux> Message-ID: On Fri, Feb 7, 2014 at 12:20 PM, Ralf Mardorf wrote: > -------- Forwarded Message -------- > :) No comment ;), no, a comment, I suspect you're aware about what I'm > talking about ;). The fingerboard is more like a classical guitar, than > a Stratocaster or Stratocaster alike guitar and all SG humbuckers from > different ages have a special unique sound _and_ a Schaller M6 Mini or > derivative definitively is better than an original (Conclusion or what > ever is the name) opened SG mechanic. I have to disagree also, SG is not at all like a classical guitar, no more and no less than a Strat, Les Paul or Ibanez fingerboard. Why do you think SG fingerboard is like a classical fingerboard? The mechanics are probably not too different from a Les Paul's, it may depend on the year of year issue, too. -- 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 ralf.mardorf at rocketmail.com Fri Feb 7 18:00:37 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 19:00:37 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] In-Reply-To: References: <1391793617.2060.50.camel@archlinux> Message-ID: <1391796037.2060.53.camel@archlinux> On Fri, 2014-02-07 at 12:47 -0500, Brett McCoy wrote: > On Fri, Feb 7, 2014 at 12:20 PM, Ralf Mardorf > wrote: > > -------- Forwarded Message -------- > > > :) No comment ;), no, a comment, I suspect you're aware about what I'm > > talking about ;). The fingerboard is more like a classical guitar, than > > a Stratocaster or Stratocaster alike guitar and all SG humbuckers from > > different ages have a special unique sound _and_ a Schaller M6 Mini or > > derivative definitively is better than an original (Conclusion or what > > ever is the name) opened SG mechanic. > > I have to disagree also, SG is not at all like a classical guitar, no > more and no less than a Strat, Les Paul or Ibanez fingerboard. Why do > you think SG fingerboard is like a classical fingerboard? The > mechanics are probably not too different from a Les Paul's, it may > depend on the year of year issue, too. Yesno ;), the SGs fingerboard is not equal, but more flat like a fingerboard of a classical guitar. From ralf.mardorf at rocketmail.com Fri Feb 7 18:13:08 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 19:13:08 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...? Message-ID: <1391796788.2060.61.camel@archlinux> On Fri, 2014-02-07 at 19:00 +0100, Ralf Mardorf wrote: > On Fri, 2014-02-07 at 12:47 -0500, Brett McCoy wrote: > > On Fri, Feb 7, 2014 at 12:20 PM, Ralf Mardorf > > wrote: > > > -------- Forwarded Message -------- > > > > > :) No comment ;), no, a comment, I suspect you're aware about what I'm > > > talking about ;). The fingerboard is more like a classical guitar, than > > > a Stratocaster or Stratocaster alike guitar and all SG humbuckers from > > > different ages have a special unique sound _and_ a Schaller M6 Mini or > > > derivative definitively is better than an original (Conclusion or what > > > ever is the name) opened SG mechanic. > > > > I have to disagree also, SG is not at all like a classical guitar, no > > more and no less than a Strat, Les Paul or Ibanez fingerboard. Why do > > you think SG fingerboard is like a classical fingerboard? The > > mechanics are probably not too different from a Les Paul's, it may > > depend on the year of year issue, too. > > Yesno ;), the SGs fingerboard is not equal, but more flat like a > fingerboard of a classical guitar. And an original LesPaul mechanics is equal to a SG's mechanics, at least for the ones I don't own, but I played owned by friends. I love my Ibanez, but it isn't as good as a Gibson, not even as good as a Fender, just the Ibanez mechanics is a little bit better, while my closed Ibanez Shaller like mechanics already is broken, but still in a better shape than the opened old original Gibson mechanics I know. From ralf.mardorf at rocketmail.com Fri Feb 7 18:33:29 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 07 Feb 2014 19:33:29 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] In-Reply-To: References: <1391793617.2060.50.camel@archlinux> Message-ID: <1391798009.2060.67.camel@archlinux> On Fri, 2014-02-07 at 12:47 -0500, Brett McCoy wrote: > On Fri, Feb 7, 2014 at 12:20 PM, Ralf Mardorf > wrote: > > -------- Forwarded Message -------- > > > :) No comment ;), no, a comment, I suspect you're aware about what I'm > > talking about ;). The fingerboard is more like a classical guitar, than > > a Stratocaster or Stratocaster alike guitar and all SG humbuckers from > > different ages have a special unique sound _and_ a Schaller M6 Mini or > > derivative definitively is better than an original (Conclusion or what > > ever is the name) opened SG mechanic. > > I have to disagree also, SG is not at all like a classical guitar, no > more and no less than a Strat, Les Paul or Ibanez fingerboard. Why do > you think SG fingerboard is like a classical fingerboard? The > mechanics are probably not too different from a Les Paul's, it may > depend on the year of year issue, too. We could continue to disagree about the tremolo ;). I own an old school crappy thing. It's nice, but already unusable for 80s Metal/Punk ;). No Floyd Rose here ;). It has got its advantages, but much more disadvantages :D. I sound detuned as Jimi did, but IMO this isn't an advantage, it's disgusting for my taste. From csanchezgs at gmail.com Fri Feb 7 18:37:18 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 7 Feb 2014 19:37:18 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <20140207174646.GF19601@linuxaudio.org> References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> Message-ID: 2014-02-07 Fons Adriaensen : > On Fri, Feb 07, 2014 at 06:05:24PM +0100, Carlos sanchiavedraz wrote: > >> P.S: nice photos of your labs on the Zth's friday interview #16, are >> you the one with the violin? > > No, I'm the one sitting at the computer in the back. Ok, unknown face then ;) (until that great day I can go to some LAC). I also try to keep privacy and a relative anonymity... all that you can in these days. > > The violins are Guarneri, some of the most expensive > instruments ever. What a pleasure it has to be to play and listen to them! > > 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) > -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From abonnements at revolwear.com Fri Feb 7 19:09:41 2014 From: abonnements at revolwear.com (Max) Date: Sat, 08 Feb 2014 04:09:41 +0900 Subject: [LAU] Pulseaudio and Jack Message-ID: <52F52F75.2020106@revolwear.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Which of the suggested option is the one people here on the list favour? http://jackaudio.org/pulseaudio_and_jack -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.14 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlL1L3UACgkQ3EB7kzgMM6Ln1ACfVyT8RXVcBzHiIkyQJU9LWBWK 4p4Anj08pRcsv5ytIxvtrebWhbkQulGA =JPXo -----END PGP SIGNATURE----- From idragosani at gmail.com Fri Feb 7 19:18:28 2014 From: idragosani at gmail.com (Brett McCoy) Date: Fri, 7 Feb 2014 14:18:28 -0500 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F52F75.2020106@revolwear.com> References: <52F52F75.2020106@revolwear.com> Message-ID: I am sure there are representatives of every option. I use option 3, I keep Jack running all of the time and route pulseaudio to it. On Fri, Feb 7, 2014 at 2:09 PM, Max wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Which of the suggested option is the one people here on the list favour? > > http://jackaudio.org/pulseaudio_and_jack > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.14 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ > > iEYEARECAAYFAlL1L3UACgkQ3EB7kzgMM6Ln1ACfVyT8RXVcBzHiIkyQJU9LWBWK > 4p4Anj08pRcsv5ytIxvtrebWhbkQulGA > =JPXo > -----END PGP SIGNATURE----- > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- 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 fons at linuxaudio.org Fri Feb 7 19:31:08 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 7 Feb 2014 19:31:08 +0000 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> Message-ID: <20140207193108.GA3525@linuxaudio.org> On Fri, Feb 07, 2014 at 07:37:18PM +0100, Carlos sanchiavedraz wrote: > Ok, unknown face then ;) (until that great day I can go to some LAC). I'm the one with the green (LAC 2013) t-shirt in one of the other pics. 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 Fri Feb 7 21:08:26 2014 From: len at ovenwerks.net (Len Ovens) Date: Fri, 7 Feb 2014 13:08:26 -0800 (PST) Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F52F75.2020106@revolwear.com> References: <52F52F75.2020106@revolwear.com> Message-ID: On Sat, 8 Feb 2014, Max wrote: > Which of the suggested option is the one people here on the list favour? > > http://jackaudio.org/pulseaudio_and_jack It depends. Each option is the best in the right place. My system happens to be set up as option 3, but if I want no PA, then I disable the bridge by telling PA to remove that module. -- Len Ovens www.ovenwerks.net From danstowell+lxau at gmail.com Fri Feb 7 21:43:26 2014 From: danstowell+lxau at gmail.com (Dan S) Date: Fri, 7 Feb 2014 21:43:26 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140205153636.GA29740@septictank.raw-sewage.fake> References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: Hi Matt, Here's a Python script which analyses a set of files in a specified folder. They have to be file formats understood by libsndfile - which allows for flac/wav/aiff and some others, but not mp3. It takes maybe 5 to 10 seconds per track, but eventually it produces a plot as a PNG file. https://gist.github.com/danstowell/8872466 (Also blogged: ) Best Dan 2014-02-05 Matt Garman : > > I have a collection of FLAC files, all ripped from my CD collection > What I would like to do is run an analysis across all the music to > determine how the bass/lower frequencies are generally mixed. For > example, how much content below (for example) 150 Hz is on the left > channel versus the right channel? > > I'm not sure if "histogram" is the right word, but in my mind what > I'd like to see, per-channel, is something like this: > > 150--125 Hz: x samples > 125--100 Hz: y samples > 100--80 Hz: z samples > ... > > Then I can look at the two channels of a song, and if the histograms > are approximately the same, I can assume the bass was mixed equally > to both channels. > > I am a programmer, and thought it would be easy to quickly hack > something up that would do this, but I have no experience with > signal processing, and as I started reading about this, I quickly > got in over my head! So I was hoping there might already exist a > tool that has this functionality. > > Note that I don't need any kind of graphical output, as this needs > to be wrapped up in some kind of batch processing script---I have > about 11,000 files to analyze! > > The motivation for this is: I have a hardware DAC (digital audio > converter) in one part of my house, and a subwoofer in another. > There is a single coax run between the DAC and subwoofer, so I can > only send one channel. If the overwhelming majority of my music has > the bass mixed equally, sending only one channel isn't a problem. > But if I choose the "L" channel to send to the sub, and much music > has the bass mixed only to the "R" channel, then I won't be able to > hear the low frequencies. I want to find out how often this might > happen. > > Thanks, > Matt > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- http://www.mcld.co.uk From david.santamauro at gmail.com Fri Feb 7 23:53:07 2014 From: david.santamauro at gmail.com (David Santamauro) Date: Fri, 07 Feb 2014 18:53:07 -0500 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F52F75.2020106@revolwear.com> References: <52F52F75.2020106@revolwear.com> Message-ID: <52F571E3.9050005@gmail.com> On 02/07/2014 02:09 PM, Max wrote: > Which of the suggested option is the one people here on the list favour? Option 3, and as Len says, removing module removes PA. From harryhaaren at gmail.com Sat Feb 8 00:09:00 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sat, 8 Feb 2014 00:09:00 +0000 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F52F75.2020106@revolwear.com> References: <52F52F75.2020106@revolwear.com> Message-ID: On Fri, Feb 7, 2014 at 7:09 PM, Max wrote: > Which of the suggested option is the one people here on the list favour? Option 1, no pulse at all. Using "raw" ALSA for other apps, libflashsupport-jack for Youtube etc->JACK. -------------- next part -------------- An HTML attachment was scrubbed... URL: From cbannister at slingshot.co.nz Sat Feb 8 00:15:52 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Sat, 8 Feb 2014 13:15:52 +1300 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> Message-ID: <20140208001552.GE19911@tal> On Thu, Feb 06, 2014 at 11:56:00PM -0500, Rick Green wrote: > I had heard the story about phonograph needles jumping out of the > track. It made sense to me. On a stereo phonograph record, in-phase > energy (L+R) is represented as side-to-side motion of the needle, > and out-of-phase energy(L-R) is up and down motion. It can also be > thought of as the left channel is recorded on one side of the > groove(at a 45deg angle), and the right channel on the other side. > It seemed plausible that a kick drum panned all the way to one side > or the other might send the needle into the adjacent groove. It would be a pretty crappy turntable that did that! Probably a plastic arm with no counterbalance with a worn dirty needle. -- "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 Sat Feb 8 02:01:55 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 03:01:55 +0100 Subject: [LAU] Pulseaudio and Jack In-Reply-To: References: <52F52F75.2020106@revolwear.com> Message-ID: <1391824915.4945.5.camel@archlinux> On Sat, 2014-02-08 at 00:09 +0000, Harry van Haaren wrote: > On Fri, Feb 7, 2014 at 7:09 PM, Max wrote: > > Which of the suggested option is the one people here on the list > favour? > > > Option 1, no pulse at all. No pulseaudio too. Assumed the OP won't recompile some apps, resp. not rebuild some packages that hard depend on pulseaudio, building a dummy package would solve this issue. Howto build dummy packages for DEB: http://www.debian.org/doc/manuals/apt-howto/ch-helpers.en.html For Arch Linux: https://wiki.archlinux.org/index.php/PKGBUILD From cbannister at slingshot.co.nz Sat Feb 8 04:52:28 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Sat, 8 Feb 2014 17:52:28 +1300 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <1391781573.2060.14.camel@archlinux> References: <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207132759.GD19601@linuxaudio.org> <1391781573.2060.14.camel@archlinux> Message-ID: <20140208045228.GH31249@tal> On Fri, Feb 07, 2014 at 02:59:33PM +0100, Ralf Mardorf wrote: > use an disgusting Audio-Technica cartridge, because I can't pay for a > needle that's needed for my good cartridge. A good cartridge or a needle > only for such a cartridge is very, very expensive. Interesting read: http://www.nytimes.com/2012/04/19/technology/personaltech/how-to-enjoy-turntables-without-obsessing-over-them.html -- "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 cbannister at slingshot.co.nz Sat Feb 8 04:57:45 2014 From: cbannister at slingshot.co.nz (Chris Bannister) Date: Sat, 8 Feb 2014 17:57:45 +1300 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <1391783100.2060.22.camel@archlinux> References: <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207132759.GD19601@linuxaudio.org> <1391783100.2060.22.camel@archlinux> Message-ID: <20140208045745.GI31249@tal> On Fri, Feb 07, 2014 at 03:25:00PM +0100, Ralf Mardorf wrote: > On Fri, 2014-02-07 at 14:59 +0100, I joked: > > If the needle should jump out of the groove, consider to remove the > > counterweight and hot-glue a coin to the cartridge ;). > > I'm serious now :D. > > Buy records from secondhand record shops instead of remastered vinyl and There's no guarantee that buying records from secondhand record shops are any better. Some of the old recordings were crap some were brilliant, you soon got to know who were the craftsman and who were just knob twiddlers. -- "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 gnome at hawaii.rr.com Sat Feb 8 06:08:29 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:08:29 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F501C4.70601@autostatic.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> <52F501C4.70601@autostatic.com> Message-ID: <52F5C9DD.6000607@hawaii.rr.com> On 02/07/2014 05:54 AM, Jeremy Jongepier wrote: > On 02/07/2014 04:27 PM, Ralf Mardorf wrote: >> Aaargh, I own a nice single coil Ibanez guitar, but I love to play the >> SG's of friends :). The fingerboard of a SG is close to a classical >> guitar, the mechanics are bad, but the "rest" is amazing. It's my >> favourite guitar, just because I grow up as a musician by imitating >> Jimi, I bought this single coil Ibanez. Today I would buy a SG :). > > As the owner of a 1970 SG I can say that the fingerboard doesn't feel at > all like the one from a classical guitar. Also the mechanics of my SG > are in pretty good shape for a guitar of 44 years old. Just to make > clear that not every SG has the same fingerboard or bad mechanics. And > then I'm not talking about how different they can sound. My SG has P90's > but I've owned one with humbuckers too which sounded completely > different. And then I almost forget the player. Robby Krieger sounds > completely different than say Tony Iommi or Angus Young. > On-topic, I think the same way about DAW's. I consider generalizing a vice. Lead guitarist I played with in 1973-1975 had a 1959 Gibson SG. It didn't feel or sound anything like my classical guitar (1973 Garcia Concert Model 3). In his hands, it was always high-treble with the wah-wah cranked for high treble; I still have a hearing loss in that range courtesy of him. Who in a garage band used earplugs back then? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:12:20 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:12:20 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: <1391786838.2060.38.camel@archlinux> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> Message-ID: <52F5CAC4.2060505@hawaii.rr.com> On 02/07/2014 05:27 AM, Ralf Mardorf wrote: > On Fri, 2014-02-07 at 09:41 -0500, Brett McCoy wrote: >> That'd be like saying everyone who plays a Gibson SG sounds the same >> or everyone who composes for a string quartet sounds the same. > > ;D > > +1 > > Alban Berg Quartett ;) > > I'm a chronic depressive German, so nothing is able to hold a candle > for my taste, than Schubert played by the Alban Berg Quartett playing > Schubert :p Or Rachmaninoff cello music played by my relative, Susan Lamb Cook? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:18:01 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:18:01 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: <1391777238.667.14.camel@archlinux> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391777238.667.14.camel@archlinux> Message-ID: <52F5CC19.4050006@hawaii.rr.com> On 02/07/2014 02:47 AM, Ralf Mardorf wrote: > On Fri, 2014-02-07 at 12:04 +0400, Louigi Verona wrote: >> Also, it is funny you say that. Doesn't every person here use same >> stuff - a couple of main synths and a limited set of plugins? > > No! I play guitars and own 19" effects and hardware synth. Look at Jack White. He loves vintage analog equipment and effects. And the guitarist in my church band keeps his little old effects pedal limping along because he says he can't get the same sound from the much newer effects box he bought to replace it. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:22:13 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:22:13 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F4B0D3.2020506@alextone.info> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <52F49664.1040501@gmail.com> <52F4B0D3.2020506@alextone.info> Message-ID: <52F5CD15.5090907@hawaii.rr.com> On 02/07/2014 12:09 AM, Alex wrote: > The idea that daws sound the "same" has been refuted fairly well by > previous commenters. > > I'll add that using sample libs is arguably the most likely to sound > similar if users record with them in their natural state, and not work > at developing a unique sound with the same base material. That and using stock synth voices. I'm lazy about that. Or maybe I just really like stock voices. ;-) -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:33:11 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:33:11 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F49664.1040501@gmail.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <52F49664.1040501@gmail.com> Message-ID: <52F5CFA7.8000101@hawaii.rr.com> On 02/06/2014 10:16 PM, Simon Wise wrote: > On 07/02/14 19:04, Louigi Verona wrote: >> "Isn't it a disadvantage of such all-in-one-daws that all music produced >> with it sounds pretty similar to each other, everybody uses the same >> samples and plugins. An creative advantage of modular linuxaudio could be >> that you've a high level of creativity and uniqueness in your music." >> >> I do not agree with the premise that because everybody uses the same >> plugins it all sounds similar. As for samples, all-in-one DAW does not >> force you to use any samples, rarely does a DAW come with decent samples >> anyway. And same plugins can be used in a very different way. >> >> Also, it is funny you say that. Doesn't every person here use same >> stuff - >> a couple of main synths and a limited set of plugins? > > or maybe some record a cello ... again, very different sounds indeed > using the same 'limited' configuration of wood, strings, bow and mic ... > depending very much on the musician using it. Part of my childish fantasy about composing for "real" instruments like strings or horns or organ is that the sampled or synthesized sound is just a placeholder until I can have a real instrumentalist play the part on a real instrument for a real recording. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:39:32 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:39:32 -1000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: <20140208045745.GI31249@tal> References: <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206181213.GA8541@gjcp.net> <20140207130405.GB19601@linuxaudio.org> <20140207132759.GD19601@linuxaudio.org> <1391783100.2060.22.camel@archlinux> <20140208045745.GI31249@tal> Message-ID: <52F5D124.5090808@hawaii.rr.com> On 02/07/2014 06:57 PM, Chris Bannister wrote: > On Fri, Feb 07, 2014 at 03:25:00PM +0100, Ralf Mardorf wrote: >> On Fri, 2014-02-07 at 14:59 +0100, I joked: >>> If the needle should jump out of the groove, consider to remove the >>> counterweight and hot-glue a coin to the cartridge ;). >> >> I'm serious now :D. >> >> Buy records from secondhand record shops instead of remastered vinyl and > > There's no guarantee that buying records from secondhand record shops > are any better. Some of the old recordings were crap some were > brilliant, you soon got to know who were the craftsman and who were just > knob twiddlers. And there's a good chance that the secondhand record hasn't particularly been cared for very well before it was sold to the secondhand store. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 06:54:03 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 20:54:03 -1000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> <20140206101728.GA13289@linuxaudio.org> <20140206145924.GB13289@linuxaudio.org> <20140206172037.GD13289@linuxaudio.org> <1391717608.690.16.camel@archlinux> Message-ID: <52F5D48B.1030109@hawaii.rr.com> On 02/06/2014 09:17 PM, Vytautas Jancauskas wrote: >> And if you gouge holes in a CD no data gets lost ;). JFTR a Beatles bass >> does sound natural, however, even a "Bring the noise"-"Bass! How low can >> you go?"-unnatural-bass mixed to one channel only wouldn't make the >> needle jump, the needle will jump if you didn't adjust the counterweight >> correctly and this btw. isn't the only thing you can adjust for a good >> record player. I'm not speaking about audiophile record players, I'm >> talking about good record players, e.g. the well known DJ model, but >> even some HiFi players are very good. No CD is able to hold a candle to >> a record played on a good record player. > > I'm not going to pretend I understood anything of this. > >> I don't know anybody who claims this bass nonsense. > > I gave you a link, but if you look around there are plenty of sources > that claim this, for example here is what soundonsound says: > > "There are few 'rules' in music production, but panning bass isn't far > off. It is usually a good idea to pan the bass and kick to the centre. > Partly this is historical Or with where the instruments themselves might usually be located on stage? I always have this image of Ye Old Rock'n'Roll Band on stage, with the drummer centered in the back, bassist to one side or the other of the drums, keyboards maybe on the other side from the bassist. Emerson, Lake & Palmer, Cream, Led Zeppelin. I think even Chuck Berry usually set up that way. I remember when my church band was recording with 16 tracks (before the death of Firewire on laptops), I'd center the drums and bass and pan the other instruments and singers to try to match their locations on stage. But that could just be me; I like live recordings where I can hear where each performer is. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Feb 8 07:02:49 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 21:02:49 -1000 Subject: [LAU] Pulseaudio and Jack In-Reply-To: References: <52F52F75.2020106@revolwear.com> Message-ID: <52F5D699.8050105@hawaii.rr.com> On 02/07/2014 02:09 PM, Harry van Haaren wrote: > On Fri, Feb 7, 2014 at 7:09 PM, Max wrote: > > Which of the suggested option is the one people here on the list favour? > > Option 1, no pulse at all. > Using "raw" ALSA for other apps, libflashsupport-jack for Youtube etc->JACK. Hmm, I don't find that in Debian Sid repository. Search for "libflashsupport" only finds flashplugin-nonfree-extrasound:i386, which only connects Flash to Esound or OSS before falling back to ALSA. Is it available from somewhere else? -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From hodginson at gmail.com Sat Feb 8 07:30:04 2014 From: hodginson at gmail.com (Chris Hogan) Date: Sat, 8 Feb 2014 18:30:04 +1100 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F5D699.8050105@hawaii.rr.com> References: <52F52F75.2020106@revolwear.com> <52F5D699.8050105@hawaii.rr.com> Message-ID: >> Hmm, I don't find that in Debian Sid repository. Search for >> "libflashsupport" only finds flashplugin-nonfree-extrasound:i386, >> which only connects Flash to Esound or OSS before falling back to ALSA. >> >> Is it available from somewhere else? It doesn't seem to have a web page but you can grab the source from here: git clone git://repo.or.cz/libflashsupport-jack.git -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Sat Feb 8 07:37:42 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 07 Feb 2014 21:37:42 -1000 Subject: [LAU] Pulseaudio and Jack In-Reply-To: References: <52F52F75.2020106@revolwear.com> <52F5D699.8050105@hawaii.rr.com> Message-ID: <52F5DEC6.3060204@hawaii.rr.com> On 02/07/2014 09:30 PM, Chris Hogan wrote: >>> Hmm, I don't find that in Debian Sid repository. Search for >>> "libflashsupport" only finds flashplugin-nonfree-extrasound:i386, > >>> which only connects Flash to Esound or OSS before falling back to ALSA. >>> >>> Is it available from somewhere else? > > It doesn't seem to have a web page but you can grab the source from here: > > git clone git://repo.or.cz/libflashsupport-jack.git Thanks, I'll see about doing that. For the longest time, I had to download any Flash video I wanted to listen to because Flash would only play through the "default" audio on my old laptop. The "default" audio was the builtin sound that didn't work anymore ... Then I figured out an .asoundrc that defined the default card to be my USB sound card - but it would only play through ALSA, not JACK. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From harryhaaren at gmail.com Sat Feb 8 13:59:28 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sat, 8 Feb 2014 13:59:28 +0000 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F5D699.8050105@hawaii.rr.com> References: <52F52F75.2020106@revolwear.com> <52F5D699.8050105@hawaii.rr.com> Message-ID: On Sat, Feb 8, 2014 at 7:02 AM, david wrote: > Is it available from somewhere else? The repo link has been posted, adding this for completeness: http://jackaudio.org/routing_flash Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From idragosani at gmail.com Sat Feb 8 14:14:31 2014 From: idragosani at gmail.com (Brett McCoy) Date: Sat, 8 Feb 2014 09:14:31 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F5C9DD.6000607@hawaii.rr.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> <52F501C4.70601@autostatic.com> <52F5C9DD.6000607@hawaii.rr.com> Message-ID: On Sat, Feb 8, 2014 at 1:08 AM, david wrote: > Lead guitarist I played with in 1973-1975 had a 1959 Gibson SG. It didn't > feel or sound anything like my classical guitar (1973 Garcia Concert Model > 3). In his hands, it was always high-treble with the wah-wah cranked for > high treble; I still have a hearing loss in that range courtesy of him. Who > in a garage band used earplugs back then? psshh... I still don't wear earplugs :-P -- 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 ralf.mardorf at rocketmail.com Sat Feb 8 14:43:23 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 15:43:23 +0100 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> <52F501C4.70601@autostatic.com> <52F5C9DD.6000607@hawaii.rr.com> Message-ID: <1391870603.7009.47.camel@archlinux> On Sat, 2014-02-08 at 09:14 -0500, Brett McCoy wrote: > psshh... I still don't wear earplugs :-P I dislike my cheap earplugs, but I plug tissues or similar into the ear canal. To rock without protection is very dangerous. OTOH for me acute hearing loss more often was caused psychologically than by too loud music. The medicine you get is poison and it's without guarantee. If you really lose hearing of some frequencies, it can't be healed anymore. Better protect your ears in the future. Btw. in Germany the maintenance of industrial health and safety standards are null and void if you work with children. By rights people should wear hearing protection when working with young children. From el.doctor at laposte.net Sat Feb 8 16:43:50 2014 From: el.doctor at laposte.net (MK aka El Doctor) Date: Sat, 08 Feb 2014 17:43:50 +0100 Subject: [LAU] io GNU/Linux new iso uploaded :) Message-ID: <1576592.CrZYUIedbD@io> Hi, Well I've been working hard and new images (32 or 64bit) are ready for testing :) "io GNU/Linux is a Live DVD/USB based on the free Operating System Debian (Sid)... and includes a large collection of preinstalled programs for all uses, especially multimedia creation" This new version brings a lot of bug fixes, updated softwares and new features: (systemd, rtirq, encrypted persistence, a getting started and more)... For screenshots, package lists, infos etc... visit: -> http://manu.kebab.free.fr/iognulinux.html A small demo video is available: -> https://www.youtube.com/watch?v=1UCB5je9lJo Preparing an USB flash drive (with persistence) HowTo: -> https://sourceforge.net/p/io-gnu-linux/wiki/USB%20install%20howto Enjoy and happy weekend to all :) MK -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: From blablack at gmail.com Sat Feb 8 17:13:57 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sat, 8 Feb 2014 17:13:57 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial Message-ID: Hello all, Today I just released the version 1.0.2 of the ams-lv2 plugins. ams-lv2 is a port of alsa-modular-synth in the lv2 format to create modular synth primarily with Ingen. Source code and more information available on the website: http://objectivewave.wordpress.com/ams-lv2/ To celebrate this release, I created two videos: The first is a demo of 3 synths created with ams-lv2 and Ingen (a bass, a lead and a pad) soloed and inside a mix: http://www.youtube.com/watch?v=LWfF71NerkQ I created as well this first tutorial on the basics of ams-lv2: http://www.youtube.com/watch?v=IuQZajaSw6M I would as well like to take advantage to thank a lot of people who helped with this project: - David for his patience to answer my 1000+ questions and for all his work on lv2 and Ingen - Robin for creating the sisco.lv2 plugins, they are featured in the demo and they are a great help when creating modular synths - Harry for the dials used in the GUI of ams-lv2 - and finally Fons for creating the great alsa-modular-synth in the first place - the whole Linux Audio community in general I really hope you enjoy these plugins and find them useful somehow! Aur?lien From csanchezgs at gmail.com Sat Feb 8 18:08:08 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sat, 8 Feb 2014 19:08:08 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <20140207193108.GA3525@linuxaudio.org> References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> <20140207193108.GA3525@linuxaudio.org> Message-ID: 2014-02-07 20:31 GMT+01:00 Fons Adriaensen : > On Fri, Feb 07, 2014 at 07:37:18PM +0100, Carlos sanchiavedraz wrote: > >> Ok, unknown face then ;) (until that great day I can go to some LAC). > > I'm the one with the green (LAC 2013) t-shirt in one of the other pics. > > 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) > Nooow I see you, great. P.S: what a well-meaning envy seeing those installations, labs and that kind if research. P.S.2: talking about well-meaning envy, and myself being some kind of a musician in some way, I've found this beautiful sentence: "Eisenberg goes on to describe musicians as "self-conscious birds" who have music both in their muscles and in their minds.And that must be a wonderful state of being. It is for this reason that I have a certain well-meaning envy of musicians." -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From ralf.mardorf at rocketmail.com Sat Feb 8 18:33:53 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 19:33:53 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> <20140207193108.GA3525@linuxaudio.org> Message-ID: <1391884433.7009.65.camel@archlinux> On Sat, 2014-02-08 at 19:08 +0100, Carlos sanchiavedraz wrote: > P.S.2: talking about well-meaning envy, and myself being some kind of > a musician in some way, I've found this beautiful sentence: > "Eisenberg goes on to describe musicians as "self-conscious birds" who > have music both in their muscles and in their minds.And that must be a > wonderful state of being. It is for this reason that I have a certain > well-meaning envy of musicians." Music seems to be a good training against dementia and even seems to protect against effects of Alzheimer's disease, OTOH a lot of disorders are more often for artists, depressions, personality disorders, left vs right cerebral hemisphere, IOW artists are more often left-hander, dyslexics etc. and even while making music seems to protect against dementia and effects of Alzheimer, musicians are more often addicted to evil, life-endangering drugs, they are more often nerds etc., so the resume seems to be that the pros come with contras?. IMO the only real pro of being a musician is to have fun, when making music and many of us "suffer" from synesthesia, so we don't need drugs to experience what most people only could experience when taking drugs. From blablack at gmail.com Sat Feb 8 18:44:12 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sat, 8 Feb 2014 18:44:12 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: On 8 Feb 2014 18:16, "mark hadman" wrote: > > Interesting. I can't try it with Ingen though, because its dependency > ganv has been unbuildable for years (Arch Linux). Does it work with > another lv2 host? > I'd say no. Only Ingen provides support for CV port and the possibility to connect plugins the modular way. Has this issue with Ganv and Arch Linux be reported to David? > On 8 February 2014 17:13, Aur?lien Leblond wrote: > > Hello all, > > > > Today I just released the version 1.0.2 of the ams-lv2 plugins. > > ams-lv2 is a port of alsa-modular-synth in the lv2 format to create > > modular synth primarily with Ingen. > > Source code and more information available on the website: > > http://objectivewave.wordpress.com/ams-lv2/ > > > > To celebrate this release, I created two videos: > > > > The first is a demo of 3 synths created with ams-lv2 and Ingen (a > > bass, a lead and a pad) soloed and inside a mix: > > http://www.youtube.com/watch?v=LWfF71NerkQ > > > > I created as well this first tutorial on the basics of ams-lv2: > > http://www.youtube.com/watch?v=IuQZajaSw6M > > > > > > I would as well like to take advantage to thank a lot of people who > > helped with this project: > > - David for his patience to answer my 1000+ questions and for all his > > work on lv2 and Ingen > > - Robin for creating the sisco.lv2 plugins, they are featured in the > > demo and they are a great help when creating modular synths > > - Harry for the dials used in the GUI of ams-lv2 > > - and finally Fons for creating the great alsa-modular-synth in the first place > > - the whole Linux Audio community in general > > > > I really hope you enjoy these plugins and find them useful somehow! > > > > Aur?lien > > _______________________________________________ > > 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 Sat Feb 8 18:47:23 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 8 Feb 2014 18:47:23 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <20140208184723.GA11049@linuxaudio.org> On Sat, Feb 08, 2014 at 05:13:57PM +0000, Aur?lien Leblond wrote: > - and finally Fons for creating the great alsa-modular-synth in the first place Credit for that should go to Matthias Nagorni. My part in AMS is maybe five percent. 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 federicogalland at gmail.com Sat Feb 8 18:57:50 2014 From: federicogalland at gmail.com (F Tux) Date: Sat, 8 Feb 2014 16:57:50 -0200 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: I was wondering where the avw.lv2 project had ended. Thanks a lot! On 2/8/14, Aur?lien Leblond wrote: > Hello all, > > Today I just released the version 1.0.2 of the ams-lv2 plugins. > ams-lv2 is a port of alsa-modular-synth in the lv2 format to create > modular synth primarily with Ingen. > Source code and more information available on the website: > http://objectivewave.wordpress.com/ams-lv2/ > > To celebrate this release, I created two videos: > > The first is a demo of 3 synths created with ams-lv2 and Ingen (a > bass, a lead and a pad) soloed and inside a mix: > http://www.youtube.com/watch?v=LWfF71NerkQ > > I created as well this first tutorial on the basics of ams-lv2: > http://www.youtube.com/watch?v=IuQZajaSw6M > > > I would as well like to take advantage to thank a lot of people who > helped with this project: > - David for his patience to answer my 1000+ questions and for all his > work on lv2 and Ingen > - Robin for creating the sisco.lv2 plugins, they are featured in the > demo and they are a great help when creating modular synths > - Harry for the dials used in the GUI of ams-lv2 > - and finally Fons for creating the great alsa-modular-synth in the first > place > - the whole Linux Audio community in general > > I really hope you enjoy these plugins and find them useful somehow! > > Aur?lien > _______________________________________________ > 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 Feb 8 18:59:15 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 19:59:15 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <1391884433.7009.65.camel@archlinux> References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> <20140207193108.GA3525@linuxaudio.org> <1391884433.7009.65.camel@archlinux> Message-ID: <1391885955.7009.74.camel@archlinux> On Sat, 2014-02-08 at 19:33 +0100, Ralf Mardorf wrote: > On Sat, 2014-02-08 at 19:08 +0100, Carlos sanchiavedraz wrote: > > P.S.2: talking about well-meaning envy, and myself being some kind of > > a musician in some way, I've found this beautiful sentence: > > "Eisenberg goes on to describe musicians as "self-conscious birds" who > > have music both in their muscles and in their minds.And that must be a > > wonderful state of being. It is for this reason that I have a certain > > well-meaning envy of musicians." > > Music seems to be a good training against dementia and even seems to > protect against effects of Alzheimer's disease, OTOH a lot of disorders > are more often for artists, depressions, personality disorders, left vs > right cerebral hemisphere, IOW artists are more often left-hander, > dyslexics etc. and even while making music seems to protect against > dementia and effects of Alzheimer, musicians are more often addicted to > evil, life-endangering drugs, they are more often nerds etc., so the > resume seems to be that the pros come with contras?. IMO the only real > pro of being a musician is to have fun, when making music and many of us > "suffer" from synesthesia, so we don't need drugs to experience what > most people only could experience when taking drugs. PS: I guess not only musicians, but all artists are more awake than most non-artists. If you e.g. draw for children, you will give a drawn horse human features by drawing the eyes in front of the head. A real horse has got the eyes on the sides of the head, but usually nobody does notice this "mistake". IOW to make music, to draw a picture, to write a story, there is the need to care about nature, to reflect opinions etc.. From gnome at hawaii.rr.com Sat Feb 8 19:06:14 2014 From: gnome at hawaii.rr.com (david) Date: Sat, 08 Feb 2014 09:06:14 -1000 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <1391884433.7009.65.camel@archlinux> References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> <20140207193108.GA3525@linuxaudio.org> <1391884433.7009.65.camel@archlinux> Message-ID: <52F68026.9010009@hawaii.rr.com> On 02/08/2014 08:33 AM, Ralf Mardorf wrote: > On Sat, 2014-02-08 at 19:08 +0100, Carlos sanchiavedraz wrote: >> P.S.2: talking about well-meaning envy, and myself being some kind of >> a musician in some way, I've found this beautiful sentence: >> "Eisenberg goes on to describe musicians as "self-conscious birds" who >> have music both in their muscles and in their minds.And that must be a >> wonderful state of being. It is for this reason that I have a certain >> well-meaning envy of musicians." > > Music seems to be a good training against dementia and even seems to > protect against effects of Alzheimer's disease, OTOH a lot of disorders > are more often for artists, depressions, personality disorders, left vs > right cerebral hemisphere, IOW artists are more often left-hander, > dyslexics etc. and even while making music seems to protect against > dementia and effects of Alzheimer, musicians are more often addicted to > evil, life-endangering drugs, they are more often nerds etc., so the > resume seems to be that the pros come with contras?. IMO the only real > pro of being a musician is to have fun, when making music and many of us > "suffer" from synesthesia, so we don't need drugs to experience what > most people only could experience when taking drugs. Also, IIRC, people with strokes or other localized brain damage who were unable to process spoken words, or remember steps in a task for more than a short while, have been able to say or remember things by singing them, instead of speaking. Music is a powerful thing, deeply and thoroughly rooted in our neurology, human culture and history. It effects our minds, bodies and society. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From ralf.mardorf at rocketmail.com Sat Feb 8 19:11:18 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 20:11:18 +0100 Subject: [LAU] [OT] Phoneblocks, a modular phone In-Reply-To: <52F68026.9010009@hawaii.rr.com> References: <20140206230951.GC29528@linuxaudio.org> <20140207174646.GF19601@linuxaudio.org> <20140207193108.GA3525@linuxaudio.org> <1391884433.7009.65.camel@archlinux> <52F68026.9010009@hawaii.rr.com> Message-ID: <1391886678.7009.77.camel@archlinux> On Sat, 2014-02-08 at 09:06 -1000, david wrote: > Also, IIRC, people with strokes or other localized brain damage who were > unable to process spoken words, or remember steps in a task for more > than a short while, have been able to say or remember things by singing > them, instead of speaking. > > Music is a powerful thing, deeply and thoroughly rooted in our > neurology, human culture and history. It effects our minds, bodies and > society. You remember correctly, JFTR https://en.wikipedia.org/wiki/Scatman_John ;). From falktx at gmail.com Sat Feb 8 19:26:34 2014 From: falktx at gmail.com (Filipe Coelho) Date: Sat, 08 Feb 2014 19:26:34 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <52F684EA.6060901@gmail.com> On 02/08/2014 06:44 PM, Aur?lien Leblond wrote: > > > On 8 Feb 2014 18:16, "mark hadman" > wrote: > > > > Interesting. I can't try it with Ingen though, because its dependency > > ganv has been unbuildable for years (Arch Linux). Does it work with > > another lv2 host? > > > > I'd say no. Only Ingen provides support for CV port and the > possibility to connect plugins the modular way. > The latest Carla development code has support for LV2 CV ports (and worker). A first beta version of it (ie, 2.0-beta1) will be released next week. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Sat Feb 8 19:39:59 2014 From: gnome at hawaii.rr.com (david) Date: Sat, 08 Feb 2014 09:39:59 -1000 Subject: [LAU] Bitwig at long last...? In-Reply-To: References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> <52F501C4.70601@autostatic.com> <52F5C9DD.6000607@hawaii.rr.com> Message-ID: <52F6880F.50505@hawaii.rr.com> On 02/08/2014 04:14 AM, Brett McCoy wrote: > On Sat, Feb 8, 2014 at 1:08 AM, david wrote: > >> Lead guitarist I played with in 1973-1975 had a 1959 Gibson SG. It didn't >> feel or sound anything like my classical guitar (1973 Garcia Concert Model >> 3). In his hands, it was always high-treble with the wah-wah cranked for >> high treble; I still have a hearing loss in that range courtesy of him. Who >> in a garage band used earplugs back then? > > psshh... I still don't wear earplugs :-P So ... you're still in a garage band? ;-) -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From danstowell+lxau at gmail.com Sat Feb 8 19:40:03 2014 From: danstowell+lxau at gmail.com (Dan S) Date: Sat, 8 Feb 2014 19:40:03 +0000 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: Just to mention: someone pointed out a glitch in my code (actually, in the underlying "audiolab" module, or so it seems) - code updated: https://gist.github.com/danstowell/8872466 Dan 2014-02-07 Dan S : > Hi Matt, > > Here's a Python script which analyses a set of files in a specified > folder. They have to be file formats understood by libsndfile - which > allows for flac/wav/aiff and some others, but not mp3. It takes maybe > 5 to 10 seconds per track, but eventually it produces a plot as a PNG > file. > > https://gist.github.com/danstowell/8872466 > > (Also blogged: ) > > Best > Dan > > 2014-02-05 Matt Garman : >> >> I have a collection of FLAC files, all ripped from my CD collection >> What I would like to do is run an analysis across all the music to >> determine how the bass/lower frequencies are generally mixed. For >> example, how much content below (for example) 150 Hz is on the left >> channel versus the right channel? >> >> I'm not sure if "histogram" is the right word, but in my mind what >> I'd like to see, per-channel, is something like this: >> >> 150--125 Hz: x samples >> 125--100 Hz: y samples >> 100--80 Hz: z samples >> ... >> >> Then I can look at the two channels of a song, and if the histograms >> are approximately the same, I can assume the bass was mixed equally >> to both channels. >> >> I am a programmer, and thought it would be easy to quickly hack >> something up that would do this, but I have no experience with >> signal processing, and as I started reading about this, I quickly >> got in over my head! So I was hoping there might already exist a >> tool that has this functionality. >> >> Note that I don't need any kind of graphical output, as this needs >> to be wrapped up in some kind of batch processing script---I have >> about 11,000 files to analyze! >> >> The motivation for this is: I have a hardware DAC (digital audio >> converter) in one part of my house, and a subwoofer in another. >> There is a single coax run between the DAC and subwoofer, so I can >> only send one channel. If the overwhelming majority of my music has >> the bass mixed equally, sending only one channel isn't a problem. >> But if I choose the "L" channel to send to the sub, and much music >> has the bass mixed only to the "R" channel, then I won't be able to >> hear the low frequencies. I want to find out how often this might >> happen. >> >> Thanks, >> Matt >> >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > -- > http://www.mcld.co.uk -- http://www.mcld.co.uk From ralf.mardorf at rocketmail.com Sat Feb 8 20:02:13 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 08 Feb 2014 21:02:13 +0100 Subject: [LAU] OT: Bitwig at long last...? Message-ID: <1391889733.7009.90.camel@archlinux> On Sat, 2014-02-08 at 09:39 -1000, david wrote: > On 02/08/2014 04:14 AM, Brett McCoy wrote: > > psshh... I still don't wear earplugs :-P > > So ... you're still in a garage band? ;-) Garages as rehearsal room are seldom in Germany. When I was young I played and today the kids play in world war two NAZI bunkers in Germany. When I was young there were posters in our rehearsal room: "Feind h?rt mit denk immer dran, vertrau nicht blind dem Nebenmann!" Disgusting, broken translation: "Take care! The enemy is always there and listening, don't trust the person next to you!" but in German it's a rhyme and really funny, in an idiotic kind of way. From idragosani at gmail.com Sat Feb 8 20:17:25 2014 From: idragosani at gmail.com (Brett McCoy) Date: Sat, 8 Feb 2014 15:17:25 -0500 Subject: [LAU] Bitwig at long last...? In-Reply-To: <52F6880F.50505@hawaii.rr.com> References: <20140122172949.M56733@mh-freiburg.de> <20140122204358.M11173@mh-freiburg.de> <52F2CD4E.3030809@linuxuse.de> <1391786838.2060.38.camel@archlinux> <52F501C4.70601@autostatic.com> <52F5C9DD.6000607@hawaii.rr.com> <52F6880F.50505@hawaii.rr.com> Message-ID: On Sat, Feb 8, 2014 at 2:39 PM, david wrote: > On 02/08/2014 04:14 AM, Brett McCoy wrote: >> >> On Sat, Feb 8, 2014 at 1:08 AM, david wrote: >> >>> Lead guitarist I played with in 1973-1975 had a 1959 Gibson SG. It didn't >>> feel or sound anything like my classical guitar (1973 Garcia Concert >>> Model >>> 3). In his hands, it was always high-treble with the wah-wah cranked for >>> high treble; I still have a hearing loss in that range courtesy of him. >>> Who >>> in a garage band used earplugs back then? >> >> >> psshh... I still don't wear earplugs :-P > > > So ... you're still in a garage band? ;-) More of a basement band, but... yeah :-D -- 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 paul at linuxaudiosystems.com Sat Feb 8 22:20:52 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sat, 8 Feb 2014 17:20:52 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: On Sat, Feb 8, 2014 at 1:44 PM, Aur?lien Leblond wrote: > > On 8 Feb 2014 18:16, "mark hadman" wrote: > > > > Interesting. I can't try it with Ingen though, because its dependency > > ganv has been unbuildable for years (Arch Linux). Does it work with > > another lv2 host? > > > > I'd say no. Only Ingen provides support for CV port and the possibility to > connect plugins the modular way. > of course, you can run ingen.lv2 inside any other host. nesting/recursion, all the way down! -------------- next part -------------- An HTML attachment was scrubbed... URL: From zettberlin at linuxuse.de Sun Feb 9 10:06:15 2014 From: zettberlin at linuxuse.de (Hartmut Noack) Date: Sun, 09 Feb 2014 11:06:15 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <52F75317.2070203@linuxuse.de> Am 08.02.2014 18:13, schrieb Aur?lien Leblond: > Hello all, > > Today I just released the version 1.0.2 of the ams-lv2 plugins. > ams-lv2 is a port of alsa-modular-synth in the lv2 format Thanks a lot for that! I still consider AMS one of the most remarkable softsynths for Linux and I am always grateful, that people carry on the good work Mathias Nagorni has started back in the day. > to create modular synth primarily with Ingen. I did not see Ingen run stable on my Fedora or my Kubuntu installations. Most of the time I was not even able to build and/or to start the binaries from the repos. How is the status: given, you have a recent stable relase of Feoda or *buntu: does Ingen work for you as, say AMS does? best regards HZN > Source code and more information available on the website: > http://objectivewave.wordpress.com/ams-lv2/ > > To celebrate this release, I created two videos: > > The first is a demo of 3 synths created with ams-lv2 and Ingen (a > bass, a lead and a pad) soloed and inside a mix: > http://www.youtube.com/watch?v=LWfF71NerkQ > > I created as well this first tutorial on the basics of ams-lv2: > http://www.youtube.com/watch?v=IuQZajaSw6M > > > I would as well like to take advantage to thank a lot of people > who helped with this project: - David for his patience to answer my > 1000+ questions and for all his work on lv2 and Ingen - Robin for > creating the sisco.lv2 plugins, they are featured in the demo and > they are a great help when creating modular synths - Harry for the > dials used in the GUI of ams-lv2 - and finally Fons for creating > the great alsa-modular-synth in the first place - the whole Linux > Audio community in general > > I really hope you enjoy these plugins and find them useful > somehow! > > Aur?lien _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From atte at youmail.dk Sun Feb 9 10:40:06 2014 From: atte at youmail.dk (Atte) Date: Sun, 09 Feb 2014 11:40:06 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <52F75B06.6050405@youmail.dk> On 02/08/2014 06:13 PM, Aur?lien Leblond wrote: > Hello all, > > Today I just released the version 1.0.2 of the ams-lv2 plugins. Please don't kill me, but I would personally much prefer a linux-native VSTi version... -- Atte http://atte.dk http://modlys.dk From jonetsu at teksavvy.com Sun Feb 9 11:19:05 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Sun, 9 Feb 2014 06:19:05 -0500 Subject: [LAU] OT: Bitwig at long last...? In-Reply-To: <1391889733.7009.90.camel@archlinux> References: <1391889733.7009.90.camel@archlinux> Message-ID: <20140209061905.786c4720@mistral> On Sat, 08 Feb 2014 21:02:13 +0100, Ralf Mardorf wrote : > On Sat, 2014-02-08 at 09:39 -1000, david wrote: > > On 02/08/2014 04:14 AM, Brett McCoy wrote: > > > psshh... I still don't wear earplugs :-P > > > > So ... you're still in a garage band? ;-) > > Garages as rehearsal room are seldom in Germany. > > When I was young I played and today the kids play in world war two > NAZI bunkers in Germany. > > When I was young there were posters in our rehearsal room: > > "Feind h?rt mit denk immer dran, vertrau nicht blind dem Nebenmann!" When I was young I had most albums by Amon D??l II and Can. And listened to them too :) From blablack at gmail.com Sun Feb 9 14:24:28 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sun, 9 Feb 2014 14:24:28 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial Message-ID: >> Hello all, >> >> Today I just released the version 1.0.2 of the ams-lv2 plugins. > > Please don't kill me, but I would personally much prefer a linux-native > VSTi version... *sight* why? There are only 3 hosts that I know of that can (or will be able to) combine plugins in a modular way: ams, Ingen and Carla. Ingen and Carla both support LV2, and can be loaded as instrument (at least for Ingen, not sure about Carla), what would linux-native VSTi bring? Aur?lien From blablack at gmail.com Sun Feb 9 14:34:15 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sun, 9 Feb 2014 14:34:15 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial Message-ID: >> to create modular synth primarily with Ingen. > > I did not see Ingen run stable on my Fedora or my Kubuntu > installations. Most of the time I was not even able to build and/or to > start the binaries from the repos. > > How is the status: given, you have a recent stable relase of Feoda or > *buntu: does Ingen work for you as, say AMS does? Yes, I'm running the latest SVN in Ubuntu and it works fine for me. I have a Ubuntu repo here with Ingen available: http://objectivewave.wordpress.com/ubuntu-repository/ I get a random crash from time to time - if I have a constant scenario to reproduce I create a bug report for David and I learn not to repeat the same scenario until he fixes the issue... Pretty much the same experience I had with AMS :) I guess if you are not especially using LV2 plugins, there wouldn't be any advantages using ams-lv2/Ingen over AMS. I started this project of porting this ams modules because of a few LV2 plugins I liked using in my synths and AMS doesn't support LV2 (I like having everything in one host as much as I can). Aur?lien From paul at linuxaudiosystems.com Sun Feb 9 15:31:27 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 9 Feb 2014 10:31:27 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F75B06.6050405@youmail.dk> References: <52F75B06.6050405@youmail.dk> Message-ID: On Sun, Feb 9, 2014 at 5:40 AM, Atte wrote: > On 02/08/2014 06:13 PM, Aur?lien Leblond wrote: > >> Hello all, >> >> Today I just released the version 1.0.2 of the ams-lv2 plugins. >> > > Please don't kill me, but I would personally much prefer a linux-native > VSTi version... > How would the VST spec allow for control voltage? (hint: it won't) -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Sun Feb 9 18:27:19 2014 From: atte at youmail.dk (Atte) Date: Sun, 09 Feb 2014 19:27:19 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F75B06.6050405@youmail.dk> Message-ID: <52F7C887.8070100@youmail.dk> On 02/09/2014 04:31 PM, Paul Davis wrote: > How would the VST spec allow for control voltage? > > (hint: it won't) Not sure what you're talking about. AMS when ran standalone is (in my setup) controlled by midi. That's what I thought might be possible with AMS as VST. But if you say so... -- Atte http://atte.dk http://modlys.dk From atte at youmail.dk Sun Feb 9 18:30:37 2014 From: atte at youmail.dk (Atte) Date: Sun, 09 Feb 2014 19:30:37 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <52F7C94D.4000009@youmail.dk> On 02/09/2014 03:24 PM, Aur?lien Leblond wrote: >> Please don't kill me, but I would personally much prefer a linux-native >> VSTi version... > > *sight* why? Because it would be great to run AMS as a vst plugin in hosts that doesn't support LV2, radium, renoise and bitwig to name a few. I'm sure LV2 is superior in many ways, but VST still is a more widespread (and thus more pragmatic) standard. -- Atte http://atte.dk http://modlys.dk From blablack at gmail.com Sun Feb 9 18:44:54 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sun, 9 Feb 2014 18:44:54 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F7C94D.4000009@youmail.dk> References: <52F7C94D.4000009@youmail.dk> Message-ID: On 9 Feb 2014 18:30, "Atte" wrote: > > On 02/09/2014 03:24 PM, Aur?lien Leblond wrote: > >>> Please don't kill me, but I would personally much prefer a linux-native >>> VSTi version... >> >> >> *sight* why? > > > Because it would be great to run AMS as a vst plugin in hosts that doesn't support LV2, radium, renoise and bitwig to name a few. > See what you want is a bit more complicated. The ams-lv2 plugins require a host that can connect plugins together in a modular way, like AMS itself or Ingen do. >From what I remember, I think Bitwig may have that at some stage, but Renoise doesn't. What you are really asking for is for Ingen to be ported as a VSTi plugin :-) Aur?lien -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Sun Feb 9 18:49:46 2014 From: atte at youmail.dk (Atte) Date: Sun, 09 Feb 2014 19:49:46 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7C94D.4000009@youmail.dk> Message-ID: <52F7CDCA.3070000@youmail.dk> On 02/09/2014 07:44 PM, Aur?lien Leblond wrote: > What you are really asking for is for Ingen to be ported as a VSTi > plugin :-) No, no, please not ingen. Just AMS... -- Atte http://atte.dk http://modlys.dk From blablack at gmail.com Sun Feb 9 18:57:11 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Sun, 9 Feb 2014 18:57:11 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F7CDCA.3070000@youmail.dk> References: <52F7C94D.4000009@youmail.dk> <52F7CDCA.3070000@youmail.dk> Message-ID: On 9 Feb 2014 18:49, "Atte" wrote: > > On 02/09/2014 07:44 PM, Aur?lien Leblond wrote: > >> What you are really asking for is for Ingen to be ported as a VSTi >> plugin :-) > > > No, no, please not ingen. Just AMS... You do know that AMS and Ingen are in their concept very similar, right? They are hosts that connect plugins (LADSPA for AMS and LV2 for Ingen) in a modular way. Outside the format they support, the other difference is the internal modules provided, and that's where ams-lv2 comes in for Ingen... Ha, you remind me of me a few months ago when I thought "it would be great to have AMS in the LV2 format" :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob at rektau.ukfsn.org Sun Feb 9 19:04:32 2014 From: rob at rektau.ukfsn.org (rob) Date: Sun, 09 Feb 2014 19:04:32 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <52F7D140.6080902@rektau.ukfsn.org> On 08/02/14 18:44, Aur?lien Leblond wrote: > > On 8 Feb 2014 18:16, "mark hadman" > wrote: > > > > Interesting. I can't try it with Ingen though, because its dependency > > ganv has been unbuildable for years (Arch Linux). Does it work with > > another lv2 host? > > > From http://dev.drobilla.net/wiki/IngenInstallation "The easiest way to build Ingen from SVN is to build the entire ? http://svn.drobilla.net/lad repository (since there are dependencies between the various projects)." build directory includes: build ingen lilv patchage README sratom wscript eugene INSTALL machina plugins serd suil ganv jalv naub raul sord waf rob From paul at linuxaudiosystems.com Sun Feb 9 19:37:33 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 9 Feb 2014 14:37:33 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F7C887.8070100@youmail.dk> References: <52F75B06.6050405@youmail.dk> <52F7C887.8070100@youmail.dk> Message-ID: The point of Aurelien's work is to create a set of plugins that can be interconnected together in arbitrary ways. Those connections consist of two types of data: audio and control voltage. The control voltage part is important. VST has no concept of "control voltage". Even if you find a host that allows arbitrary connections between plugins, it will not have any way to support control voltage data connections. This will completely break almost every idea at the heart of AMS and Aurelien's LV2 plugins. VST, like most plugin APIs, only really knows about audio and MIDI. AMS does not fit into that paradigm, in the same way that no self-respecting hardware modular synthesizer lets you patch MIDI from module. They may support MIDI I/O, but the main data that flows around between modules is control voltage. On Sun, Feb 9, 2014 at 1:27 PM, Atte wrote: > On 02/09/2014 04:31 PM, Paul Davis wrote: > > How would the VST spec allow for control voltage? >> >> (hint: it won't) >> > > Not sure what you're talking about. > > AMS when ran standalone is (in my setup) controlled by midi. That's what I > thought might be possible with AMS as VST. But if you say so... > > > -- > Atte > > http://atte.dk http://modlys.dk > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sun Feb 9 19:49:11 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 9 Feb 2014 19:49:11 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F7D140.6080902@rektau.ukfsn.org> References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: <20140209194911.GA10498@linuxaudio.org> On Sun, Feb 09, 2014 at 07:04:32PM +0000, rob wrote: > "The easiest way to build Ingen from SVN is to build the entire ? > http://svn.drobilla.net/lad repository (since there are dependencies > between the various projects)." > > build directory includes: > > build ingen lilv patchage README sratom wscript > eugene INSTALL machina plugins serd suil > ganv jalv naub raul sord waf Tried that yesterday evening. The configure step for ingen complains about a missing boost/* file. But I do have boost installed, including the development parts. 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 looplog at gmail.com Sun Feb 9 19:52:15 2014 From: looplog at gmail.com (michael noble) Date: Mon, 10 Feb 2014 04:52:15 +0900 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F7D140.6080902@rektau.ukfsn.org> References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: On Mon, Feb 10, 2014 at 4:04 AM, rob wrote: > From http://dev.drobilla.net/wiki/IngenInstallation > > "The easiest way to build Ingen from SVN is to build the entire > http://svn.drobilla.net/lad repository (since there are dependencies > between the various projects)." > > build directory includes: > > build ingen lilv patchage README sratom wscript > eugene INSTALL machina plugins serd suil > ganv jalv naub raul sord waf > Unfortunately there is wide gulf between the "easiest way to build" applications and the easiest way to install and manage them. I guess Mark's point is that it has been impossible to build ingen from the AUR scripts for some time, which is sadly true. Trying to mix svn of some of drobilla libraries with AUR or Arch packages of those libraries, or even the full svn tree with apps that depend on the official packages, quickly results in a mess. -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Sun Feb 9 19:53:07 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 9 Feb 2014 14:53:07 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <20140209194911.GA10498@linuxaudio.org> References: <52F7D140.6080902@rektau.ukfsn.org> <20140209194911.GA10498@linuxaudio.org> Message-ID: On Sun, Feb 9, 2014 at 2:49 PM, Fons Adriaensen wrote: > On Sun, Feb 09, 2014 at 07:04:32PM +0000, rob wrote: > > > "The easiest way to build Ingen from SVN is to build the entire > > http://svn.drobilla.net/lad repository (since there are dependencies > > between the various projects)." > > > > build directory includes: > > > > build ingen lilv patchage README sratom wscript > > eugene INSTALL machina plugins serd suil > > ganv jalv naub raul sord waf > > Tried that yesterday evening. The configure step for ingen > complains about a missing boost/* file. But I do have boost > installed, including the development parts. > where is boost installed? it does not come with a pkg-config file, so if it is not discoverable "by default" it always causes problems. -------------- next part -------------- An HTML attachment was scrubbed... URL: From matthew.garman at gmail.com Sun Feb 9 21:08:41 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Sun, 9 Feb 2014 15:08:41 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? Message-ID: Hi, I started the thread "How is the bass mixed? Per-channel frequency analysis? Histogram?" a few weeks ago. One suggestion that came up a few times was to simply downmix the bass to mono before sending it out. How would I go about this? Currently, this system doesn't have any audio. What I intend to do is send audio out over USB to a DAC. Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" to mono. Ideally the frequency should be configurable, but I think about 150Hz or below is a good start. If possible, I'd like to keep things simple, and have only ALSA and MPD running. As far as I can tell, I don't think I can do this at the MPD level. I saw an example where I can easily downmix all output to mono. I'm not opposed to that, but it would be nice to selectively downmix only the lower (subwoofer) frequencies. Thanks again! Matt From fons at linuxaudio.org Sun Feb 9 21:27:05 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 9 Feb 2014 21:27:05 +0000 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: Message-ID: <20140209212705.GB10498@linuxaudio.org> On Sun, Feb 09, 2014 at 03:08:41PM -0600, Matt Garman wrote: > I started the thread "How is the bass mixed? Per-channel frequency > analysis? Histogram?" a few weeks ago. One suggestion that came up a > few times was to simply downmix the bass to mono before sending it > out. > > How would I go about this? Currently, this system doesn't have any > audio. What I intend to do is send audio out over USB to a DAC. > Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" > to mono. Ideally the frequency should be configurable, but I think > about 150Hz or below is a good start. > > If possible, I'd like to keep things simple, and have only ALSA and > MPD running. As far as I can tell, I don't think I can do this at the > MPD level. You really need to provide a bit more information here. If your sub is an active one (with a built-in amplifier) it will have the required filter. And probably it will have two inputs which it will add to mono. So you just connect the two outputs of your DAC. Many active subs also provide outputs for the amplifier of the main speakers, which will have the bass (which is used by the sub) filtered out. In that case just connect your main amplifier or active speakers there. If your equipment is different, then at least provide a clear and complete descriptn, nobody will be able to help without that. Note that it is normally more important to remove the bass signal for the main speakers than to remove the mid/high part for the sub. 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 matthew.garman at gmail.com Sun Feb 9 21:33:02 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Sun, 9 Feb 2014 15:33:02 -0600 Subject: [LAU] How is the bass mixed? Per-channel frequency analysis? Histogram? In-Reply-To: References: <20140205153636.GA29740@septictank.raw-sewage.fake> Message-ID: Hi Dan, Thank you for taking the time to put that together! Looks like I have a fair amount of learning to do before I can make sense of exactly what's going on (I mean, I get the gist at a very high level, but don't understand the details enough to hack the code). I ran it on a few tracks that I thought would be interesting. Clearly this list has a number of folks with mixing experience and/or knowledge, maybe they can comment on what I'm seeing: 1. Leo Kottke "The Driving of the Year Nail" - first track from 6- and 12-String Guitar. Looks like there's a definite bias towards the left channel. http://raw-sewage.net/images/stereoscope/kottke.png I would have expected both channels to be perfectly equal, since this is exactly one instrument. (I have no mixing or studio experience, keep in mind, just a guy who likes music. So this is what my uninformed intuition says.) 2. The Beatles "While My Guitar Gently Weeps" - from The While Album. http://raw-sewage.net/images/stereoscope/beatles.png I picked this since there was a mention of Beatles albums having the bass mixed only to one side. If I could hack the code to do a low pass filter, this might be an interesting track. But, it appears to me there's a definite bias for the lowest frequencies to be panned to the right side. 3. Above & Beyond "Let Go" - from Anjunabeats Vol 8. http://raw-sewage.net/images/stereoscope/above_beyond.png Just curious what some electronic music looked like... This could well be a mono mix? 4. Beethoven "Moonlight Sonata" - from a collection of Beethoven's piano sonatas (not sure who the performer is or label, too lazy to go pull the CD) http://raw-sewage.net/images/stereoscope/beethoven.png This graph looks kind of weird. Is there really that little high-frequency information? Anyway, thanks again Dan! I'll keep playing with the tool and trying to learn to see if I can come up with any more interesting info. -Matt On Fri, Feb 7, 2014 at 3:43 PM, Dan S wrote: > Hi Matt, > > Here's a Python script which analyses a set of files in a specified > folder. They have to be file formats understood by libsndfile - which > allows for flac/wav/aiff and some others, but not mp3. It takes maybe > 5 to 10 seconds per track, but eventually it produces a plot as a PNG > file. > > https://gist.github.com/danstowell/8872466 > > (Also blogged: ) > > Best > Dan > > 2014-02-05 Matt Garman : >> >> I have a collection of FLAC files, all ripped from my CD collection >> What I would like to do is run an analysis across all the music to >> determine how the bass/lower frequencies are generally mixed. For >> example, how much content below (for example) 150 Hz is on the left >> channel versus the right channel? >> >> I'm not sure if "histogram" is the right word, but in my mind what >> I'd like to see, per-channel, is something like this: >> >> 150--125 Hz: x samples >> 125--100 Hz: y samples >> 100--80 Hz: z samples >> ... >> >> Then I can look at the two channels of a song, and if the histograms >> are approximately the same, I can assume the bass was mixed equally >> to both channels. >> >> I am a programmer, and thought it would be easy to quickly hack >> something up that would do this, but I have no experience with >> signal processing, and as I started reading about this, I quickly >> got in over my head! So I was hoping there might already exist a >> tool that has this functionality. >> >> Note that I don't need any kind of graphical output, as this needs >> to be wrapped up in some kind of batch processing script---I have >> about 11,000 files to analyze! >> >> The motivation for this is: I have a hardware DAC (digital audio >> converter) in one part of my house, and a subwoofer in another. >> There is a single coax run between the DAC and subwoofer, so I can >> only send one channel. If the overwhelming majority of my music has >> the bass mixed equally, sending only one channel isn't a problem. >> But if I choose the "L" channel to send to the sub, and much music >> has the bass mixed only to the "R" channel, then I won't be able to >> hear the low frequencies. I want to find out how often this might >> happen. >> >> Thanks, >> Matt >> >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > -- > http://www.mcld.co.uk From robin at gareus.org Sun Feb 9 23:03:03 2014 From: robin at gareus.org (Robin Gareus) Date: Mon, 10 Feb 2014 00:03:03 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: <52F80927.8060701@gareus.org> On 02/09/2014 08:52 PM, michael noble wrote: > On Mon, Feb 10, 2014 at 4:04 AM, rob wrote: >> "The easiest way to build Ingen from SVN is to build the entire >> http://svn.drobilla.net/lad repository (since there are dependencies >> between the various projects)." [..] > Trying to mix svn of some > of drobilla libraries with AUR or Arch packages of those libraries, or even > the full svn tree with apps that depend on the official packages, quickly > results in a mess. > There's no need to mix. -------- export PKG_CONFIG_PATH=$HOME/tmp/lv2/lib/pkgconfig/ export LD_LIBRARY_PATH=$HOME/tmp/lv2/lib svn co http://lv2plug.in/repo/trunk lv2 cd lv2 ./waf configure --prefix=$HOME/tmp/lv2/ || exit ./waf || exit ./waf install cd .. svn co http://svn.drobilla.net/lad/trunk lad cd lad ./waf configure --prefix=$HOME/tmp/lv2/ || exit ./waf || exit ./waf install exec ~/tmp/lv2/bin/ingen -------- From matthew.garman at gmail.com Sun Feb 9 23:51:54 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Sun, 9 Feb 2014 17:51:54 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <20140209212705.GB10498@linuxaudio.org> References: <20140209212705.GB10498@linuxaudio.org> Message-ID: On Sun, Feb 9, 2014 at 3:27 PM, Fons Adriaensen wrote: > On Sun, Feb 09, 2014 at 03:08:41PM -0600, Matt Garman wrote: >> How would I go about this? Currently, this system doesn't have any >> audio. What I intend to do is send audio out over USB to a DAC. >> Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" >> to mono. Ideally the frequency should be configurable, but I think >> about 150Hz or below is a good start. > > You really need to provide a bit more information here. Sure, no problem. What I have is: (1) Linux server with a media collection in a basement network closet (2) A powered subwoofer on the main floor, with only a single coax run between the sub and network closet in the basement (3) In-ceiling speakers on the main floor, all wires terminating in same network closet So the fundamental problem is that there's no physical locality between the server and the speakers. What I was thinking was to use a USB DAC that has dual outputs. One output would go to a power amp, which in turn would go to a speaker selector (so I can power multiple sets of speakers). The DAC's other output needs to go to the sub. But there's the problem: I only have a single coax run, so I can only send one channel. This means, either pick one and hope most music doesn't pan the bass (the point of the other thread I started), or come up with some way to mixdown the signal to mono so I can pass it over the single conductor. I thought about doing this in hardware, see this thread: http://www.audiocircle.com/index.php?topic=123314.0 But it occurred to me I could possibly spare some expense by mixing down the signal at the server level. In reality, based on the previous thread, I think I could probably send just one channel and be OK most of the time. But if I can easily get it 100% of the time, why not? :) Note that in this solution, I'm sending the full-range signal to both the in-ceiling speakers and the sub. The sub has a builtin crossover, and the speakers can handle the full signal. As a side note, if you read the AudioCircle thread above, you'll see I already have an AVR that can send a bass signal over coax... my goal is to get rid of the AVR though, as I have more speakers than it has outputs (would rather not use the AVR and speaker selector), plus I want to move control off the AVR and onto the Linux server (mpd). Let me know if anything is unclear or if you have any other ideas! Thanks, Matt From federicogalland at gmail.com Mon Feb 10 00:52:54 2014 From: federicogalland at gmail.com (F Tux) Date: Sun, 9 Feb 2014 22:52:54 -0200 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> Message-ID: I know you said you wanted to keep it simple, but I'd like to point out that I have a similar setup running at home. I listen to all my music with a guitar amplifier for the treble and a bass amplifier for the bass. Obviously, I have to downmix the signal to mono (I don't miss the stereo sound, I just mind hearing every single instrument as clear as possible). The signal coming from my music player is routed into 2 separated EQs which I use to filter the high frequencies from the bass amp and the low from the guitar amp. You could use highpass and lowpass filters for this, but using EQs also serves the purpose of removing troublesome high harmonics in the guitar amp, and also, equalizing the output. All this I do with the calf plugins (never bothered to try any other, because they've suited me good enough so far) but you can also do that with any gui-less plugin. If I had to downmix the bass I'd get the player's output through a lowpass and then downmix it by connecting left and right outputs of the filter to the channel that's output to the subwoofer. At the same time, you can route the untouched signal to the treble speakers adding (possibly adding some plugin in the middle to compensate the latency difference). I know this is a bit more involved than you originally planned but I doubt there is a simpler way with the ALSA tools available, and this way you will end up with a completely customized setup. Some of the stuff I described is only based on my trial and error experience. The enlightened people on the list will surely know how to polish this method though I use what I have described everyday and it just works. Sorry for the long text, have a good one! On Sun, Feb 9, 2014 at 8:51 PM, Matt Garman wrote: > On Sun, Feb 9, 2014 at 3:27 PM, Fons Adriaensen > wrote: > > On Sun, Feb 09, 2014 at 03:08:41PM -0600, Matt Garman wrote: > >> How would I go about this? Currently, this system doesn't have any > >> audio. What I intend to do is send audio out over USB to a DAC. > >> Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" > >> to mono. Ideally the frequency should be configurable, but I think > >> about 150Hz or below is a good start. > > > > You really need to provide a bit more information here. > > Sure, no problem. What I have is: > (1) Linux server with a media collection in a basement network closet > (2) A powered subwoofer on the main floor, with only a single coax > run between the sub and network closet in the basement > (3) In-ceiling speakers on the main floor, all wires terminating > in same network closet > > So the fundamental problem is that there's no physical locality > between the server and the speakers. What I was thinking was to use a > USB DAC that has dual outputs. One output would go to a power amp, > which in turn would go to a speaker selector (so I can power multiple > sets of speakers). > > The DAC's other output needs to go to the sub. But there's the > problem: I only have a single coax run, so I can only send one > channel. This means, either pick one and hope most music doesn't pan > the bass (the point of the other thread I started), or come up with > some way to mixdown the signal to mono so I can pass it over the > single conductor. I thought about doing this in hardware, see this > thread: > http://www.audiocircle.com/index.php?topic=123314.0 > But it occurred to me I could possibly spare some expense by mixing > down the signal at the server level. > > In reality, based on the previous thread, I think I could probably > send just one channel and be OK most of the time. But if I can easily > get it 100% of the time, why not? :) > > Note that in this solution, I'm sending the full-range signal to both > the in-ceiling speakers and the sub. The sub has a builtin crossover, > and the speakers can handle the full signal. > > As a side note, if you read the AudioCircle thread above, you'll see I > already have an AVR that can send a bass signal over coax... my goal > is to get rid of the AVR though, as I have more speakers than it has > outputs (would rather not use the AVR and speaker selector), plus I > want to move control off the AVR and onto the Linux server (mpd). > > Let me know if anything is unclear or if you have any other ideas! > > Thanks, > Matt > _______________________________________________ > 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 federicogalland at gmail.com Mon Feb 10 00:59:19 2014 From: federicogalland at gmail.com (F Tux) Date: Sun, 9 Feb 2014 22:59:19 -0200 Subject: [LAU] Pulseaudio and Jack In-Reply-To: <52F52F75.2020106@revolwear.com> References: <52F52F75.2020106@revolwear.com> Message-ID: Option 1. No pulse. For youtube, I copy the link, and then press alt-y which is bind to the following script (depends on xclip, mplayer and youtube-dl): #!/bin/sh > > URL=`xclip -o` > > RAWURL=`youtube-dl -gf 43/18/34 --cookies /tmp/ytdl-cookie.txt "$URL"` > > mplayer -cache-min 50 -cookies -cookies-file /tmp/ytdl-cookie.txt "$RAWURL" > > rm /tmp/ytdl-cookie.txt > > exit 0 > On Fri, Feb 7, 2014 at 4:09 PM, Max wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Which of the suggested option is the one people here on the list favour? > > http://jackaudio.org/pulseaudio_and_jack > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.14 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ > > iEYEARECAAYFAlL1L3UACgkQ3EB7kzgMM6Ln1ACfVyT8RXVcBzHiIkyQJU9LWBWK > 4p4Anj08pRcsv5ytIxvtrebWhbkQulGA > =JPXo > -----END PGP SIGNATURE----- > _______________________________________________ > 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 Mon Feb 10 01:39:48 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 10 Feb 2014 01:39:48 +0000 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> Message-ID: <20140210013948.GC10498@linuxaudio.org> On Sun, Feb 09, 2014 at 05:51:54PM -0600, Matt Garman wrote: > So the fundamental problem is that there's no physical locality > between the server and the speakers. What I was thinking was to use a > USB DAC that has dual outputs. One output would go to a power amp, > which in turn would go to a speaker selector (so I can power multiple > sets of speakers). > > The DAC's other output needs to go to the sub. But there's the > problem: I only have a single coax run, so I can only send one > channel. I can't make sense of this. One output of the DAC goes to the power amp (for the main speakers) The other output goes to the sub. So you won't have stereo. In that case, just send the mono signal (L+R) to both channels of the DAC. If you don't do that, you'll just be listening to one channel. And then I really wonder why you bother about the bass - you'll be missing the entire second channel anyway. And why do you use *coax* cable between the DAC and the sub ? A normal screened cable will do... If you're a bit handy with a soldering iron, all it takes to mix the DAC outputs to mono would be two resistors and some short pieces of wire and connectors. 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 gnome at hawaii.rr.com Mon Feb 10 01:40:27 2014 From: gnome at hawaii.rr.com (david) Date: Sun, 09 Feb 2014 15:40:27 -1000 Subject: [LAU] Pulseaudio and Jack In-Reply-To: References: <52F52F75.2020106@revolwear.com> Message-ID: <52F82E0B.6060600@hawaii.rr.com> I use the FlashGot or Video Download Helper extensions in Firefox. Youtube-dl I use for longer downloads. On 02/09/2014 02:59 PM, F Tux wrote: > Option 1. > > No pulse. For youtube, I copy the link, and then press alt-y which is > bind to the following script (depends on xclip, mplayer and youtube-dl): > > #!/bin/sh > > URL=`xclip -o` > > RAWURL=`youtube-dl -gf 43/18/34 --cookies /tmp/ytdl-cookie.txt "$URL"` > > mplayer -cache-min 50 -cookies -cookies-file /tmp/ytdl-cookie.txt > "$RAWURL" > > rm /tmp/ytdl-cookie.txt > > exit 0 > > > > On Fri, Feb 7, 2014 at 4:09 PM, Max > wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Which of the suggested option is the one people here on the list favour? > > http://jackaudio.org/pulseaudio_and_jack > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.14 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ > > iEYEARECAAYFAlL1L3UACgkQ3EB7kzgMM6Ln1ACfVyT8RXVcBzHiIkyQJU9LWBWK > 4p4Anj08pRcsv5ytIxvtrebWhbkQulGA > =JPXo > -----END PGP SIGNATURE----- -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From althompson58 at gmail.com Mon Feb 10 01:44:09 2014 From: althompson58 at gmail.com (Al) Date: Sun, 09 Feb 2014 20:44:09 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial Message-ID: CV is different than MIDI. Atte wrote: >On 02/09/2014 04:31 PM, Paul Davis wrote: > >> How would the VST spec allow for control voltage? >> >> (hint: it won't) > >Not sure what you're talking about. > >AMS when ran standalone is (in my setup) controlled by midi. That's what >I thought might be possible with AMS as VST. But if you say so... > >-- >Atte > >http://atte.dk http://modlys.dk >_______________________________________________ >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 Mon Feb 10 07:51:25 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 10 Feb 2014 08:51:25 +0100 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <20140210013948.GC10498@linuxaudio.org> References: <20140209212705.GB10498@linuxaudio.org> <20140210013948.GC10498@linuxaudio.org> Message-ID: <1392018685.762.45.camel@archlinux> On Mon, 2014-02-10 at 01:39 +0000, Fons Adriaensen wrote: > If you're a bit handy with a soldering iron, all it takes to > mix the DAC outputs to mono would be two resistors and some > short pieces of wire and connectors. If the OP doesn't use tube amplifiers it should look like this: http://1.bp.blogspot.com/_BGB-FuSV33o/S7qQlfl_1aI/AAAAAAAAAHY/QO2t7rZP5N0/s400/stereotomono.gif When using tube amplifiers you might want to add transformers. From ralf.mardorf at rocketmail.com Mon Feb 10 08:08:33 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 10 Feb 2014 09:08:33 +0100 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <1392018685.762.45.camel@archlinux> References: <20140209212705.GB10498@linuxaudio.org> <20140210013948.GC10498@linuxaudio.org> <1392018685.762.45.camel@archlinux> Message-ID: <1392019713.762.47.camel@archlinux> On Mon, 2014-02-10 at 08:51 +0100, Ralf Mardorf wrote: > On Mon, 2014-02-10 at 01:39 +0000, Fons Adriaensen wrote: > > If you're a bit handy with a soldering iron, all it takes to > > mix the DAC outputs to mono would be two resistors and some > > short pieces of wire and connectors. > > If the OP doesn't use tube amplifiers it should look like this: > > http://1.bp.blogspot.com/_BGB-FuSV33o/S7qQlfl_1aI/AAAAAAAAAHY/QO2t7rZP5N0/s400/stereotomono.gif > > When using tube amplifiers you might want to add transformers. PS: "Thanks for the useful post. I have one suggestion, though: I think it might be better to use 1K resistors rather than 10K. The input impedance of a typical amp should be in the region of 20K. At audio frequencies (where impedance-matching doesn't matter) it is best if the input impedance is at least x10 the output impedance. Assuming an output impedance in the range 100-600 ohms (plausible for an MP3 player headphone output), that would keep the total output impedance below 2K which should result in more signal voltage across the amp's input." - http://marcusmorris.blogspot.de/2010/04/stereo-to-mono-converter-how-to-do-it.html From looplog at gmail.com Mon Feb 10 09:47:18 2014 From: looplog at gmail.com (michael noble) Date: Mon, 10 Feb 2014 18:47:18 +0900 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F80927.8060701@gareus.org> References: <52F7D140.6080902@rektau.ukfsn.org> <52F80927.8060701@gareus.org> Message-ID: On Mon, Feb 10, 2014 at 8:03 AM, Robin Gareus wrote: > There's no need to mix. Thanks for the tip! That looks eminently useful. -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Mon Feb 10 11:22:46 2014 From: atte at youmail.dk (Atte) Date: Mon, 10 Feb 2014 12:22:46 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7C94D.4000009@youmail.dk> <52F7CDCA.3070000@youmail.dk> Message-ID: <52F8B686.60700@youmail.dk> On 02/09/2014 07:57 PM, Aur?lien Leblond wrote: > You do know that AMS and Ingen are in their concept very similar, right? Yes. Two important differences, though: 1) AMS has build-in modules, so is fully self-contained. 2) AMS is stable (at least compared to ingen). I've had tons of fun with AMS, made some really nice sounds with it, but without a plugin it just never was used as much in real life as it could have. Ingen (originally om), on the other hand, has always been a drag to build, keep up with and get stable. It might be me, but I really, really tried. VSTi is simply the first choice of format to support for larger players (energyxt, renoise and now bitwig) and some smaller (I love radium), maybe out of habit, maybe because they can reuse code from their windows implementation. Having AMS as linux native VSTi would be absolutely awesome! I'm not even sure money could buy you a more powerful synth on linux. Sure there's Loomer Aspect (bought that) and some minor ones, but AMS is the bomb! -- Atte http://atte.dk http://modlys.dk From atte at youmail.dk Mon Feb 10 11:32:00 2014 From: atte at youmail.dk (Atte) Date: Mon, 10 Feb 2014 12:32:00 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F75B06.6050405@youmail.dk> <52F7C887.8070100@youmail.dk> Message-ID: <52F8B8B0.3080808@youmail.dk> On 02/09/2014 08:37 PM, Paul Davis wrote: > VST, like most plugin APIs, only really knows about audio and MIDI. AMS > does not fit into that paradigm, in the same way that no self-respecting > hardware modular synthesizer lets you patch MIDI from module. They may > support MIDI I/O, but the main data that flows around between modules is > control voltage. Hmmm. So it's not possible to wrap AMS (with it's self contained modules and functionality intact) in VSTi? I'd be surprised, but I never coded a VSTi, so how am I? I just find it really hard to believe, since AMS on my box is a self-contained piece of software that connects with the outside world through a gui (like VSTi), midi I/O (like VSTi) and audio (like VSTi). But unfortunately that doesn't seem to be possible... It seems we are talking about two things: I'm talking about the "whole of AMS, canvas + buildin modules, but nothing else", it seems to me you're talking about "individual modules as VSTi and then try to connect them in my host". Or? -- Atte http://atte.dk http://modlys.dk From shanipribadi at gmx.net Mon Feb 10 12:11:23 2014 From: shanipribadi at gmx.net (Shani Hadiyanto Pribadi) Date: Mon, 10 Feb 2014 19:11:23 +0700 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? Message-ID: On 10 February 2014 06:51, Matt Garman wrote: > Sure, no problem. What I have is: > (1) Linux server with a media collection in a basement network closet > (2) A powered subwoofer on the main floor, with only a single coax > run between the sub and network closet in the basement > (3) In-ceiling speakers on the main floor, all wires terminating > in same network closet Are the in-ceiling speakers connected in pairs? If they are then you need 3 channels, L,R,M. where M = L+R. If you have a soundcard with at least 3 channels and insist on using alsa output then you can use http://alsa.opensrc.org/Low-pass_filter_for_subwoofer_channel_%28HOWTO%29 to upmix from 2 channels to 3 channels in ALSA. (you can ignore the part about low pass). Or you could instead use the jack mpd output plugins and just connect the channels like MPD_out_L -> HW_L -> HW_M MPD_out_R -> HW_R -> HW_M There is also jackminimix for a OSC controlled mixer. I think ecasound can also be used for that purpose. > Note that in this solution, I'm sending the full-range signal to both > the in-ceiling speakers and the sub. The sub has a builtin crossover, > and the speakers can handle the full signal. > > As a side note, if you read the AudioCircle thread above, you'll see I > already have an AVR that can send a bass signal over coax... my goal > is to get rid of the AVR though, as I have more speakers than it has > outputs (would rather not use the AVR and speaker selector), plus I > want to move control off the AVR and onto the Linux server (mpd). Now this is where I lost you, how many speakers do you have exactly, and how many amps for them? Doesn't it mean that you need as many amps for each speakers you have if you ditch the speaker selector? If indeed you want to control it like that then you also need a soundcard with as many output channels as the number of power amps channels for the speakers. From markhadman at googlemail.com Mon Feb 10 12:26:41 2014 From: markhadman at googlemail.com (mark hadman) Date: Mon, 10 Feb 2014 12:26:41 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: On 9 February 2014 19:52, michael noble wrote: >> >> "The easiest way to build Ingen from SVN is to build the entire >> http://svn.drobilla.net/lad repository (since there are dependencies between >> the various projects)." >> > > I guess Mark's point is that it has been impossible to build ingen from the > AUR scripts for some time, which is sadly true. Trying to mix svn of some of > drobilla libraries with AUR or Arch packages of those libraries, or even the > full svn tree with apps that depend on the official packages, quickly > results in a mess. > Ganv doesn't build at all; I've already tried the ganv-svn PKGBUILD in Arch. Canvas.cpp makes a call to graphviz (which is 'a mess' to paraphrase someone involved in maintaining ganv) with the wrong number of arguments. I guess I could also try graphviz-git (instead of the Arch binary) in the vague hope that the authors of graphviz have reverted the change that broke ganv, but graphviz-git requests over 100MB of dependencies to build, for which I don't have the bandwidth to burn unless someone out there can assure me that it's worthwhile. From ralf.mardorf at rocketmail.com Mon Feb 10 12:31:47 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 10 Feb 2014 13:31:47 +0100 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: Message-ID: <1392035507.762.59.camel@archlinux> On Mon, 2014-02-10 at 19:11 +0700, Shani Hadiyanto Pribadi wrote: > (you can ignore the part about low pass). On Sun, 2014-02-09 at 21:27 +0000, Fons Adriaensen wrote: > Note that it is normally more important to remove the bass signal for > the main speakers than to remove the mid/high part for the sub. I have never listened to a subwoofer system that did satisfy my taste for natural sound. At least not when I was aware about a subwoofer, perhaps some systems were very good and I didn't notice the subwoofer. I suspect that the subwoofer anyway only will transmit low frequencies, but as Fons pointed out, I suspect that many systems I heard were that bad, perhaps because they missed to filter out the bass for the mid/hi speakers correctly. JFTR when mixing everything to mono, IOW also for the mid/hi speakers, than the OP might not miss the HiHat that is more to one side than to the other, but it could happen that some effects, vocals and other signals will be completely eliminated, especially if the OP does listen to simple popular music, usually made by DJs. Some of those "engineers" use out of phase effects to spread the stereo impression. From paul at linuxaudiosystems.com Mon Feb 10 14:05:49 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Mon, 10 Feb 2014 09:05:49 -0500 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: <52F8B8B0.3080808@youmail.dk> References: <52F75B06.6050405@youmail.dk> <52F7C887.8070100@youmail.dk> <52F8B8B0.3080808@youmail.dk> Message-ID: On Mon, Feb 10, 2014 at 6:32 AM, Atte wrote: > > So it's not possible to wrap AMS (with it's self contained modules and > functionality intact) in VSTi? I'd be surprised, but I never coded a VSTi, > so how am I? > you could wrap AMS (or Ingen) as a plugin. In fact Ingen is available as an LV2 plugin already. It would have MIDI I/O at its "edges" and control voltage signals internally. > > It seems we are talking about two things: I'm talking about the "whole of > AMS, canvas + buildin modules, but nothing else", it seems to me you're > talking about "individual modules as VSTi and then try to connect them in > my host". Or? Aurelien's work involved turning the individual AMS modules into LV2 plugins, not AMS. You want a port of AMS itself to linux native VST(i). Totally different things. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Feb 10 14:10:53 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 10 Feb 2014 15:10:53 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] In-Reply-To: <1391798009.2060.67.camel@archlinux> References: <1391793617.2060.50.camel@archlinux> <1391798009.2060.67.camel@archlinux> Message-ID: <52F8DDED.8080500@autostatic.com> On 02/07/2014 07:33 PM, Ralf Mardorf wrote: > We could continue to disagree about the tremolo ;). I own an old school > crappy thing. It's nice, but already unusable for 80s Metal/Punk ;). No > Floyd Rose here ;). It has got its advantages, but much more > disadvantages :D. I sound detuned as Jimi did, but IMO this isn't an > advantage, it's disgusting for my taste. My SG happens to be equipped with a Bigsby tremelo. Awesome piece of equipment but you need to know how to use it ;) Best, 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 atte at youmail.dk Mon Feb 10 14:16:18 2014 From: atte at youmail.dk (Atte) Date: Mon, 10 Feb 2014 15:16:18 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F75B06.6050405@youmail.dk> <52F7C887.8070100@youmail.dk> <52F8B8B0.3080808@youmail.dk> Message-ID: <52F8DF32.4020303@youmail.dk> On 02/10/2014 03:05 PM, Paul Davis wrote: > Aurelien's work involved turning the individual AMS modules into LV2 > plugins, not AMS. You want a port of AMS itself to linux native VST(i). Exactly! > Totally different things. Exactly! -- Atte http://atte.dk http://modlys.dk From jeremy at autostatic.com Mon Feb 10 14:22:03 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 10 Feb 2014 15:22:03 +0100 Subject: [LAU] [Fwd: Re: Bitwig at long last...?] In-Reply-To: <1391796037.2060.53.camel@archlinux> References: <1391793617.2060.50.camel@archlinux> <1391796037.2060.53.camel@archlinux> Message-ID: <52F8E08B.20101@autostatic.com> On 02/07/2014 07:00 PM, Ralf Mardorf wrote: > Yesno ;), the SGs fingerboard is not equal, but more flat like a > fingerboard of a classical guitar. From all the electric guitars I own and have owned the SG does not have the flattest fingerboard. I think my Telecaster with its maple neck has a flatter fingerboard. And I used to own a Rickenbacker 620 (of which I very much regret I sold it even thought I bought an SG from that money) and that one had the flattest fingerboard I've ever encountered on an electric guitar. 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 matthew.garman at gmail.com Mon Feb 10 15:49:05 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Mon, 10 Feb 2014 09:49:05 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <20140210013948.GC10498@linuxaudio.org> References: <20140209212705.GB10498@linuxaudio.org> <20140210013948.GC10498@linuxaudio.org> Message-ID: On Sun, Feb 9, 2014 at 7:39 PM, Fons Adriaensen wrote: > I can't make sense of this. > > One output of the DAC goes to the power amp (for the main speakers) > The other output goes to the sub. > > So you won't have stereo. In that case, just send the mono signal > (L+R) to both channels of the DAC. If you don't do that, you'll > just be listening to one channel. And then I really wonder why > you bother about the bass - you'll be missing the entire second > channel anyway. I should have said, "One *stereo* output of the DAC..." I overloaded and loosely used the term "output" in my original description. What I meant was, the DAC has two *stereo* outputs. In other words, four total conductors (two "L" outputs and two "R" outputs). I intend for one stereo output pair to go to the power amp. That leaves another stereo pair left for the sub. But that stereo output pair for the sub requires two conductors, and I only have one (a single coax run). > And why do you use *coax* cable between the DAC and the sub ? > A normal screened cable will do... That's the way the house was built. Pulling more wire is of course another option, but not one I'm willing to entertain. :) > If you're a bit handy with a soldering iron, all it takes to > mix the DAC outputs to mono would be two resistors and some > short pieces of wire and connectors. That's not out of the question, but I just wanted to see if I could maybe hack it at the software level first. From matthew.garman at gmail.com Mon Feb 10 15:59:01 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Mon, 10 Feb 2014 09:59:01 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> Message-ID: On Mon, Feb 10, 2014 at 6:08 AM, Shani Hadiyanto Pribadi wrote: > Now this is where I lost you, how many speakers do you have exactly, > and how many amps for them? Doesn't it mean that you need as many amps > for each speakers you have if you ditch the speaker selector? > > If indeed you want to control it like that then you also need a soundcard > with as many output channels as the number of power amps channels for > the speakers. I currently have three pairs of speakers, and intend to add a fourth as soon as this is all squared away (the wiring for the fourth is already in place, I just need to connect them to speakers). The existing three pairs of speakers are the in-ceiling speakers I've been talking about. The fourth pair of speakers will be traditional full-range speakers. That's one reason why I want to send the full-range signal to the power amp (and in turn the speaker selector). I think you might be confusing the AVR with the speaker selector; they are two different devices. The speaker selector is a passive device (it doesn't require external power). It simply allows an amplified, speaker-level signal to be multiplexed. Without such a device, I would indeed need to have a 1:1 power amp to speaker pair ratio. Here's a concrete example of a speaker selector, Monoprice product 8229: http://www.monoprice.com/Product?c_id=109&cp_id=10903&cs_id=1090305&p_id=8229&seq=1&format=2 From matthew.garman at gmail.com Mon Feb 10 16:08:21 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Mon, 10 Feb 2014 10:08:21 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <1392035507.762.59.camel@archlinux> References: <1392035507.762.59.camel@archlinux> Message-ID: On Mon, Feb 10, 2014 at 6:31 AM, Ralf Mardorf wrote: > JFTR when mixing everything to mono, IOW also for the mid/hi speakers, > than the OP might not miss the HiHat that is more to one side than to > the other, but it could happen that some effects, vocals and other > signals will be completely eliminated, especially if the OP does listen > to simple popular music, usually made by DJs. Some of those "engineers" > use out of phase effects to spread the stereo impression. This is why I would prefer to only downmix frequencies below 150 Hz or so. That is, preserve the upper frequencies as much as possible, and only meddle with the frequencies the subwoofer will handle. The existence of phase effects never occurred to me. That's an interesting complication. From fons at linuxaudio.org Mon Feb 10 16:11:02 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 10 Feb 2014 16:11:02 +0000 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> Message-ID: <20140210161102.GA31923@linuxaudio.org> On Mon, Feb 10, 2014 at 09:59:01AM -0600, Matt Garman wrote: > The existing three pairs of speakers are the in-ceiling speakers I've been > talking about. The fourth pair of speakers will be traditional > full-range speakers. That's one reason why I want to send the > full-range signal to the power amp (and in turn the speaker selector). If these in-ceiling speakers are small ones (that really need a sub) they won't appreciate high level LF signals they can't reproduce. In the best case this will result in distortion, in the worst you could easily damage them. 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 matthew.garman at gmail.com Mon Feb 10 16:22:16 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Mon, 10 Feb 2014 10:22:16 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <20140210161102.GA31923@linuxaudio.org> References: <20140209212705.GB10498@linuxaudio.org> <20140210161102.GA31923@linuxaudio.org> Message-ID: On Mon, Feb 10, 2014 at 10:11 AM, Fons Adriaensen wrote: > On Mon, Feb 10, 2014 at 09:59:01AM -0600, Matt Garman wrote: > >> The existing three pairs of speakers are the in-ceiling speakers I've been >> talking about. The fourth pair of speakers will be traditional >> full-range speakers. That's one reason why I want to send the >> full-range signal to the power amp (and in turn the speaker selector). > > If these in-ceiling speakers are small ones (that really need a sub) > they won't appreciate high level LF signals they can't reproduce. > In the best case this will result in distortion, in the worst you > could easily damage them. I'll have to pull one to check and see what they are exactly. I received no documentation. I'm currently sending a full range signal to them and I don't hear any obvious distortion. I guess I've always assumed they had a built-in passive crossover network to filter out frequencies the driver can't handle. They definitely don't reproduce low-level signals, though, as there's definitely missing sound without the sub. -Matt From jh at brainiac.com Mon Feb 10 16:52:09 2014 From: jh at brainiac.com (Joe Hartley) Date: Mon, 10 Feb 2014 11:52:09 -0500 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> <20140210013948.GC10498@linuxaudio.org> Message-ID: <20140210115209.69c7e4a4a0ec3dd1ad26671b@brainiac.com> On Mon, 10 Feb 2014 09:49:05 -0600 Matt Garman wrote: > I overloaded and loosely used the term "output" in my original > description. What I meant was, the DAC has two *stereo* outputs. In > other words, four total conductors (two "L" outputs and two "R" > outputs). This becomes a matter of a simple Y-adapter to combine the two outputs into one. Don't worry at all about filtering out the signals at the server end, the subwoofer will have a low-pass filter that will cut out anything above a certain frequency anyway. On some, the frequency of the low-pass filter is adjustable. I believe you're adding a lot of needless complexity to this project. How bass frequencies are panned is irrelevent, as a simple Y adapter solves that. Filtering at the server is irrelevant, since the subwoofer contains a low-pass filter. It's all about getting the signal to the sub using the available coax run, and if it's a standard 75 ohm cable, you should be fine as long as you get the right adapters and make sure they're securely fitted at each end. You could also put new adapters on the cables, as there are compression RCA connectors but that may require additional know-how. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From matthew.garman at gmail.com Mon Feb 10 17:28:00 2014 From: matthew.garman at gmail.com (Matt Garman) Date: Mon, 10 Feb 2014 11:28:00 -0600 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: <20140210115209.69c7e4a4a0ec3dd1ad26671b@brainiac.com> References: <20140209212705.GB10498@linuxaudio.org> <20140210013948.GC10498@linuxaudio.org> <20140210115209.69c7e4a4a0ec3dd1ad26671b@brainiac.com> Message-ID: On Mon, Feb 10, 2014 at 10:52 AM, Joe Hartley wrote: > This becomes a matter of a simple Y-adapter to combine the two outputs > into one. Don't worry at all about filtering out the signals at the > server end, the subwoofer will have a low-pass filter that will cut out > anything above a certain frequency anyway. On some, the frequency of > the low-pass filter is adjustable. I'm not sure this is right. Based on what I was able to glean from searching the web, there are a couple issues with this: (1) Depending on how the DAC outputs are wired, using a Y-adapter on one set of stereo outputs might also cause the other set of stereo outputs to be mono as well. (2) At least a basic resistor network (one has already been described in this thread) is needed to "correctly" sum the two channels. Now I can't find the link that gave an explanation of why this is, but my takeaway was that a simple Y-adapter isn't the way to go. From patrick at rumblebee.de Mon Feb 10 18:18:57 2014 From: patrick at rumblebee.de (Patrick) Date: Mon, 10 Feb 2014 19:18:57 +0100 Subject: [LAU] Steinberg UR22 error -5 after patching Message-ID: <52F91811.7080404@rumblebee.de> As some other users I'm trying to get the Steinberg/Yamaha UR22 to work. I added Clemens' patch to quirks-table.h and compiled a new kernel (although I noticed later that patching alsa sources and creating a package should've been enough?). The USB LED on the card now stopped blinking after booting just as it does on windows as soon as I installed the drivers there, but still it is not recognized as a (working) sound card. dmesg output is: [ 7803.974700] snd-usb-audio: probe of 1-2:1.0 failed with error -5 [ 7803.994164] snd-usb-audio: probe of 1-2:1.1 failed with error -5 [ 7803.994182] usbcore: registered new interface driver snd-usb-audio I tried to disable onboard/HMDI sound devices, which did not help. (onboard via BIOS and modprobe.d conf) Does anyone have an idea what could help to do next? Patrick From shanipribadi at gmx.net Mon Feb 10 21:31:35 2014 From: shanipribadi at gmx.net (Shani Hadiyanto Pribadi) Date: Tue, 11 Feb 2014 04:31:35 +0700 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: <20140209212705.GB10498@linuxaudio.org> Message-ID: On 10 February 2014 22:59, Matt Garman wrote: > I currently have three pairs of speakers, and intend to add a fourth > as soon as this is all squared away (the wiring for the fourth is > already in place, I just need to connect them to speakers). The > existing three pairs of speakers are the in-ceiling speakers I've been > talking about. The fourth pair of speakers will be traditional > full-range speakers. That's one reason why I want to send the > full-range signal to the power amp (and in turn the speaker selector). > > I think you might be confusing the AVR with the speaker selector; they > are two different devices. The speaker selector is a passive device > (it doesn't require external power). It simply allows an amplified, > speaker-level signal to be multiplexed. Without such a device, I > would indeed need to have a 1:1 power amp to speaker pair ratio. I thought that you wanted to control the speaker selection from linux, and since the speaker selector is not computer controlled then the other way to achieve it would have been to have the channels controllable from linux. Ah, since you have 4 output channels then the solution I provided in my previous e-mail will work just fine for you. (I'll link it again incase you missed it http://alsa.opensrc.org/Low-pass_filter_for_subwoofer_channel_%28HOWTO%29 ) You will only use 3 of the channels (set it up as L, R, M), no need to fuss with a Y-adapter then. Although if you could use jackd as the audio output instead of alsa, then it would be easier to just use Fons' awesome zita-lrx to upmix the channels, with the added bonus of a crossover. http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html I would also like to second Fons' suggestion to hi-pass the signals sent to the ceiling speakers, you can turn off the highpass when you switch to the fullrange speakers. Audio quality aside, sending LF to speakers that are too small to produce them is just a waste of power. Too much cone excursion will damage the ceiling speakers. From normalperson at yhbt.net Tue Feb 11 07:33:27 2014 From: normalperson at yhbt.net (Eric Wong) Date: Tue, 11 Feb 2014 07:33:27 +0000 Subject: [LAU] On the fly bass downmixing to mono (mpd or alsa)? In-Reply-To: References: Message-ID: <20140211073327.GA16053@dcvr.yhbt.net> Matt Garman wrote: > I started the thread "How is the bass mixed? Per-channel frequency > analysis? Histogram?" a few weeks ago. One suggestion that came up a > few times was to simply downmix the bass to mono before sending it > out. > > How would I go about this? Currently, this system doesn't have any > audio. What I intend to do is send audio out over USB to a DAC. > Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" > to mono. Ideally the frequency should be configurable, but I think > about 150Hz or below is a good start. Instead of mpd, you could try my dtas-player component of dtas (duct-tape audio suite)[1]. dtas-player is nearly as much a *nix shell as a music player. It is written in Ruby, but runs arbitrary shell commands of your choosing (sox by default). I suggest starting with a shell script using sox like below: (My sox knowledge is rusty, so the following may not be 100% correct:) ------------------------ /path/to/sum-mono.sh ------------------------ #!/bin/sh set -e : note: TRIMFX, RGFX, INFILE, and SOXFMT vars come from dtas-player COMMON="sox \"$INFILE\" -p $TRIMFX" MONO="remix 1,2 1,2" : stacking two filters should give a Linkwitz-Reily crossover at 150Hz XO=${XO-150} lo="$COMMON lowpass $XO lowpass $XO $MONO" hi="$COMMON highpass $XO highpass $XO" exec sox --combine=mix -v1 "|$lo" -v1 "|$hi" $SOXFMT - $RGFX ---------------------------------------------------------------------- Then, after dtas-player is started in a different terminal: $ dtas-ctl source ed sox command=$HOME/bin/sum-mono.sh $ dtas-ctl play At the core, dtas-player is really like running the following in a shell: source-command | sink-command Except it's possible to replace source-command without restarting sink-command (and vice-versa) while preserving the same pipe. To restore default behavior: $ dtas-ctl source ed sox command= [1] - http://dtas.80x24.org/ - a bunch of scripts by me, a GUI-phobe :) Hopefully I'll have more time for this project soon. Disclaimer: I was also an mpd developer and even project leader for a short while, but decided to go off on my craziness. Anyways I find it much more enjoyable to just glue sox/ecasound/whatever commands together with pipes than deal with maintaining a music player in C (or C++). From beatleboy07 at gmail.com Tue Feb 11 08:21:44 2014 From: beatleboy07 at gmail.com (Clifford Dunn) Date: Tue, 11 Feb 2014 00:21:44 -0800 Subject: [LAU] Focusrite Scarlett 2i2 and KXStudio Message-ID: Hi list, This is my first post as I'm a newcomer to pure Linux use. My issue is that I'm using the newest LiveCD install of KXStudio, and trying to use my Focusrite Scarlett 2i2 interface. Sound comes out, and the sound quality is good, but it gets a lot of interrupts. This is visible by the input gains flashing amber, and audio clicks and pops somewhat regularly. I've seen this same problem on various forums, but no solutions. Does anyone have any advice? Thanks! Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn From mlang at delysid.org Tue Feb 11 15:56:23 2014 From: mlang at delysid.org (Mario Lang) Date: Tue, 11 Feb 2014 16:56:23 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: <20140206075059.GF2392@moule.localdomain> (Alexandre Ratchov's message of "Thu, 6 Feb 2014 08:50:59 +0100") References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <87vbwvatsl.fsf@fx.delysid.org> <53039.86.105.95.182.1391522299.squirrel@boosthardware.com> <87r47hvpdo.fsf@fx.delysid.org> <20140206075059.GF2392@moule.localdomain> Message-ID: <87ppmtr6c8.fsf@fx.delysid.org> Alexandre Ratchov writes: > On Wed, Feb 05, 2014 at 11:13:23AM +0100, Mario Lang wrote: >> "Patrick Shirkey" writes: >> >> > On Wed, February 5, 2014 12:31 am, Mario Lang wrote: >> >> raf writes: >> >> >> >>> Hello, >> >>> >> >>> you'l probably be happy to know the existence of three great tools : >> >>> midish, linuxsampler and Nama. >> >>> 1) midish is a command line midi sequencer with a lot of great features >> >>> http://www.midish.org/ >> >> >> >> midish looks rather interesting. However, the manual.html basically >> >> just explains how to record data from an input device. Does latest >> >> midish support creating MIDI data from scratch, and if so, is there >> >> perhaps some examples on how to do that? >> >> >> > >> > Check this section : >> > >> > http://www.midish.org/manual.html#ev >> > >> > You can compose note on/off events and save the sequence as a song or >> > export the song to .mid >> >> A simple example on how to actually do that would be appreciated. > > Hi, > > It's kinda painful, as the tool was designed to work with an input > device. > > You could create a track and add events one by one, ex: > > onew piano {0 0} > tnew mytrack > taddev 1 0 0 {non piano 64 90} > taddev 1 1 0 {noff piano 64 0} > > see: > http://www.midish.org/manual.html#func_taddev > http://www.midish.org/manual.html#ev_ev > > you'll get warnings about unterminated notes and/or other > anomalies, that you can ignore until all events are added. Once > you're done, you could run tcheck to fix any anomaly, just in case. OH, that last paragraph was helpful. I was actually taken aback by the warnings when I did my own (incomplete) attempt to duplicate your example above. While I see (now) that midish was originally designed for a different use case (direct MIDI input) I still think it might be helpful for others to add a section to the manual which basically contains your posting I am replying to :-). The basic example on how to create MIDI events from scratch, and maybe a paragraph about why it is so cumbersome and a reference to tcheck regarding the warnings, would have been everything I needed when I read the manual :-). And mind you, I actually *read the manual* :-) > To make the process less painful, you could define routines to make > certain things automatic, depending on your needs. I will look into this, but honestly, from the syntax above, it doesn't look like midish can fullfil my requirements as explain in the beginning of this thread. I have learnt alot about its potential uses though, so this was productive, thank you. -- CYa, ????? From willgodfrey at musically.me.uk Tue Feb 11 18:02:48 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 11 Feb 2014 18:02:48 +0000 Subject: [LAU] jaaa Message-ID: <20140211180248.4ace4343@debian> I've just been using this to do some tests on an Amp and it's produced some interesting and useful results. However, I would really like a log frequency scale. Is this a practical future option? I'm using V 0.8.4 which comes with debian testing. -- 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 markhadman at googlemail.com Tue Feb 11 18:09:16 2014 From: markhadman at googlemail.com (mark hadman) Date: Tue, 11 Feb 2014 18:09:16 +0000 Subject: [LAU] jaaa In-Reply-To: <20140211180248.4ace4343@debian> References: <20140211180248.4ace4343@debian> Message-ID: Have you tried japa by the same author? It has a log frequency scale, although I don't know how it stacks up feature-wise. On 11 February 2014 18:02, Will Godfrey wrote: > I've just been using this to do some tests on an Amp and it's produced some > interesting and useful results. However, I would really like a log frequency > scale. > > Is this a practical future option? > > I'm using V 0.8.4 which comes with debian testing. > > -- > 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. > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From arnold at arnoldarts.de Tue Feb 11 19:18:58 2014 From: arnold at arnoldarts.de (Arnold Krille) Date: Tue, 11 Feb 2014 20:18:58 +0100 Subject: [LAU] jaaa In-Reply-To: References: <20140211180248.4ace4343@debian> Message-ID: <20140211201858.4334ceef@orinoco> Am Tue, 11 Feb 2014 18:09:16 +0000 schrieb mark hadman : > Have you tried japa by the same author? It has a log frequency scale, > although I don't know how it stacks up feature-wise. The only chance for jaaa to win against japa feature-wise is when you want less features to win... - Arnold -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 230 bytes Desc: not available URL: From fons at linuxaudio.org Tue Feb 11 19:22:57 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 11 Feb 2014 19:22:57 +0000 Subject: [LAU] jaaa In-Reply-To: <20140211180248.4ace4343@debian> References: <20140211180248.4ace4343@debian> Message-ID: <20140211192257.GA6237@linuxaudio.org> On Tue, Feb 11, 2014 at 06:02:48PM +0000, Will Godfrey wrote: > I've just been using this to do some tests on an Amp and it's produced some > interesting and useful results. However, I would really like a log frequency > scale. The F-scale is linear because the resolution of the analyser is independent of frequency. It provides as much 'detail' between 10.1 and 10.2 kHz as it does between 100 and 200 Hz. Using a log scale would mean that most of that information is thrown away. > Is this a practical future option? It hasn't been for the last 10 years, and that is likely to stay that way. For the type of measurements that JAAA is designed for a log scale wouldn't make much sense. The only type of measurement that is usually presented with a log frequency scale is the traditional 'frequency response', plot and Jaaa isn't the right tool to create 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 lievenmoors at gmail.com Tue Feb 11 20:08:05 2014 From: lievenmoors at gmail.com (Lieven Moors) Date: Tue, 11 Feb 2014 21:08:05 +0100 Subject: [LAU] A text-only environment for composing electronic music? In-Reply-To: References: <87zjmki22c.fsf@fx.delysid.org> <4105AFCB-B02B-46F9-97CB-2C3D5FA8AF98@gmail.com> <52E3E09B.8050508@gmx.net> <52E4F580.4020204@gmx.net> Message-ID: <20140211200805.GA421@satellite> On Sun, Jan 26, 2014 at 08:21:44AM -0800, Len Ovens wrote: > > The one thing I haven't managed to get to work on screen is scrolling up as > I could in VTs or xterms (with shift pageup etc.) I end up running a command > twice first to find out it runs text off the screen and a second time with a > pager. Probably I am missing something. I use tmux as well and I set this in the configuration file to be able to scroll up with shift-pageup. set -g terminal-overrides 'xterm*:smcup@:rmcup@' I think something similar could work with screen as well... On the other hand, it is almost as easy to use Ctrl-b pageup (in tmux again). It enters copy mode automatically... greetings, lieven From p8rpp at aol.com Tue Feb 11 20:16:12 2014 From: p8rpp at aol.com (Peter P.) Date: Tue, 11 Feb 2014 21:16:12 +0100 Subject: [LAU] jaaa In-Reply-To: <20140211192257.GA6237@linuxaudio.org> References: <20140211180248.4ace4343@debian> <20140211192257.GA6237@linuxaudio.org> Message-ID: <20140211201612.GA22511@aol.de> * Fons Adriaensen [2014-02-11 20:23]: > On Tue, Feb 11, 2014 at 06:02:48PM +0000, Will Godfrey wrote: > > > I've just been using this to do some tests on an Amp and it's produced some > > interesting and useful results. However, I would really like a log frequency > > scale. > > The F-scale is linear because the resolution of the analyser is > independent of frequency. It provides as much 'detail' between > 10.1 and 10.2 kHz as it does between 100 and 200 Hz. Using a log > scale would mean that most of that information is thrown away. > > > Is this a practical future option? > > It hasn't been for the last 10 years, and that is likely to > stay that way. For the type of measurements that JAAA is > designed for a log scale wouldn't make much sense. > > The only type of measurement that is usually presented with > a log frequency scale is the traditional 'frequency response', > plot and Jaaa isn't the right tool to create that. As I am just sitting on my desk trying to find a tool to teach basic filter frequency responses to my students, I wonder what such a tool could be. best, P From fons at linuxaudio.org Tue Feb 11 20:25:09 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 11 Feb 2014 20:25:09 +0000 Subject: [LAU] jaaa In-Reply-To: <20140211201612.GA22511@aol.de> References: <20140211180248.4ace4343@debian> <20140211192257.GA6237@linuxaudio.org> <20140211201612.GA22511@aol.de> Message-ID: <20140211202509.GB6237@linuxaudio.org> On Tue, Feb 11, 2014 at 09:16:12PM +0100, Peter P. wrote: > As I am just sitting on my desk trying to find a tool to teach basic > filter frequency responses to my students, I wonder what such a tool > could be. Japa will show them. Connect Japa:pink_noise -> EQ:in EQ:out -> Japa:in-1 Japa:pink_noise -> Japa:in-2 Select input 1 on A, input 2 on B, slow respose, and display A/B. 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 jpsandys at gmail.com Tue Feb 11 21:35:43 2014 From: jpsandys at gmail.com (Jeff Sandys) Date: Tue, 11 Feb 2014 13:35:43 -0800 Subject: [LAU] A text-only environment for composing electronic music? Message-ID: Date: Sat, 25 Jan 2014 15:11:23 +0100 > From: Mario Lang > To: linux-audio-user at lists.linuxaudio.org > Subject: [LAU] A text-only environment for composing electronic music? > Message-ID: <87zjmki22c.fsf at fx.delysid.org> > Content-Type: text/plain; charset=utf-8 > > Hi. > > I am looking for a programmable (text mode) sequencer solution. > I know that Linux has a few small languages for creating > MIDI files, like MMA. Even LilyPond can be tricked into being a MIDI > file generating language. However, none of the solutions I have seen so > far could be easily integrated as the center/hub of a full composition. > > > -- > CYa, > ????? > > I didn't see these text based music programs in this thread. For a MIDI based composition environment athenacl is a very dense python based command line composition program. It does have a steep learning curve but the thick documentation is good. I don't think it is exactly what you are looking for but if you are interested in creating your own composition environment, this might be worthwhile. It seems to be intended for music students who can extend it with python. http://www.flexatone.org/athena.html If you are more interested in live coding ixi is a fun and easy to comprehend command line program, built on top of SuperCollider. It probably wouldn't be the center of a full composition, but its command line syntax is sweet. http://ixi-audio.net/ixilang/index.html Renick Bell blew me away at LAC2013 using his Conductive library for Haskell. It appears that you could compose a whole piece as a program or revise the program on the fly. http://www.renickbell.net/doku.php?id=conductive -- Jeff Sandys -------------- next part -------------- An HTML attachment was scrubbed... URL: From lievenmoors at gmail.com Tue Feb 11 21:50:38 2014 From: lievenmoors at gmail.com (Lieven Moors) Date: Tue, 11 Feb 2014 22:50:38 +0100 Subject: [LAU] Script to clean up Ardour unused files ? In-Reply-To: References: <20140126155318.082bb58a@mistral> <20140126164618.29d4bc51@mistral> <201401271045.40327.zotz@100jamz.com> <20140128193523.2fd7928d@mistral> <20140129012533.GA31425@linuxaudio.org> <20140129021012.GB31425@linuxaudio.org> Message-ID: <20140211215038.GB421@satellite> On Tue, Jan 28, 2014 at 09:21:00PM -0500, Paul Davis wrote: > On Tue, Jan 28, 2014 at 9:10 PM, Fons Adriaensen wrote: > > > On Tue, Jan 28, 2014 at 08:32:39PM -0500, Paul Davis wrote: > > > > > loading a snapshot and then saving will alter the state of the > > > snapshot that was loaded, not the "current session" or any other > > > snapshot. > > > > That exactly is the problem. I would expect things to work as you > > decribe when a 'save as' file is reloaded, and that is also what > > happens of course. But I wouldn't expect a snapshot to be modified > > by a normal 'save' but only when the user explicitly asks for it, > > by using 'new snapshot' with the same name (which would have to be > > typed in manually). Overwriting snapshots to me feels like rewriting > > history, or bypassing a version control system. > > > > The root of the problem is that files created by 'new snapshot' and > > 'save as' are identical, and both are treated as 'session files' when > > reloaded. But the user's intention behind creating either of them > > is likely to be different. That intention is handled correctly when > > the files are created (a snapshot won't be modified by later changes), > > but not when they are reloaded. > > > > the only intended difference between "save as" and "snapshot" is whether or > not ardour does subsequent saves (while in the same instance) to the *new* > session file or the current one. > > both operations are intended to create snapshots and snapshots are never > read-only, except by user intent. I agree with Fons here, and I have been bitten by this more than once myself. Snapshots are often used as a safety measure, and now it is too easy to accidentily save your changes to a session that was not meant to change. The fact that the original session, and the snapshots often look very similar, makes such accidental saves very likely. I agree it is very flexible that snapshots can be treated as ordinary sessions, and that shouldn't change. If only there would be an option in ardour (that would be on by default) that would warn the user that he is saving changes to a snapshot, and NOT to the original session. lieven From fons at linuxaudio.org Tue Feb 11 22:24:51 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 11 Feb 2014 22:24:51 +0000 Subject: [LAU] Script to clean up Ardour unused files ? In-Reply-To: <20140211215038.GB421@satellite> References: <20140126164618.29d4bc51@mistral> <201401271045.40327.zotz@100jamz.com> <20140128193523.2fd7928d@mistral> <20140129012533.GA31425@linuxaudio.org> <20140129021012.GB31425@linuxaudio.org> <20140211215038.GB421@satellite> Message-ID: <20140211222451.GC6237@linuxaudio.org> On Tue, Feb 11, 2014 at 10:50:38PM +0100, Lieven Moors wrote: > I agree with Fons here, and I have been bitten by this more than once > myself. Snapshots are often used as a safety measure, and now it is too > easy to accidentily save your changes to a session that was not meant to > change. The fact that the original session, and the snapshots often look > very similar, makes such accidental saves very likely. I agree it is very > flexible that snapshots can be treated as ordinary sessions, and that > shouldn't change. If only there would be an option in ardour (that would > be on by default) that would warn the user that he is saving changes to > a snapshot, and NOT to the original session. There is a very simple solution for this. Currently 'Save as' and 'New snapshot' save exactly the same file, the only difference being that the latter doesn't change the current session name. Now if 'Save as' would add a property 'is_session=True' and 'New snapshot' would not do that, then Ardour could do the right thing when loading a file, i.e. only change the current session name if this property is set and True. Then if the user loads a snapshot AND wants to modify it, or change the session name, then all he/she needs to do is use 'Save as' again. Or there could be an option in the 'Load' dialog. 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 beatleboy07 at gmail.com Tue Feb 11 23:39:21 2014 From: beatleboy07 at gmail.com (Clifford Dunn) Date: Tue, 11 Feb 2014 15:39:21 -0800 Subject: [LAU] Jack and WiFi Message-ID: So my previous email about the Focusrite can be discarded...I linked all the clicking and pops to the wifi. With it turned off, I have no issue. This is fine and great for my music production, but what do I do if I want to browse the net and listen to my music collection? Currently, it'll crackle like crazy. Thanks! Cliff Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn From paul at linuxaudiosystems.com Wed Feb 12 01:17:29 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 11 Feb 2014 20:17:29 -0500 Subject: [LAU] Jack and WiFi In-Reply-To: References: Message-ID: On Tue, Feb 11, 2014 at 6:39 PM, Clifford Dunn wrote: > So my previous email about the Focusrite can be discarded...I linked > all the clicking and pops to the wifi. With it turned off, I have no > issue. This is fine and great for my music production, but what do I > do if I want to browse the net and listen to my music collection? > Use the wired connection Seriously, this is a known problem on Linux, Windows and OS X. Certain WiFi devices and/or their drivers ruin scheduling latency, and thus ruin low latency audio. Not a very nice thing to discover but there's plenty more where that came from: http://manual.ardour.org/setting-up-your-system/the-right-computer-system-for-digital-audio/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Wed Feb 12 02:03:28 2014 From: len at ovenwerks.net (Len Ovens) Date: Tue, 11 Feb 2014 18:03:28 -0800 (PST) Subject: [LAU] Jack and WiFi In-Reply-To: References: Message-ID: On Tue, 11 Feb 2014, Clifford Dunn wrote: > So my previous email about the Focusrite can be discarded...I linked > all the clicking and pops to the wifi. With it turned off, I have no > issue. This is fine and great for my music production, but what do I > do if I want to browse the net and listen to my music collection? > Currently, it'll crackle like crazy. Wired network. I don't know your particular issue, but in my case if I set the latency high enough jack was ok. My pops were once a minute with wifi on. Once every 5 sec with it turned off and gone with the kernel module disabled with modeprobe -r. I also found that removing the kernel mod for the internal audio (or disabling it in bios) was helpful. You might try a USB wireless plug, a different wireless TX may cause less problems. It would have to be plugged into a USB port on a different irq than the audio. newer computers are faster, but seem to require higher latency. My old (more than 10 year old) P4 can do less than 1ms (as jack calculates things... the audio IF adds 1 ms to that) latency with no xruns, but because it is slower, I can run less effects. So far it has not been an issue, but I am not a keyboard player either :) For recording I use about 5ms latency and record with no effects. I mix down closer to 50ms latency and add any effects then. -- Len Ovens www.ovenwerks.net From tim at quitte.de Wed Feb 12 10:11:01 2014 From: tim at quitte.de (Tim Goetze) Date: Wed, 12 Feb 2014 11:11:01 +0100 (CET) Subject: [LAU] [ANN] CAPS 0.9.19 Message-ID: CAPS 0.9.19 =========== http://quitte.de/dsp/caps.html The latest release of this collection of LADSPA plugins extends the dynamics modulation capabilities of the 'virtual guitar amplifier' AmpVTS and fixes a nasty bug in the Noisegate circuit that had been causing spurious gain fluctuations in closed gate state. http://quitte.de/dsp/caps.html#Download Upgrading is recommended. Enjoy, Tim From touchstyle at gmail.com Fri Feb 14 17:01:03 2014 From: touchstyle at gmail.com (Touch Style) Date: Fri, 14 Feb 2014 18:01:03 +0100 Subject: [LAU] Trouble with zita-mu1 installation Message-ID: Ciao! I am iterested in zita-mu1 ( http://kokkinizita.linuxaudio.org/linuxaudio/zita-mu1-doc/quickguide.html). As described in http://linuxmusicians.com/viewtopic.php?f=48&t=12120&p=49453&hilit=zita#p49453, I am trying to compiling this uesful tool myself (the .deb package is not present in KXStudio 12.04 repos). Has someone done it? Thanks! -- *Dedenis AKA Touchstyle* *EMail : touchstyle (at) gmail (dot) com* *Web : http://www.touchstyle.it * -------------- next part -------------- An HTML attachment was scrubbed... URL: From rennabh at gmail.com Fri Feb 14 17:58:43 2014 From: rennabh at gmail.com (Renato) Date: Fri, 14 Feb 2014 18:58:43 +0100 Subject: [LAU] progressive time stretch Message-ID: <20140214185843.3dac8e7a@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a percussion piece written in hydrogen and recorded from there into ardour 3, one track per instrument. At a certain point I'd like to have an accelerando, i.e. a progressive tempo increase. What are my options? Best would be able to do it from within ardour, having many tempo changes, ardour set as jack master, so that when I play that part it gradually increases the jack playback tempo. The second solution would be to export the mix from ardour and use another program to increase the tempo in the .wav file. The problem is I'm not sure how exactly go about for neither solution... for the first one, could I use a knob on my midi controller to generate a smooth succession of tempo changes? How? For the second, which program could I use? I seem to recall there was some utility to generate rubberband map files, but I can't find it ATM. cheers, renato -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJS/llYAAoJEBz6xFdttjrf2XsH/2hP0asPC0gbje4AT6aNTf5f vByKypVexk4VD9OdHzCO3ypIaEU7zZPa2jKNcBkvKIsGF6BTUNkwwhpl0LNEAOJe tch/9j1boZV/+aM1lZ2/wEkKfAmC9WXG+8bW1hIcaENdTd40sbdPdQvrP3stSL4s sr5OuLSY+c9Gr/mZeXBkphqINFjBlrtEPFROh/709MBnctC5Y7iJto7XkogYz7C8 A5fHETF5xpvJ69L3HkOXNqGzzgDTAN8cC7rlXptqZ3LjyUd+6gNFAmruGICmeWss Khiy4htOXlNsVXWW7u/7VPhPxgDrfvDh8dJadFfU/XhtIxU+089uzEzVUeCJjsI= =tL4C -----END PGP SIGNATURE----- From rob at rektau.ukfsn.org Fri Feb 14 18:15:53 2014 From: rob at rektau.ukfsn.org (rob) Date: Fri, 14 Feb 2014 18:15:53 +0000 Subject: [LAU] Trouble with zita-mu1 installation In-Reply-To: References: Message-ID: <52FE5D59.4050104@rektau.ukfsn.org> On 14/02/14 17:01, Touch Style wrote: > Ciao! > I am iterested in zita-mu1 > (http://kokkinizita.linuxaudio.org/linuxaudio/zita-mu1-doc/quickguide.html). > As described in > http://linuxmusicians.com/viewtopic.php?f=48&t=12120&p=49453&hilit=zita#p49453 > , I am trying to compiling this uesful tool myself (the .deb package is > not present in KXStudio 12.04 repos). > > Has someone done it? > > Thanks! > > -- > / > / > /Dedenis AKA Touchstyle/ > /EMail : touchstyle (at) gmail (dot) com/ > /Web : http://www.touchstyle.it/ You need x11proto-core-dev A useful tool, if available, is apt-file. rob From markhadman at googlemail.com Fri Feb 14 20:29:12 2014 From: markhadman at googlemail.com (mark hadman) Date: Fri, 14 Feb 2014 20:29:12 +0000 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: Well, I managed to get this all up and running on Arch Linux. I had to edit the ganv-svn PKGBUILD, as it was stuck using an old revision from svn (from a couple of years ago I guess). Getting it to build from the newest version resolved the issues with it not building against graphviz. I also had to edit the 'provides' field in several PKGBUILDS and rebuild them, as it seems there is some strict version checking going on somewhere in the drobilla config files, which does not allow for building one svn against another svn package (it's checking for the presence of release versions rather than svn revisions). And after all that, I was able to (for the first time ever) get ingen up and running. Unfortunately a couple of minutes playing around seems to be all I can get before hitting a segfault (which has always been my experience with AMS anyway, on various different machines and distros). Now, I don't have much experience of bugfinding, so perhaps before I file issue # 1 on the ams-lv2 bug tracker, somebody could suggest a sensible route to determining whether it's ams-lv2 or ingen that's to blame. Learning, as ever.... On 10 February 2014 12:26, mark hadman wrote: > On 9 February 2014 19:52, michael noble wrote: > >>> >>> "The easiest way to build Ingen from SVN is to build the entire >>> http://svn.drobilla.net/lad repository (since there are dependencies between >>> the various projects)." >>> >> >> I guess Mark's point is that it has been impossible to build ingen from the >> AUR scripts for some time, which is sadly true. Trying to mix svn of some of >> drobilla libraries with AUR or Arch packages of those libraries, or even the >> full svn tree with apps that depend on the official packages, quickly >> results in a mess. >> > Ganv doesn't build at all; I've already tried the ganv-svn PKGBUILD in > Arch. Canvas.cpp makes a call to graphviz (which is 'a mess' to > paraphrase someone involved in maintaining ganv) with the wrong number > of arguments. I guess I could also try graphviz-git (instead of the > Arch binary) in the vague hope that the authors of graphviz have > reverted the change that broke ganv, but graphviz-git requests over > 100MB of dependencies to build, for which I don't have the bandwidth > to burn unless someone out there can assure me that it's worthwhile. From prettyvanilla at posteo.at Fri Feb 14 20:39:54 2014 From: prettyvanilla at posteo.at (prettyvanilla) Date: Fri, 14 Feb 2014 21:39:54 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: <52F7D140.6080902@rektau.ukfsn.org> Message-ID: <52FE7F1A.5020408@posteo.at> On 02/14/2014 09:29 PM, mark hadman wrote: > Well, I managed to get this all up and running on Arch Linux. I had to > edit the ganv-svn PKGBUILD, as it was stuck using an old revision from > svn (from a couple of years ago I guess). Getting it to build from the > newest version resolved the issues with it not building against > graphviz. I also had to edit the 'provides' field in several PKGBUILDS > and rebuild them, as it seems there is some strict version checking > going on somewhere in the drobilla config files, which does not allow > for building one svn against another svn package (it's checking for > the presence of release versions rather than svn revisions). > > And after all that, I was able to (for the first time ever) get ingen > up and running. Unfortunately a couple of minutes playing around seems > to be all I can get before hitting a segfault (which has always been > my experience with AMS anyway, on various different machines and > distros). Now, I don't have much experience of bugfinding, so perhaps > before I file issue # 1 on the ams-lv2 bug tracker, somebody could > suggest a sensible route to determining whether it's ams-lv2 or ingen > that's to blame. > > Learning, as ever.... > I made working PKGBUILDs for the required packages as well, which seem to work fine for me so far. Most depend on the respective svn versions explicitly, as that is probably the safer bet in general. I sent them to speps, the current AUR maintainer, but am still waiting on a response. For anyone interested in the meantime: ingen-svn: http://pastebin.com/BhDW3t14 raul-svn: http://pastebin.com/ggy9deaY ganv-svn: http://pastebin.com/GhJ2sbSa lilv-svn: http://pastebin.com/6hjRzCKE suil-svn: http://pastebin.com/qhAt9MDw lv2-svn: http://pastebin.com/DG0gTWsZ Cheers, prettyvanilla From el.doctor at laposte.net Fri Feb 14 22:37:26 2014 From: el.doctor at laposte.net (MK aka El Doctor) Date: Fri, 14 Feb 2014 23:37:26 +0100 Subject: [LAU] ams-lv2 1.0.2 and a tutorial In-Reply-To: References: Message-ID: <4176476.TrBVxWZpgB@io> Le dimanche 9 f?vrier 2014, 14:24:28 Aur?lien Leblond a ?crit : > >> Hello all, > >> > >> Today I just released the version 1.0.2 of the ams-lv2 plugins. > > > > Please don't kill me, but I would personally much prefer a linux-native > > VSTi version... > > *sight* why? > > There are only 3 hosts that I know of that can (or will be able to) > combine plugins in a modular way: ams, Ingen and Carla. > > Ingen and Carla both support LV2, and can be loaded as instrument (at > least for Ingen, not sure about Carla), what would linux-native VSTi > bring? > > Aur?lien > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Cool stuff tout ?a ^^ ;) MK -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: From rennabh at gmail.com Fri Feb 14 22:57:04 2014 From: rennabh at gmail.com (Renato) Date: Fri, 14 Feb 2014 23:57:04 +0100 Subject: [LAU] progressive time stretch In-Reply-To: References: <20140214185843.3dac8e7a@gmail.com> Message-ID: <20140214235704.661d0e49@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Fri, 14 Feb 2014 20:32:29 +0100 raf wrote: > Hello, > > i'd recommend to create a tempo map (vs timestreching which would > lead to artefacts) As you mention recording in ardour, you could use > it to create the tempo map, hi, what is a "tempo map" and how do you create it in ardour? couldn't find anything about it. I don't want to manually enter single tempo changes in ardour meanwhile I'm looking into klick, it seems it should work (though can't get it to sync to jack transport ATM) - but this way the tempo changes will not get recorded into ardour... i.e. every time I press play I should also somehow start klick... renato -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJS/p9DAAoJEBz6xFdttjrfu/YH/jL8ab4WU/JFEJBXuymG9Ljs Ui7nRuwlhMak+cSwsj8ikmDLNkJbdW0TIfsWJEGcNtJIrpSgGPPWUcIP0RNJ3UFC grcpY8UNROawf6cu7lwcU+3rjDCkepv4BsbR1m4rRTQZawPbzqc4AzeE4AzQMGR0 Whu14iqhA//VO5oD9+Q+KztlxXCtziKlOB0pwegdshwWuby9Y+JABKwwtUg7LMi3 tuu1/XKTbtUXuYoDic/M1Msl3jQuzq8l3Z69Dw2CU8biCeEwsCXT235J7pD3pba8 uc2AHtrA8re6Hgs+3SXs+TzzC7IHbRZGYOmGyxnQm3hTClLWSoDyhXnHoBjALG0= =SXwS -----END PGP SIGNATURE----- From edogawa at aon.at Sat Feb 15 09:03:07 2014 From: edogawa at aon.at (Edgar Aichinger) Date: Sat, 15 Feb 2014 10:03:07 +0100 Subject: [LAU] progressive time stretch In-Reply-To: <20140214235704.661d0e49@gmail.com> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> Message-ID: <1860410.xT5MlFYSHG@edhp> Am Freitag, 14. Februar 2014, 23:57:04 schrieb Renato: > On Fri, 14 Feb 2014 20:32:29 +0100 > raf wrote: > > > Hello, > > > > i'd recommend to create a tempo map (vs timestreching which would > > lead to artefacts) As you mention recording in ardour, you could use > > it to create the tempo map, > > hi, what is a "tempo map" and how do you create it in ardour? couldn't > find anything about it. I don't want to manually enter single tempo > changes in ardour I think he means putting several decreasing tempo markers for your accelerando to ardour's timeline, then re-record hydrogen while both sync to jack transport - then hydrogen should follow ardour's tempo and there would be no need for timestretching... Edgar > > meanwhile I'm looking into klick, it seems it should work (though can't > get it to sync to jack transport ATM) - but this way the tempo changes > will not get recorded into ardour... i.e. every time I press play I > should also somehow start klick... > > renato > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From rmouneyres at gmail.com Sat Feb 15 13:13:54 2014 From: rmouneyres at gmail.com (raf) Date: Sat, 15 Feb 2014 14:13:54 +0100 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: References: Message-ID: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> Hello, i face a terribly simple problem : i can't find any stereo panner ladspa plugin ! by stereo i mean : 2 audio channels inputs, pan control, 2 audio channel outputs can someone confirm there is not any or am i missing something ? Rapha?l From fons at linuxaudio.org Sat Feb 15 13:46:01 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 15 Feb 2014 13:46:01 +0000 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> References: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> Message-ID: <20140215134601.GB24505@linuxaudio.org> On Sat, Feb 15, 2014 at 02:13:54PM +0100, raf wrote: > i face a terribly simple problem : i can't find any stereo panner ladspa plugin ! > by stereo i mean : 2 audio channels inputs, pan control, 2 audio channel outputs > > can someone confirm there is not any or am i missing something ? One reason may be that such a thing is not really well defined. How do you want it to work ? * L and R inputs panned separately (A2 style), * Interacting width and pan controls (A3 style), * As a 'balance' control, * Other ? 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 rmouneyres at gmail.com Sat Feb 15 13:59:36 2014 From: rmouneyres at gmail.com (raf) Date: Sat, 15 Feb 2014 14:59:36 +0100 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: <20140215134601.GB24505@linuxaudio.org> References: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> <20140215134601.GB24505@linuxaudio.org> Message-ID: <583BD8C0-84F7-4B43-B8EF-AE85D61AADBE@gmail.com> >> i face a terribly simple problem : i can't find any stereo panner ladspa plugin ! >> by stereo i mean : 2 audio channels inputs, pan control, 2 audio channel outputs >> >> can someone confirm there is not any or am i missing something ? > > One reason may be that such a thing is not really well defined. > How do you want it to work ? > > * L and R inputs panned separately (A2 style), > * Interacting width and pan controls (A3 style), > * As a 'balance' control, > * Other ? for my use case i meant a "balance" control, like found in analog consoles. Having a width control (A3 style) can be a plus, but not absolutely necessary. Jonathan (@non-mixer) explained me there are multiple interpretations of what a "simple" balance control can be (i think he meant the "pan law"), and i have to admit that any would fit my needs, as i couldn't tell which has been existing in the various analog or digital consoles i have used in my life. I have only found an LV2 example here http://www.nongnu.org/ll-plugins/lv2pftci/ where the actual audio code is about 20 lines, so i may be able to duplicate that for a ladspa plugin, until someone did it before me ? Rapha?l From fons at linuxaudio.org Sat Feb 15 16:51:53 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 15 Feb 2014 16:51:53 +0000 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: <583BD8C0-84F7-4B43-B8EF-AE85D61AADBE@gmail.com> References: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> <20140215134601.GB24505@linuxaudio.org> <583BD8C0-84F7-4B43-B8EF-AE85D61AADBE@gmail.com> Message-ID: <20140215165153.GA8722@linuxaudio.org> On Sat, Feb 15, 2014 at 02:59:36PM +0100, raf wrote: > for my use case i meant a "balance" control, like found in > analog consoles. Having a width control (A3 style) can be > a plus, but not absolutely necessary. There are two plugins in this set. The first, stereo width, provides a limited range balance followed by a stereo width control. The second, stereo panner, has a full-range balance control followed by individual panners for L and R. This is probably the one you want. Both are de-zippered of course. 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 rennabh at gmail.com Sat Feb 15 17:15:24 2014 From: rennabh at gmail.com (Renato) Date: Sat, 15 Feb 2014 18:15:24 +0100 Subject: [LAU] progressive time stretch In-Reply-To: <1860410.xT5MlFYSHG@edhp> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> Message-ID: <20140215181524.114b376d@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 15 Feb 2014 10:03:07 +0100 Edgar Aichinger wrote: > Am Freitag, 14. Februar 2014, 23:57:04 schrieb Renato: > > On Fri, 14 Feb 2014 20:32:29 +0100 > > raf wrote: > > > > > Hello, > > > > > > i'd recommend to create a tempo map (vs timestreching which would > > > lead to artefacts) As you mention recording in ardour, you could > > > use it to create the tempo map, > > > > hi, what is a "tempo map" and how do you create it in ardour? > > couldn't find anything about it. I don't want to manually enter > > single tempo changes in ardour > > I think he means putting several decreasing tempo markers for your > accelerando to ardour's timeline, then re-record hydrogen while both > sync to jack transport - then hydrogen should follow ardour's tempo > and there would be no need for timestretching... > well yeah that was my idea from the start... probably I should have be more clear (and not use "time stretch" in the subject). The problem is that manually inputing tempo changes won't give a smooth enough accelerando, and it's not a very nice way of doing things anyway. I'm looking for a more automated solution Does anyone know if the script klick2ardour.py is supposed to work with ardour 3 sessions? It does nothing to my session -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJS/6CwAAoJEBz6xFdttjrf/vUH/1RaegskCqwXXl0urdCBpUYK rqGNPu6xolqyP+IIpJejCNEGs1YUz1TEH8J/Ek4ixaqARwXBWQv78lyyvTu2CDj7 XpoqUwidxRaoigbQPhBuVlEys6uw6j95jdIU20jxvQGdPXJx7OqEZv4oaje1XIS/ +ew0wBEE/y2TtePO4uM9BYY75iKqo2GCK8auA6WXpWvkoYHZuI1cRdIXMTHFb7CB rJNgpb9OkL/OMnVmVr07ZqY9lXTzaVaSSvow8yc9JmOZ3qzFaM6suS4lBQs0+aSK f4pDItoaLIDNfqpzyfXRM3H/cEPoxl4DWP7PZ9daO0ZCDpbOjl99JKxCZ2o69BI= =kPeG -----END PGP SIGNATURE----- From willgodfrey at musically.me.uk Sat Feb 15 17:28:46 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sat, 15 Feb 2014 17:28:46 +0000 Subject: [LAU] progressive time stretch In-Reply-To: <20140215181524.114b376d@gmail.com> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> Message-ID: <20140215172846.01f24f49@debian> On Sat, 15 Feb 2014 18:15:24 +0100 Renato wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Sat, 15 Feb 2014 10:03:07 +0100 > Edgar Aichinger wrote: > > > Am Freitag, 14. Februar 2014, 23:57:04 schrieb Renato: > > > On Fri, 14 Feb 2014 20:32:29 +0100 > > > raf wrote: > > > > > > > Hello, > > > > > > > > i'd recommend to create a tempo map (vs timestreching which would > > > > lead to artefacts) As you mention recording in ardour, you could > > > > use it to create the tempo map, > > > > > > hi, what is a "tempo map" and how do you create it in ardour? > > > couldn't find anything about it. I don't want to manually enter > > > single tempo changes in ardour > > > > I think he means putting several decreasing tempo markers for your > > accelerando to ardour's timeline, then re-record hydrogen while both > > sync to jack transport - then hydrogen should follow ardour's tempo > > and there would be no need for timestretching... > > > > well yeah that was my idea from the start... probably I should have > be more clear (and not use "time stretch" in the subject). The problem > is that manually inputing tempo changes won't give a smooth enough > accelerando, and it's not a very nice way of doing things anyway. I'm > looking for a more automated solution > > Does anyone know if the script klick2ardour.py is supposed to work with > ardour 3 sessions? It does nothing to my session Don't know if this is any help, but in the current version of Rosegarden you can mark a starting tempo and a finishing tempo and tell it to slowly go from one to the other. You can go either faster or slower, and set as many markers as you like. -- 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 rmouneyres at gmail.com Sat Feb 15 19:18:11 2014 From: rmouneyres at gmail.com (raf) Date: Sat, 15 Feb 2014 20:18:11 +0100 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: <20140215165153.GA8722@linuxaudio.org> References: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> <20140215134601.GB24505@linuxaudio.org> <583BD8C0-84F7-4B43-B8EF-AE85D61AADBE@gmail.com> <20140215165153.GA8722@linuxaudio.org> Message-ID: >> for my use case i meant a "balance" control, like found in >> analog consoles. Having a width control (A3 style) can be >> a plus, but not absolutely necessary. > > thank you Fons and Joel for pointing to those two plugins. I've just installed them, and will report how magically they behave for me tonight or tomorrow. Rapha?l From rmouneyres at gmail.com Sat Feb 15 20:21:50 2014 From: rmouneyres at gmail.com (raf) Date: Sat, 15 Feb 2014 21:21:50 +0100 Subject: [LAU] Rep: simple LADSPA stereo panner ? References: Message-ID: <814C8105-C615-4DFA-AC11-2204CC1F1994@gmail.com> >> Let me know how these work out for you :) > > here we are, a quick test to say, YES those two plugin perfectly do the job. > Using the balance control is just enough for what i need. > >> >> Plugin Name: "Stereo width" >> Plugin Label: "stereowidth" >> Plugin Unique ID: 1955 >> Maker: "Fons Adriaensen " > > playing with the stereo width can lead to very special results, not exactly what i expected, but interesting for mixing. > I'll take a closer look at the control behavior. > >> Plugin Name: "Stereo balance and panner" >> Plugin Label: "stpanner" >> Plugin Unique ID: 1956 >> Maker: "Fons Adriaensen " > Sure this one is the simplest and effective for a "standard" balance control. Using the L and R control can help if width reduction is needed, and the balance control becomes less brutal Great tools, thanks guys. Rapha?l From rennabh at gmail.com Sat Feb 15 22:42:23 2014 From: rennabh at gmail.com (Renato) Date: Sat, 15 Feb 2014 23:42:23 +0100 Subject: [LAU] progressive time stretch In-Reply-To: <20140215172846.01f24f49@debian> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> <20140215172846.01f24f49@debian> Message-ID: <20140215234223.2954a7f1@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 15 Feb 2014 17:28:46 +0000 Will Godfrey wrote: > Don't know if this is any help, but in the current version of > Rosegarden you can mark a starting tempo and a finishing tempo and > tell it to slowly go from one to the other. You can go either faster > or slower, and set as many markers as you like. Thanks, it's of great help, as ATM it seems the only way I'll have some success with this :) The trouble is Hydrogen, Ardour and Rosegarden are all following Jack Transport allright, but I can't see how to declare Rosegarden as master in the preferences (nor Ardour as slave) - Hydrogen instead has a "J. Master" button, which is unpressed. Right now I have a tempo ramp up defined in rosegarden, and as the transport goes through it I can see the BPMs changing in Rosegarden, but not in Ardour nor in Hydrogen... and the playback speed remains the same. The transport though is indeed syncronized (hitting pause pauses all three programs and so on). Maybe there were some recent changes to how Jack Transport works? Am I missing something or is this behaviour indeed strange? renato -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJS/+1VAAoJEBz6xFdttjrfMk4H/iPQYmyWXkOvgTjcNzx4ExKV gUiUc2O8W5Z0SUOis35/XW0ixRQzLNz6WcE0BnTUfXQdck/Uaf0t79MuGytT+2Ri dWsOURED+DUTH2qITC3oqy0PhO1zS5ps/sP4t6RsSD8ZqIIMo9At7u/iv1ZYOzTd Urer25KDDVqvf1+714FnscG46X4wQZ5nbIhkyvO4KZ73DzHcEuKXfZg+tA+cx2H3 M4MDnzbZdazPBtcf10qdEs5TGdXhnG6mVl23sX7IpW/YN4zqOnhfDFCwEbZrRdJL kVjPOYQu5VQBxW/8cIHRlLgH0ZDSZOVLEWqqi3mIMm76BQW9NUCAiX6US9a8alg= =CieA -----END PGP SIGNATURE----- From rennabh at gmail.com Sat Feb 15 22:56:34 2014 From: rennabh at gmail.com (Renato) Date: Sat, 15 Feb 2014 23:56:34 +0100 Subject: [LAU] progressive time stretch In-Reply-To: <20140215234223.2954a7f1@gmail.com> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> <20140215172846.01f24f49@debian> <20140215234223.2954a7f1@gmail.com> Message-ID: <20140215235634.23109d16@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > Right now I have a tempo ramp up defined in rosegarden, and as the > transport goes through it I can see the BPMs changing in Rosegarden, > but not in Ardour nor in Hydrogen... and the playback speed remains > the same. The transport though is indeed syncronized (hitting pause > pauses all three programs and so on). > I actually saw now that play, pause and stop operations are indeed synchronized, but the transport moves faster in Rosegarden, due to the tempo ramp up: after that Rosegarden is ahead of the others, and its tranpsport is moving faster. Is this a bug somewhere? -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJS//CiAAoJEBz6xFdttjrfCdYH/3Q/VfzhIzPIupRgOnRSW3I7 QajIrfpgQ/BPoOXS8ozgPpP8c/lQ8fjY+VgyR3+1zVDYOZVsl1xJtvp4xwmuYL7y GZSQ5necbktzOAmhchsDci+26Oz1S+BwCpJjqAHhibcVgHiTd+7Kb2ga1fA1viNc fjGQIGiVgsKWtfSCPwoDJ4YjkL1kKc6wk+boAz1FKP9r+cPCvCsjSu1o6mRMYQ6X b0Mb57Urwjfypx3HC84YfYwCKW3dmuqg9bLOUM4GfM1zWuhImjHbz2x/jWZHcO5o DfC3tdNUV/EU0vT9AIUVYL9zFLHUYLF0zDayiaYpEZECDMDRqv3uLRNNGZAy7Hk= =O4uD -----END PGP SIGNATURE----- From paul at linuxaudiosystems.com Sun Feb 16 01:18:53 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sat, 15 Feb 2014 20:18:53 -0500 Subject: [LAU] progressive time stretch In-Reply-To: <20140215234223.2954a7f1@gmail.com> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> <20140215172846.01f24f49@debian> <20140215234223.2954a7f1@gmail.com> Message-ID: On Sat, Feb 15, 2014 at 5:42 PM, Renato wrote: > ----but I can't see how to declare > Rosegarden as master in the preferences (nor Ardour as slave) - > Session > Properties > Sync .... choose the external clock source Then use the clock source button just right of the transport buttons to switch between the internal clock and the chosen external clock source. > Right now I have a tempo ramp up defined in rosegarden, and as the > transport goes through it I can see the BPMs changing in Rosegarden, > but not in Ardour nor in Hydrogen... ardour does not pay any attention to JACK tempo information. -------------- next part -------------- An HTML attachment was scrubbed... URL: From moshwe at gmail.com Sun Feb 16 11:04:10 2014 From: moshwe at gmail.com (Moshe Werner) Date: Sun, 16 Feb 2014 13:04:10 +0200 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: Another song from the event. https://www.youtube.com/watch?v=AJfKyThzHEg Enjoy Moshe On Mon, Nov 18, 2013 at 12:10 AM, Moshe Werner wrote: > Hi all, > > this is the first song released from a small live show that I recorded > recently. > The Band is "Shivat Zion", an Israeli Reggae band. > > https://www.youtube.com/watch?v=KZS57xoBET4 > > Everything was done with Linux software. > > Software I used: > > Arch Linux > Ardour 3 > Calf Lv2 effects (several) > GxZita_reverb (which I just recently discovered, and what should I > say... what a sweeet sounding reverb, new favorite) > Also played around with the new meters.lv2 bundle, I think though that I > didn't get the full idea of R128 yet. > Invada Dynamics processing > TAP > > > Everything regarding the Video work was done by my brother Michael. > He used KDEnlive to edit the video, also on an Arch machine. > > Hope you enjoy. > > Cheers, > > Moshe > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Sun Feb 16 11:37:02 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sun, 16 Feb 2014 12:37:02 +0100 Subject: [LAU] simple LADSPA stereo panner ? In-Reply-To: References: <246C5303-B521-4DBC-9C41-8E42192C0E04@gmail.com> <20140215134601.GB24505@linuxaudio.org> <583BD8C0-84F7-4B43-B8EF-AE85D61AADBE@gmail.com> <20140215165153.GA8722@linuxaudio.org> Message-ID: <5300A2DE.7050209@autostatic.com> On 02/15/2014 08:18 PM, raf wrote: >>> for my use case i meant a "balance" control, like found in >>> analog consoles. Having a width control (A3 style) can be >>> a plus, but not absolutely necessary. >> >> > > thank you Fons and Joel for pointing to those two plugins. > I've just installed them, and will report how magically they behave for me tonight or tomorrow. > > Rapha?l Hello Rapha?l, Maybe balance-lv2 could be an option: https://github.com/x42/balance.lv2 Best, 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 rennabh at gmail.com Sun Feb 16 13:29:41 2014 From: rennabh at gmail.com (Renato) Date: Sun, 16 Feb 2014 14:29:41 +0100 Subject: [LAU] progressive time stretch In-Reply-To: References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> <20140215172846.01f24f49@debian> <20140215234223.2954a7f1@gmail.com> Message-ID: <20140216142941.750221e6@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 15 Feb 2014 20:18:53 -0500 Paul Davis wrote: > On Sat, Feb 15, 2014 at 5:42 PM, Renato wrote: > > > ----but I can't see how to declare > > Rosegarden as master in the preferences (nor Ardour as slave) - > > > > Session > Properties > Sync .... choose the external clock source Hi, thanks; isn't it strange to have it there and not in the Transport tab in Preferences, where all the other sync options are? Last night after lot of cursing I finally did manage to do what I was after, with Ardour and klick. The two things I was doing wrong were having Ardour set as Jack Master (because I was looking for that option in the Preferences) and not understanding how klick works. Basically, if you have a tempo map file like this firstpart: 10 120 secondpart: 2 120 4 120-135 lastpart: 10 135 you just have to run "klick -T -f tempo_map_file" and, whenever in hydrogen (for example) you'll move the playhead to measure 13, it will start ramping up the tempo to 135 in 4 measures. My misunderstanding was that it would start counting measures from the moment you executed the klick command, instead like this of course it's much better, you can even put that command in your non-session and forget about it thanks to all that helped me :=) renato -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.22 (GNU/Linux) iQEcBAEBAgAGBQJTAL1IAAoJEBz6xFdttjrfke4IAJJkxXNtoZkEA8vEEzpQgiQL y16e22QpaaICeRplqNu7p5/N850SgOVqTOESBMSaa5BTJuQTr2cUskHOA+3HScUv GnbKy4e6MQp2j2OSt3NLXbpWmXaD9IXXQ4VRpPjvhQXElYW4VJXAev5vz/kIih7I FxkUzUbgez9mgcxeQRDafa5x3UptHix6hzGQ7DyPfV8TVPWTGWVzJboLJprT6i0x /S1tuhnrWbClYrWP+Q7rxltBV9T0voIy6OY4CSwzTZRAEfiagl/jdiAEFYpRXqpy xdLh2W5UsLS5y0f6B5KipkGFDWpZOoq6T5nhb9zn6leEOwkJFi8rH9bFz4/+uMo= =Gtnt -----END PGP SIGNATURE----- From paul at linuxaudiosystems.com Sun Feb 16 14:06:17 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 16 Feb 2014 09:06:17 -0500 Subject: [LAU] progressive time stretch In-Reply-To: <20140216142941.750221e6@gmail.com> References: <20140214185843.3dac8e7a@gmail.com> <20140214235704.661d0e49@gmail.com> <1860410.xT5MlFYSHG@edhp> <20140215181524.114b376d@gmail.com> <20140215172846.01f24f49@debian> <20140215234223.2954a7f1@gmail.com> <20140216142941.750221e6@gmail.com> Message-ID: On Sun, Feb 16, 2014 at 8:29 AM, Renato wrote: > > > Session > Properties > Sync .... choose the external clock source > > Hi, thanks; isn't it strange to have it there and not in the Transport > tab in Preferences, where all the other sync options are? > the choice of external clock is considered to be session specific, not global to ardour. in some sessions you might sync to LTC, on others to MTC, on others to JACK and on others just use the internal clock. thus, it is not a global preference and so (like all other session properties) it lives in Session > Properties, not in Edit > Preferences. --p -------------- next part -------------- An HTML attachment was scrubbed... URL: From egor.sanin at gmail.com Sun Feb 16 15:03:20 2014 From: egor.sanin at gmail.com (Egor Sanin) Date: Sun, 16 Feb 2014 10:03:20 -0500 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: This is really great, thanks a lot for sharing! It looks like you guys had a lot of fun. On 2/16/14, Moshe Werner wrote: > Another song from the event. > > https://www.youtube.com/watch?v=AJfKyThzHEg > > Enjoy > > Moshe > > > On Mon, Nov 18, 2013 at 12:10 AM, Moshe Werner wrote: > >> Hi all, >> >> this is the first song released from a small live show that I recorded >> recently. >> The Band is "Shivat Zion", an Israeli Reggae band. >> >> https://www.youtube.com/watch?v=KZS57xoBET4 >> >> Everything was done with Linux software. >> >> Software I used: >> >> Arch Linux >> Ardour 3 >> Calf Lv2 effects (several) >> GxZita_reverb (which I just recently discovered, and what should I >> say... what a sweeet sounding reverb, new favorite) >> Also played around with the new meters.lv2 bundle, I think though that I >> didn't get the full idea of R128 yet. >> Invada Dynamics processing >> TAP >> >> >> Everything regarding the Video work was done by my brother Michael. >> He used KDEnlive to edit the video, also on an Arch machine. >> >> Hope you enjoy. >> >> Cheers, >> >> Moshe >> >> >> >> > From moshwe at gmail.com Sun Feb 16 15:23:15 2014 From: moshwe at gmail.com (Moshe Werner) Date: Sun, 16 Feb 2014 17:23:15 +0200 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: We did! ;) -------------- next part -------------- An HTML attachment was scrubbed... URL: From brendan.jones.it at gmail.com Sun Feb 16 18:21:16 2014 From: brendan.jones.it at gmail.com (Brendan Jones) Date: Sun, 16 Feb 2014 19:21:16 +0100 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: <5301019C.4040800@gmail.com> On 02/16/2014 12:04 PM, Moshe Werner wrote: > Another song from the event. > > https://www.youtube.com/watch?v=AJfKyThzHEg > > Enjoy > > Moshe > > > On Mon, Nov 18, 2013 at 12:10 AM, Moshe Werner > wrote: > > Hi all, > > this is the first song released from a small live show that I > recorded recently. > The Band is "Shivat Zion", an Israeli Reggae band. > > https://www.youtube.com/watch?v=KZS57xoBET4 > > Everything was done with Linux software. > > Software I used: > > Arch Linux > Ardour 3 > Calf Lv2 effects (several) > GxZita_reverb (which I just recently discovered, and what should I > say... what a sweeet sounding reverb, new favorite) > Also played around with the new meters.lv2 bundle, I think though > that I didn't get the full idea of R128 yet. > Invada Dynamics processing > TAP > > > Everything regarding the Video work was done by my brother Michael. > He used KDEnlive to edit the video, also on an Arch machine. > > Hope you enjoy. > I did - so many smiles. What festival? From willgodfrey at musically.me.uk Sun Feb 16 19:19:27 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sun, 16 Feb 2014 19:19:27 +0000 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: <20140216191927.24538f09@debian> On Sun, 16 Feb 2014 13:04:10 +0200 Moshe Werner wrote: > Another song from the event. > > https://www.youtube.com/watch?v=AJfKyThzHEg > > Enjoy > > Moshe > > > On Mon, Nov 18, 2013 at 12:10 AM, Moshe Werner wrote: > > > Hi all, > > > > this is the first song released from a small live show that I recorded > > recently. > > The Band is "Shivat Zion", an Israeli Reggae band. > > > > https://www.youtube.com/watch?v=KZS57xoBET4 > > > > Everything was done with Linux software. > > > > Software I used: > > > > Arch Linux > > Ardour 3 > > Calf Lv2 effects (several) > > GxZita_reverb (which I just recently discovered, and what should I > > say... what a sweeet sounding reverb, new favorite) > > Also played around with the new meters.lv2 bundle, I think though that I > > didn't get the full idea of R128 yet. > > Invada Dynamics processing > > TAP > > > > > > Everything regarding the Video work was done by my brother Michael. > > He used KDEnlive to edit the video, also on an Arch machine. > > > > Hope you enjoy. > > > > Cheers, > > > > Moshe Really enjoyed this. 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 moshwe at gmail.com Sun Feb 16 19:51:04 2014 From: moshwe at gmail.com (Moshe Werner) Date: Sun, 16 Feb 2014 21:51:04 +0200 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: <5301019C.4040800@gmail.com> References: <5301019C.4040800@gmail.com> Message-ID: On 16 ?Feb 2014 20:21, "Brendan Jones" wrote: > > On 02/16/2014 12:04 PM, Moshe Werner wrote: >> >> Another song from the event. >> >> https://www.youtube.com/watch?v=AJfKyThzHEg >> >> Enjoy >> >> Moshe >> >> >> On Mon, Nov 18, 2013 at 12:10 AM, Moshe Werner > > wrote: >> >> Hi all, >> >> this is the first song released from a small live show that I >> recorded recently. >> The Band is "Shivat Zion", an Israeli Reggae band. >> >> https://www.youtube.com/watch?v=KZS57xoBET4 >> >> Everything was done with Linux software. >> >> Software I used: >> >> Arch Linux >> Ardour 3 >> Calf Lv2 effects (several) >> GxZita_reverb (which I just recently discovered, and what should I >> say... what a sweeet sounding reverb, new favorite) >> Also played around with the new meters.lv2 bundle, I think though >> that I didn't get the full idea of R128 yet. >> Invada Dynamics processing >> TAP >> >> >> Everything regarding the Video work was done by my brother Michael. >> He used KDEnlive to edit the video, also on an Arch machine. >> >> Hope you enjoy. >> > > I did - so many smiles. What festival? > It was self organized by my dad, my brother and myself. It took place in my parents garden. -------------- next part -------------- An HTML attachment was scrubbed... URL: From beatleboy07 at gmail.com Sun Feb 16 22:48:38 2014 From: beatleboy07 at gmail.com (Clifford Dunn) Date: Sun, 16 Feb 2014 14:48:38 -0800 Subject: [LAU] Audio and Bluetooth Message-ID: Hi List, Since I discovered that there is a great deal of interference happening when I have both wifi and USB audio running, it certainly has been a drastic improvement to turn that off when I'm working. I'm currently building a foot pedal controller. I have an Arduino Diecimila that is going to transmit the simple on/off as well as continuous controller info. My thought is to make it a bluetooth device and have wireless communication with my computer. Since I currently have no other bluetooth devices to test it out, does anyone have any information on how much bluetooth has the potential to interfere with my audio? Is it better to simply keep it as a bluetooth device? Thanks again! Cliff Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn From harryhaaren at gmail.com Mon Feb 17 00:22:42 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Mon, 17 Feb 2014 00:22:42 +0000 Subject: [LAU] Audio and Bluetooth In-Reply-To: References: Message-ID: On Sun, Feb 16, 2014 at 10:48 PM, Clifford Dunn wrote: > I'm currently building a foot pedal controller. I have an Arduino > Diecimila that is going to transmit the simple on/off as well as > continuous controller info. Cool, and nice project. > My thought is to make it a bluetooth > device and have wireless communication with my computer. I'd advise against it: I know perhaps the "wireless" thing is nice on stage, but in the end a cable you can (usually anway!) rely on, while with bluetooth, I wouldn't be too sure. I'll suggest two options: A) Get a Arduino Uno (so you have the flashable USB chip), and make it appear as a class compliant USB MIDI device. ALSA will pick it up, and automatically list it as a MIDI I/O device. Done. B) Use a hardware MIDI output from the Arduino: setting the serial baudrate to 31250 (midi baud rate), and writing the bytes you want using Serial.write() does the job. The hardware MIDI output is very simple: http://arduino.cc/en/uploads/Tutorial/MIDI_bb.png Depending on if you interface has hardware MIDI I/O, B might be OK, and saves buying an Arduino Uno. That said, the simplicity of just plug & play USB MIDI anywhere is awesome! HTH, -Harry PS: I have python script to read serial data using the "PySerial" module, and turn that into ALSA MIDI somewhere... leftovers from a similar project :) That allows using the Decimilia as a USB MIDI device, so long as the python script is running. Note that its obviously not ideal: with Python being rubbish for speed, and its a very non-portable solution since the Py-ALSA-MIDI / PySerial modules aren't common. -------------- next part -------------- An HTML attachment was scrubbed... URL: From pshirkey at boosthardware.com Mon Feb 17 01:22:17 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Mon, 17 Feb 2014 12:22:17 +1100 (EST) Subject: [LAU] Audio and Bluetooth In-Reply-To: References: Message-ID: <55151.86.105.95.182.1392600137.squirrel@boosthardware.com> On Mon, February 17, 2014 9:48 am, Clifford Dunn wrote: > Hi List, > > Since I discovered that there is a great deal of interference > happening when I have both wifi and USB audio running, it certainly > has been a drastic improvement to turn that off when I'm working. > > I'm currently building a foot pedal controller. I have an Arduino > Diecimila that is going to transmit the simple on/off as well as > continuous controller info. My thought is to make it a bluetooth > device and have wireless communication with my computer. Since I > currently have no other bluetooth devices to test it out, does anyone > have any information on how much bluetooth has the potential to > interfere with my audio? Is it better to simply keep it as a bluetooth > device? > Pulse Audio has pretty advanced bluetooth support via the bluez API. One of the "features" for professional audio production that is currently missing is a way to use a bluetooth controller with JACK transport. AFAIK no one has taken on that job yet. I am very interested to hear if you make any progress with that. It should be possible to route the signals from PA through to an OSC or MIDI controller or maybe we just need a plugin for bridging directly between PA and JACK. -- Patrick Shirkey Boost Hardware Ltd From linuxaudio at cryptomys.de Mon Feb 17 09:34:21 2014 From: linuxaudio at cryptomys.de (Martin Homuth-Rosemann) Date: Mon, 17 Feb 2014 01:34:21 -0800 (PST) Subject: [LAU] Audio and Bluetooth In-Reply-To: References: Message-ID: <1392629661750-89480.post@n7.nabble.com> Harry van Haaren wrote > ... > I'll suggest two options: > A) Get a Arduino Uno (so you have the flashable USB chip), and make it > appear as a class compliant USB MIDI device. ALSA will pick it up, and > automatically list it as a MIDI I/O device. Done. > > B) Use a hardware MIDI output from the Arduino: setting the serial > baudrate > to 31250 (midi baud rate), and writing the bytes you want using > Serial.write() does the job. The hardware MIDI output is very simple: > http://arduino.cc/en/uploads/Tutorial/MIDI_bb.png > ... Hi, there's a third option, use the VUSB software USB emulation with a simple AVR chip. I used this SW to build a simple MIDI I/O device that works fine with Linux: http://forums.obdev.at/viewtopic.php?f=8&t=1352 There are already some projects using my MIDI code: http://www.obdev.at/products/vusb/prjall.html Ciao, Martin -- View this message in context: http://linux-audio.4202.n7.nabble.com/Audio-and-Bluetooth-tp89477p89480.html Sent from the linux-audio-user mailing list archive at Nabble.com. From jonetsu at teksavvy.com Mon Feb 17 11:39:26 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Mon, 17 Feb 2014 06:39:26 -0500 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: Message-ID: <20140217063926.3f73286d@mistral> On Mon, 18 Nov 2013 00:10:44 +0200, Moshe Werner wrote : Hi, > this is the first song released from a small live show that I recorded > recently. > The Band is "Shivat Zion", an Israeli Reggae band. > > https://www.youtube.com/watch?v=KZS57xoBET4 That must be the more jazzy piece of the band! Do you have a translation of the lyrics ? From jonetsu at teksavvy.com Mon Feb 17 11:43:13 2014 From: jonetsu at teksavvy.com (jonetsu at teksavvy.com) Date: Mon, 17 Feb 2014 06:43:13 -0500 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: References: <1869357.0PMbqxoHEm@dovidhalevi> <1384811206.60191.YahooMailNeo@web122602.mail.ne1.yahoo.com> Message-ID: <20140217064313.73d3cc57@mistral> On Mon, 18 Nov 2013 23:53:54 +0200, Moshe Werner wrote : > You forgot to mention th Oud:) That's a fancy one. From moshwe at gmail.com Mon Feb 17 11:48:36 2014 From: moshwe at gmail.com (Moshe Werner) Date: Mon, 17 Feb 2014 13:48:36 +0200 Subject: [LAU] Music made with Linux "Naale Lezion - Shivat Zion" In-Reply-To: <20140217063926.3f73286d@mistral> References: <20140217063926.3f73286d@mistral> Message-ID: Sorry but I dont have the translation:) On Mon, Feb 17, 2014 at 1:39 PM, jonetsu at teksavvy.com wrote: > On Mon, 18 Nov 2013 00:10:44 +0200, > Moshe Werner wrote : > > Hi, > > > this is the first song released from a small live show that I recorded > > recently. > > The Band is "Shivat Zion", an Israeli Reggae band. > > > > https://www.youtube.com/watch?v=KZS57xoBET4 > > That must be the more jazzy piece of the band! Do you have a > translation of the lyrics ? > > > > _______________________________________________ > 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 angelv at iac.es Mon Feb 17 16:27:51 2014 From: angelv at iac.es (Angel de Vicente) Date: Mon, 17 Feb 2014 16:27:51 +0000 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? Message-ID: Hi, I'm completely new to Common Music (http://sourceforge.net/projects/commonmusic/), so I'm hoping somebody can give me a hand here. Before I go to the effort of starting to learn it, I wanted to know if it is possible to somehow import audio files with it and treat them as objects (say for example that I have a number of noise sounds and that I want to generate some music by randomly playing them for a length of time)? A quick look at the Common Music webpage and some of the examples there didn't clear this up for me. At the same time, what would be the best way to start learning how to use it? From what I read, the book http://www.amazon.com/Notes-Metalevel-Introduction-Computer-Composition/dp/9026519753/ is a very good starting point, but apparently all the examples, etc. are for an old version of Common Music (CM2), so I'm not sure if whatever one can learn from that book will be of much use with the current version of CM3. Any pointers welcome. Thanks a lot, -- ??ngel de Vicente http://www.iac.es/galeria/angelv/ --------------------------------------------------------------------------------------------- ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci?n de Datos, acceda a http://www.iac.es/disclaimer.php WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en From dlphillips at woh.rr.com Mon Feb 17 22:23:29 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 17 Feb 2014 17:23:29 -0500 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: References: Message-ID: <53028BE1.9000909@woh.rr.com> On 02/17/2014 11:27 AM, Angel de Vicente wrote: > Hi, > > I'm completely new to Common Music > (http://sourceforge.net/projects/commonmusic/), so I'm hoping somebody > can give me a hand here. Before I go to the effort of starting to learn > it, I wanted to know if it is possible to somehow import audio files > with it and treat them as objects (say for example that I have a number > of noise sounds and that I want to generate some music by randomly > playing them for a length of time)? A quick look at the Common Music > webpage and some of the examples there didn't clear this up for me. Hi Angel, CM is now part of the Grace system. I've used it for many years, primarily as a generator for Csound scores. Alas, I've never tried random file playback with it. I suppose it could be done with Grace, but I'd probably use SuperCollider3 for that purpose. That said, Grace is fantastic, so I've cc'd this reply to Rick Taube, the author of CM/Grace, perhaps he can advise you re: your intended use. > At the same time, what would be the best way to start learning how to > use it? From what I read, the book > http://www.amazon.com/Notes-Metalevel-Introduction-Computer-Composition/dp/9026519753/ > is a very good starting point, but apparently all the examples, etc. are > for an old version of Common Music (CM2), so I'm not sure if whatever > one can learn from that book will be of much use with the current > version of CM3. > > Any pointers welcome. Thanks a lot, > At this time the best way to learn is by running the examples and studying their code. The SAL language - Bill Schottstaedt's S7 Scheme implementation - is very easy to learn, the examples are musical and inspiring. Best, dp From rm at mh-freiburg.de Mon Feb 17 23:01:36 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Tue, 18 Feb 2014 00:01:36 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: <53028BE1.9000909@woh.rr.com> References: <53028BE1.9000909@woh.rr.com> Message-ID: <20140217225429.M18792@mh-freiburg.de> On Mon, 17 Feb 2014 17:23:29 -0500, Dave Phillips wrote > [...] > At this time the best way to learn is by running the examples and > studying their code. The SAL language - Bill Schottstaedt's S7 > Scheme implementation - is very easy to learn, the examples are > musical and inspiring. Isn't this slightly misleading? There is SAL, an Algol-like language and there is S7, a dialect of scheme. Cheers, RalfD > Best, > > dp > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- R. Mattes - Hochschule fuer Musik Freiburg rm at inm.mh-freiburg.de From dlphillips at woh.rr.com Tue Feb 18 01:08:37 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Mon, 17 Feb 2014 20:08:37 -0500 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: <20140217225429.M18792@mh-freiburg.de> References: <53028BE1.9000909@woh.rr.com> <20140217225429.M18792@mh-freiburg.de> Message-ID: <5302B295.8060601@woh.rr.com> On 02/17/2014 06:01 PM, R. Mattes wrote: > On Mon, 17 Feb 2014 17:23:29 -0500, Dave Phillips wrote >> [...] >> At this time the best way to learn is by running the examples and >> studying their code. The SAL language - Bill Schottstaedt's S7 >> Scheme implementation - is very easy to learn, the examples are >> musical and inspiring. > Isn't this slightly misleading? There is SAL, an Algol-like language > and there is S7, a dialect of scheme. Ralf's correct, my bad. Best, dp > Cheers, RalfD > >> Best, >> >> dp >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > -- > R. Mattes - > Hochschule fuer Musik Freiburg > rm at inm.mh-freiburg.de > > From bruviaro at scu.edu Tue Feb 18 03:28:02 2014 From: bruviaro at scu.edu (Bruno Ruviaro) Date: Mon, 17 Feb 2014 19:28:02 -0800 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: References: Message-ID: I'd suggest taking a look at SuperCollider as well. It would be very easy to accomplish what you want with SC. B On Monday, February 17, 2014, Angel de Vicente wrote: > Hi, > > I'm completely new to Common Music > (http://sourceforge.net/projects/commonmusic/), so I'm hoping somebody > can give me a hand here. Before I go to the effort of starting to learn > it, I wanted to know if it is possible to somehow import audio files > with it and treat them as objects (say for example that I have a number > of noise sounds and that I want to generate some music by randomly > playing them for a length of time)? A quick look at the Common Music > webpage and some of the examples there didn't clear this up for me. > > At the same time, what would be the best way to start learning how to > use it? From what I read, the book > > http://www.amazon.com/Notes-Metalevel-Introduction-Computer-Composition/dp/9026519753/ > is a very good starting point, but apparently all the examples, etc. are > for an old version of Common Music (CM2), so I'm not sure if whatever > one can learn from that book will be of much use with the current > version of CM3. > > Any pointers welcome. Thanks a lot, > -- > ?ngel de Vicente > http://www.iac.es/galeria/angelv/ > > --------------------------------------------------------------------------------------------- > ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci?n de > Datos, acceda a http://www.iac.es/disclaimer.php > WARNING: For more information on privacy and fulfilment of the Law > concerning the Protection of Data, consult > http://www.iac.es/disclaimer.php?lang=en > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From beatleboy07 at gmail.com Tue Feb 18 07:41:21 2014 From: beatleboy07 at gmail.com (Clifford Dunn) Date: Mon, 17 Feb 2014 23:41:21 -0800 Subject: [LAU] Audio and Bluetooth In-Reply-To: <1392629661750-89480.post@n7.nabble.com> References: <1392629661750-89480.post@n7.nabble.com> Message-ID: I'm surprised that people are using the Arduino devices for MIDI. Has no one been able to get the full resolution out of the Analog In pins? For the simple switches, I don't see it being a problem, but I'd rather have more than 128 numbers for continuous controllers. Has anyone had success with that? (I suppose we're no longer talking about Linux audio.) Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn On Mon, Feb 17, 2014 at 1:34 AM, Martin Homuth-Rosemann wrote: > Harry van Haaren wrote >> ... >> I'll suggest two options: >> A) Get a Arduino Uno (so you have the flashable USB chip), and make it >> appear as a class compliant USB MIDI device. ALSA will pick it up, and >> automatically list it as a MIDI I/O device. Done. >> >> B) Use a hardware MIDI output from the Arduino: setting the serial >> baudrate >> to 31250 (midi baud rate), and writing the bytes you want using >> Serial.write() does the job. The hardware MIDI output is very simple: >> http://arduino.cc/en/uploads/Tutorial/MIDI_bb.png >> ... > > Hi, there's a third option, use the VUSB software USB emulation with a > simple AVR chip. I used this SW to build a simple MIDI I/O device that works > fine with Linux: > http://forums.obdev.at/viewtopic.php?f=8&t=1352 > There are already some projects using my MIDI code: > http://www.obdev.at/products/vusb/prjall.html > > Ciao, Martin > > > > -- > View this message in context: http://linux-audio.4202.n7.nabble.com/Audio-and-Bluetooth-tp89477p89480.html > Sent from the linux-audio-user mailing list archive at Nabble.com. > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From angelv at iac.es Tue Feb 18 09:13:22 2014 From: angelv at iac.es (Angel de Vicente) Date: Tue, 18 Feb 2014 09:13:22 +0000 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: (Bruno Ruviaro's message of "Mon, 17 Feb 2014 19:28:02 -0800") References: Message-ID: Hi, thanks Dave and Bruno for the replies. I was keen on Common Music, having programmed in Lisp for a number of years, but I did have a look at SuperCollider, and certainly my use case was trivial with it. Time to experiment. Thanks a lot, ??ngel Bruno Ruviaro writes: > I'd suggest taking a look at SuperCollider as well. It would be very easy to accomplish what you want with SC.?? > > B > > On Monday, February 17, 2014, Angel de Vicente wrote: > > Hi, > > I'm completely new to Common Music > (http://sourceforge.net/projects/commonmusic/), so I'm hoping somebody > can give me a hand here. Before I go to the effort of starting to learn > it, I wanted to know if it is possible to somehow import audio files > with it and treat them as objects (say for example that I have a number > of noise sounds and that I want to generate some music by randomly > playing them for a length of time)? A quick look at the Common Music > webpage and some of the examples there didn't clear this up for me. > > At the same time, what would be the best way to start learning how to > use it? From what I read, the book > http://www.amazon.com/Notes-Metalevel-Introduction-Computer-Composition/dp/9026519753/ > is a very good starting point, but apparently all the examples, etc. are > for an old version of Common Music (CM2), so I'm not sure if whatever > one can learn from that book will be of much use with the current > version of CM3. > > Any pointers welcome. Thanks a lot, > -- > ??ngel de Vicente > http://www.iac.es/galeria/angelv/ > --------------------------------------------------------------------------------------------- > ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci??n de Datos, acceda a http://www.iac.es/disclaimer.php > WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en > -- ??ngel de Vicente http://www.iac.es/galeria/angelv/ --------------------------------------------------------------------------------------------- ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci?n de Datos, acceda a http://www.iac.es/disclaimer.php WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en From dlphillips at woh.rr.com Tue Feb 18 10:45:36 2014 From: dlphillips at woh.rr.com (Dave Phillips) Date: Tue, 18 Feb 2014 05:45:36 -0500 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: References: Message-ID: <530339D0.4070703@woh.rr.com> On 02/18/2014 04:13 AM, Angel de Vicente wrote: > ...having programmed in Lisp for a number of years... Then you might also want to check out OpenMusic : http://repmus.ircam.fr/openmusic/linux Best, dp From blablack at gmail.com Tue Feb 18 12:55:27 2014 From: blablack at gmail.com (=?ISO-8859-1?Q?Aur=E9lien_Leblond?=) Date: Tue, 18 Feb 2014 12:55:27 +0000 Subject: [LAU] b-step - sequencer linux native vst-plugin Message-ID: Saw that today on twitter b-step - sequencer vst-plugin http://b-step.monoplugs.com/ It comes with a linux native vst. I didn't get the chance to test the demo, but the video sounds nice... Aur?lien From jeremy at autostatic.com Tue Feb 18 13:13:57 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 18 Feb 2014 14:13:57 +0100 Subject: [LAU] b-step - sequencer linux native vst-plugin In-Reply-To: References: Message-ID: <53035C95.1090800@autostatic.com> On 02/18/2014 01:55 PM, Aur?lien Leblond wrote: > Saw that today on twitter > > b-step - sequencer vst-plugin http://b-step.monoplugs.com/ > > It comes with a linux native vst. > I didn't get the chance to test the demo, but the video sounds nice... > > Aur?lien Hi Aur?lien, I've briefly tested this plugin but didn't really dig the bass guitar orientation. Also, with the advent of the LV2 versions of the QMidiArp modules I think I prefer those. The plugin looks great though. It's open source big brother Cythar is worth checking out too. Best, 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 martin.peach at sympatico.ca Tue Feb 18 13:34:45 2014 From: martin.peach at sympatico.ca (Martin Peach) Date: Tue, 18 Feb 2014 08:34:45 -0500 Subject: [LAU] Audio and Bluetooth In-Reply-To: References: <1392629661750-89480.post@n7.nabble.com> Message-ID: On 2014-02-18 02:41, Clifford Dunn wrote: > I'm surprised that people are using the Arduino devices for MIDI. Has > no one been able to get the full resolution out of the Analog In pins? > For the simple switches, I don't see it being a problem, but I'd > rather have more than 128 numbers for continuous controllers. Has > anyone had success with that? (I suppose we're no longer talking about > Linux audio.) You can use two controller numbers separated by 32 to send the two halves of the value, so controller 2 could send the high 7 bits and controller 34 the low 3 to get the full 10 bits from the Arduino ADC. I have built a pedal using a PIC that does that. With Arduino you can also SLIP-encode OSC packets and send them through the USB serial port at a higher speed than MIDI. Martin From bjb-linux-audio-user at deus.net Tue Feb 18 15:43:26 2014 From: bjb-linux-audio-user at deus.net (Ben Bell) Date: Tue, 18 Feb 2014 15:43:26 +0000 Subject: [LAU] Arduino MIDI controllers In-Reply-To: References: Message-ID: <20140218154329.2D88662040@lists.linuxaudio.org> On Sun, Feb 16, 2014 at 02:48:38PM -0800, Clifford Dunn wrote: > I'm currently building a foot pedal controller. I have an Arduino > Diecimila that is going to transmit the simple on/off as well as > continuous controller info. My thought is to make it a bluetooth > device and have wireless communication with my computer. Since I Interesting. I've got part-completed project based around an Arduino which is a 3D (well, 2D plus something approximating pressure) midi CC generator. It currently works pretty well but I ran out of time top work on it when trying to deal with some timing issues in my midi merge code (it occasionally drops an event when there's a lot of traffic). I'd be interested to hear and see what other people are up to in this area :) Slightly OT as it's not inherently a Linux thing, I suppose. From lievenmoors at gmail.com Tue Feb 18 17:46:55 2014 From: lievenmoors at gmail.com (Lieven Moors) Date: Tue, 18 Feb 2014 18:46:55 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: <5302B295.8060601@woh.rr.com> References: <53028BE1.9000909@woh.rr.com> <20140217225429.M18792@mh-freiburg.de> <5302B295.8060601@woh.rr.com> Message-ID: <20140218174654.GB620@satellite> On Mon, Feb 17, 2014 at 08:08:37PM -0500, Dave Phillips wrote: > > On 02/17/2014 06:01 PM, R. Mattes wrote: > >On Mon, 17 Feb 2014 17:23:29 -0500, Dave Phillips wrote > >>[...] > >>At this time the best way to learn is by running the examples and > >>studying their code. The SAL language - Bill Schottstaedt's S7 > >>Scheme implementation - is very easy to learn, the examples are > >>musical and inspiring. > >Isn't this slightly misleading? There is SAL, an Algol-like language > >and there is S7, a dialect of scheme. > > Ralf's correct, my bad. > You should check out CLM (Common Lisp Music) as well, written by Bill Schottstaedt. You used to be able to combine CLM with CM. CLM is much more audio-oriented. lieven From rmnmichon at gmail.com Tue Feb 18 19:27:19 2014 From: rmnmichon at gmail.com (Romain Michon) Date: Tue, 18 Feb 2014 11:27:19 -0800 Subject: [LAU] Workshop on Audio Plug-Ins Design in Faust Message-ID: Hi Folks, I'll be giving a workshop on Audio Plug-Ins Design in Faust this summer at the Center for Computer Research in Music and Acoustics (Stanford University, USA) this summer (Mon, 07/07/2014 - Fri, 07/11/2014). More informations are available on this page: https://ccrma.stanford.edu/workshops/faust-workshop-2014 You can also contact me directly if you have specific questions: rmichon at ccrma.stanford.edu Hope to see you there! Cheers, -- Romain Michon PhD Candidate Center for Computer Research in Music and Acoustics Stanford Universityhttp://ccrma.stanford.edu/~rmichon -------------- next part -------------- An HTML attachment was scrubbed... URL: From silvain at freeshell.de Tue Feb 18 23:25:25 2014 From: silvain at freeshell.de (F. Silvain) Date: Wed, 19 Feb 2014 00:25:25 +0100 (CET) Subject: [LAU] Jconvolver: Can't initialise engine. Message-ID: <1402190023001.14657@freeshell.de> Hey hey, with jconvolver I get exactly that warning: Can't initialise engine. I haven't found any means to get further output. I use jconvolver 0.9.2, just compiled again to make sure, that some updates didn't cause confusion. Compiler is gcc 4.8.2. Any advise is precious. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From fons at linuxaudio.org Tue Feb 18 23:40:39 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 18 Feb 2014 23:40:39 +0000 Subject: [LAU] Jconvolver: Can't initialise engine. In-Reply-To: <1402190023001.14657@freeshell.de> References: <1402190023001.14657@freeshell.de> Message-ID: <20140218234039.GC9991@linuxaudio.org> On Wed, Feb 19, 2014 at 12:25:25AM +0100, F. Silvain wrote: > with jconvolver I get exactly that warning: > Can't initialise engine. > I haven't found any means to get further output. I use jconvolver > 0.9.2, just compiled again to make sure, that some updates didn't > cause confusion. Compiler is gcc 4.8.2. Please provide the config file you try 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 silvain at freeshell.de Tue Feb 18 23:45:13 2014 From: silvain at freeshell.de (F. Silvain) Date: Wed, 19 Feb 2014 00:45:13 +0100 (CET) Subject: [LAU] Jconvolver: Can't initialise engine. In-Reply-To: <20140218234039.GC9991@linuxaudio.org> References: <1402190023001.14657@freeshell.de> <20140218234039.GC9991@linuxaudio.org> Message-ID: <1402190040030.14926@freeshell.de> Fons Adriaensen, Feb 19 2014: ... > Please provide the config file you try to use. The weird.conf supplied with jconvolver and this: *** rev.conf *** /convolver/new 2 2 256 204800 # rev.wav has two channels /impulse/read 1 1 0.1 0 0 0 1 rev.wav /impulse/read 2 2 0.1 0 0 0 2 rev.wav *** end of rev.conf *** Thank you From fons at linuxaudio.org Wed Feb 19 00:10:32 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Wed, 19 Feb 2014 00:10:32 +0000 Subject: [LAU] Jconvolver: Can't initialise engine. In-Reply-To: <1402190040030.14926@freeshell.de> References: <1402190023001.14657@freeshell.de> <20140218234039.GC9991@linuxaudio.org> <1402190040030.14926@freeshell.de> Message-ID: <20140219001031.GD9991@linuxaudio.org> On Wed, Feb 19, 2014 at 12:45:13AM +0100, F. Silvain wrote: > The weird.conf supplied with jconvolver and this: > *** rev.conf *** > /convolver/new 2 2 256 204800 > # rev.wav has two channels > /impulse/read 1 1 0.1 0 0 0 1 rev.wav > /impulse/read 2 2 0.1 0 0 0 2 rev.wav > *** end of rev.conf *** Looks OK. The only thing I can imagine ATM is that your Jack period size is < 16, or > 8192, or not a power of 2. 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 silvain at freeshell.de Wed Feb 19 00:18:38 2014 From: silvain at freeshell.de (F. Silvain) Date: Wed, 19 Feb 2014 01:18:38 +0100 (CET) Subject: [LAU] Jconvolver: Can't initialise engine. In-Reply-To: <1402190040030.14926@freeshell.de> References: <1402190023001.14657@freeshell.de> <20140218234039.GC9991@linuxaudio.org> <1402190040030.14926@freeshell.de> Message-ID: <1402190118001.15511@freeshell.de> Thank you Fons, somehow the JACK period size did get knocked. Sorry for the inconvenience. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From shakti at bayarea.net Wed Feb 19 09:59:14 2014 From: shakti at bayarea.net (Tracey Hytry) Date: Wed, 19 Feb 2014 01:59:14 -0800 Subject: [LAU] Arduino MIDI controllers In-Reply-To: <20140218154329.2D88662040@lists.linuxaudio.org> References: <20140218154329.2D88662040@lists.linuxaudio.org> Message-ID: <20140219015914.ba5bd51d13f11a4045dfa1eb@bayarea.net> If you are interested in these things check out the PJRC site. They sell some pretty inexpensive controllers with built in USB. They also have USB class compliant MIDI libraries if you're programming in the arduino environment. https://www.pjrc.com/teensy/index.html https://www.pjrc.com/teensy/teensyduino.html From email.rafa at gmail.com Thu Feb 20 20:34:06 2014 From: email.rafa at gmail.com (Rafael Vega) Date: Thu, 20 Feb 2014 15:34:06 -0500 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? Message-ID: Hi, has anyone tried a usb sound card with the BBB? I want to connect my guitar to it and run some PD patches so something with low latency would be niiice :) Also wondering how hard it would be to directly connect some ADCs and DACs. Has anyone tried? -- Rafael Vega email.rafa at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jacob01 at gmx.net Thu Feb 20 21:11:15 2014 From: jacob01 at gmx.net (Jacob) Date: Thu, 20 Feb 2014 22:11:15 +0100 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? In-Reply-To: References: Message-ID: <53066F73.7060900@gmx.net> Hi, On 20.02.2014 21:34, Rafael Vega wrote: > > Also wondering how hard it would be to directly connect some ADCs and > DACs. It's probably not too hard. The BBB's AM335x has 2 "Multichannel Audio Serial Ports" each providing serial audio I/O ports which can be used in I2S, SPDIF, AES-3 and other formats. I don't how well this is supported under Linux, but in theory (AFAICS) the Cortex could support 4 x stereo in & 4 x stereo out. Search the web for "bbb" and "i2s". HTH, Jacob From jeremy at autostatic.com Thu Feb 20 22:11:01 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 20 Feb 2014 23:11:01 +0100 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? In-Reply-To: References: Message-ID: <53067D75.7080500@autostatic.com> On 02/20/2014 09:34 PM, Rafael Vega wrote: > Hi, has anyone tried a usb sound card with the BBB? I want to connect my > guitar to it and run some PD patches so something with low latency would be > niiice :) > > Also wondering how hard it would be to directly connect some ADCs and DACs. > Has anyone tried? Hi Rafael, I've owned a BBB for a short while but sold it mainly because I couldn't get any USB interface to work with it. I think this was related to what kind of power adapter was used. Could well be I overlooked something but since I didn't really dig the BBB I didn't investigate any further. Best, 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 martin.peach at sympatico.ca Thu Feb 20 22:14:56 2014 From: martin.peach at sympatico.ca (Martin Peach) Date: Thu, 20 Feb 2014 17:14:56 -0500 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? In-Reply-To: References: Message-ID: On 2014-02-20 15:34, Rafael Vega wrote: > Hi, has anyone tried a usb sound card with the BBB? I want to connect my > guitar to it and run some PD patches so something with low latency would > be niiice :) > I have a Behringer UCA-202 attached to a BBB running Ubuntu. It's running a csound score that implements a variable delay line. Audio is 48kHz stereo 16bit via ALSA, not sure what the latency is exactly but it's small. > Also wondering how hard it would be to directly connect some ADCs and > DACs. Has anyone tried? The PRUSS is very interesting for high speed hardware interfacing, but it seems difficult to integrate with linux, you would need to write kernel drivers to work with ALSA or JACK. Connecting DACs and ADCs using I2C or SPI could work but how to synchronize with the audio system? Martin From ken at restivo.org Thu Feb 20 22:22:45 2014 From: ken at restivo.org (Ken Restivo) Date: Thu, 20 Feb 2014 14:22:45 -0800 Subject: [LAU] Cutting vorbis files and resetting their times? Message-ID: <20140220222245.GA30971@q400a.mobile.restivo.org> I have a feeling I've asked this before, but I don't remember if there ever was an answer. I'm trying to do the following: 1) Cut an Ogg Vorbis file into chunks 2) Reset the start time of each chunk back to zero, and have its end time be the end time of the clip from zero. Thing #1 is very easy; oggz-chop does the job well. But thing #2 seems un-possible due to perhaps some design flaw in vorbis? The start times of vorbis files cut up with oggz-chop or similar tools is broken: it shows a start time of whatever was the time of the clip in the original file. This causes certain players (including Airtime) to lose control of their bladder: they either refuse to play the file or play silence for X number of hours until the start time of the clip. Yeah yeah, I know, I could just convert the file to WAV, then re-encode it. But... I do not want to do that. First of all, it reduces the quality. Secondly, there's something just upsetting my OCD nature, about not being able to do this without re-encoding. Any clues? I don't mind writing some C (or whatever) and wading through docs, if I had some expert advice on how to approach the problem (or at least confirmation that it is indeed possible). It's almost like I'd have to have something that reads the blocks one by one, then calculates the new time, and writes the block out with the new time? Is that a sensible way to do it? -ken From angelv at iac.es Thu Feb 20 23:29:33 2014 From: angelv at iac.es (Angel de Vicente) Date: Thu, 20 Feb 2014 23:29:33 +0000 Subject: [LAU] Cutting vorbis files and resetting their times? In-Reply-To: <20140220222245.GA30971@q400a.mobile.restivo.org> (Ken Restivo's message of "Thu, 20 Feb 2014 14:22:45 -0800") References: <20140220222245.GA30971@q400a.mobile.restivo.org> Message-ID: Ken Restivo writes: > I have a feeling I've asked this before, but I don't remember if there ever was an answer. > > I'm trying to do the following: > > 1) Cut an Ogg Vorbis file into chunks > 2) Reset the start time of each chunk back to zero, and have its end time be the end time of the clip from zero. > > Thing #1 is very easy; oggz-chop does the job well. > > But thing #2 seems un-possible due to perhaps some design flaw in vorbis? vcut (http://www.linuxfromscratch.org/blfs/view/svn/multimedia/vorbistools.html) does fix the times of the cut pieces to start from zero. Cheers, -- ??ngel de Vicente http://www.iac.es/galeria/angelv/ --------------------------------------------------------------------------------------------- ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci?n de Datos, acceda a http://www.iac.es/disclaimer.php WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en From ken at restivo.org Fri Feb 21 01:30:57 2014 From: ken at restivo.org (Ken Restivo) Date: Thu, 20 Feb 2014 17:30:57 -0800 Subject: [LAU] Cutting vorbis files and resetting their times? In-Reply-To: References: <20140220222245.GA30971@q400a.mobile.restivo.org> Message-ID: <20140221013057.GA2070@q400a.mobile.restivo.org> On Thu, Feb 20, 2014 at 11:29:33PM +0000, Angel de Vicente wrote: > Ken Restivo writes: > > > I have a feeling I've asked this before, but I don't remember if there ever was an answer. > > > > I'm trying to do the following: > > > > 1) Cut an Ogg Vorbis file into chunks > > 2) Reset the start time of each chunk back to zero, and have its end time be the end time of the clip from zero. > > > > Thing #1 is very easy; oggz-chop does the job well. > > > > But thing #2 seems un-possible due to perhaps some design flaw in vorbis? > > vcut > (http://www.linuxfromscratch.org/blfs/view/svn/multimedia/vorbistools.html) > does fix the times of the cut pieces to start from zero. > Perfect! Thanks! -ken From csanchezgs at gmail.com Fri Feb 21 08:35:03 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 21 Feb 2014 09:35:03 +0100 Subject: [LAU] APC devices for Audio Message-ID: Hello dear all. Lately I've found this devices by APC that seem quite interesting to me given my interest and projects around Raspberry PI and Linux Audio: http://apc.io/products/rock/ http://apc.io/products/8750a/ I love this presentation as a book,, http://apc.io/products/paper/ ... and that, presenting a device in a nice and not PC-like way, is something I've been thinkinf of for a long time, but I couldn't do it myself without a lot of building and DIY. Does anybody have or tried one as it is? And with Linux? -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From nettings at stackingdwarves.net Fri Feb 21 09:42:33 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Fri, 21 Feb 2014 10:42:33 +0100 Subject: [LAU] Cutting vorbis files and resetting their times? In-Reply-To: <20140220222245.GA30971@q400a.mobile.restivo.org> References: <20140220222245.GA30971@q400a.mobile.restivo.org> Message-ID: <53071F89.3090800@stackingdwarves.net> On 02/20/2014 11:22 PM, Ken Restivo wrote: > 1) Cut an Ogg Vorbis file into chunks 2) Reset the start time of each > chunk back to zero, and have its end time be the end time of the clip > from zero. > Yeah yeah, I know, I could just convert the file to WAV, then > re-encode it. But... I do not want to do that. First of all, it > reduces the quality. i may be wrong, but i'd say the quality loss is pretty much negligible, as long as you use the same decoder and quality settings for re-encoding. a few years ago, i experimented with gradual degradation of repeatedly encoded and re-encoded files for artistic purposes (using an mp3 codec), and the amount of "degradation" was disappointing, to say the least... i guess what happens is the psychoacoustic model removes information, when you decode, the new wav has all those simplifications in it, and the next encoder down the line has nothing left to do. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From anders.vinjar at bek.no Fri Feb 21 11:57:34 2014 From: anders.vinjar at bek.no (anders.vinjar at bek.no) Date: Fri, 21 Feb 2014 12:57:34 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? References: Message-ID: <87y514smoh.fsf@bek.no> A> Before I go to the effort of starting to learn it, I wanted to A> know if it is possible to somehow import audio files with it... Hi Angel. This should be rather straightforward w. CM3/Grace. When starting Grace (CM3's front-end), you get access to piles of examples for working with soundfiles, processing or synthesis. Try looking in the menu "Audio"->"Instrument Browser" (or hit Ctrl-i), and load something which looks close. A> At the same time, what would be the best way to start learning A> how to use it? To get started, CM3/Grace has extensive built-in help, w. interactive examples and tutorials, both using SAL and Scheme, and links to further documentation around on the web. Cheers, -anders From email.rafa at gmail.com Fri Feb 21 15:59:26 2014 From: email.rafa at gmail.com (Rafael Vega) Date: Fri, 21 Feb 2014 10:59:26 -0500 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? In-Reply-To: References: Message-ID: Thanks Martin! I will pick up one of those Beringers :) On Thu, Feb 20, 2014 at 5:14 PM, Martin Peach wrote: > On 2014-02-20 15:34, Rafael Vega wrote: > >> Hi, has anyone tried a usb sound card with the BBB? I want to connect my >> guitar to it and run some PD patches so something with low latency would >> be niiice :) >> >> > I have a Behringer UCA-202 attached to a BBB running Ubuntu. It's running > a csound score that implements a variable delay line. Audio is 48kHz stereo > 16bit via ALSA, not sure what the latency is exactly but it's small. > > > Also wondering how hard it would be to directly connect some ADCs and >> DACs. Has anyone tried? >> > > The PRUSS is very interesting for high speed hardware interfacing, but it > seems difficult to integrate with linux, you would need to write kernel > drivers to work with ALSA or JACK. Connecting DACs and ADCs using I2C or > SPI could work but how to synchronize with the audio system? > > > Martin > -- Rafael Vega email.rafa at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From csanchezgs at gmail.com Fri Feb 21 18:39:06 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 21 Feb 2014 19:39:06 +0100 Subject: [LAU] [OT] Free Course online about the Higgs Boson Message-ID: Hi dear all. Just wanted to let you know about this amazing course that the University of Edimburg has online during 7 weeks, and there's still days to sign in: https://www.futurelearn.com/courses/higgs As maybe you've notice, I'm quite eclectic and curious about anything, and Science (Physics in this case) is one of my favourite matters, so I'm in the course. It's now in the thick of the 2nd week and I tell you that it is amazing the quality and also the simplicity they've achieved given the complexity of the subjects, with a lot of instructional videos and some articles and texts, and even Mr. Higgs is there himself. Kindest Regards. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From lievenmoors at gmail.com Fri Feb 21 20:27:02 2014 From: lievenmoors at gmail.com (Lieven Moors) Date: Fri, 21 Feb 2014 21:27:02 +0100 Subject: [LAU] [LAD] JACK latency API clarifications In-Reply-To: <5307AA38.3080603@stackingdwarves.net> References: <20140221185221.GA566@satellite> <5307AA38.3080603@stackingdwarves.net> Message-ID: <20140221202701.GB566@satellite> On Fri, Feb 21, 2014 at 08:34:16PM +0100, J?rn Nettingsmeier wrote: > On 02/21/2014 07:52 PM, Lieven Moors wrote: > >>it was part of the API very early on, then we decided we didn't want to > >>impose the possibility of change on clients. as time goes on, it becomes > >>clear (to me at least) that we should have implemented it. > > > >What would be use cases for changing the sample rate dynamically? > > > having wired up a complex signal graph, which for the most part depends on > the studio, not on the project at hand, and then having to deal with > different projects in different sample rates. > > say your studio involves three monitoring setups, one main stereo, one > nearfield, and one surround, you are using jack to do EQ on those things, in > my case there's an ambisonic decoder in the loop as well. that means the > jack graph is already quite elaborated. in that case, it would be nice to > leave it running while switching from, say, a cd project at 44k1 to a tv > thing at 48k. > > as it is now, i have decided to do _everything_ at 48k (i have no second > thoughts about a final resampling step), but if a client brings material at, > say, 96k, i have to downsample first. sometimes i wish for an easy way to > reclock a graph. obviously, nobody expects this to be gapless. fading > everthing down and then taking a few seconds to reclock everything would be > fine. > > but then, many pieces of software in my chain would need changes. for > instance, an important piece of dsp for me is jconvolver, as it sits in > front of all my speakers. > of course, the impulse responses i use for EQ and room correction only make > sense for a given sample rate - it would have to be changed to swap one set > of IRs for another during a reclocking call, and of course that needs to be > configured and the user actually needs to provide those different IRs. > Yes, I see... I got into the habbit of using the same sample rate for all my projects as well. And I can remember a few times I wished to change the sample rate on the fly. Now I wonder if this would be difficult to implement. Do many clients expect the sample rate to remain stable? Aren't most clients checking for the sample rate in the process callback anyway? Of course, clients depending on samples or IR's would have to play back at the wrong rate... lieven From paul at linuxaudiosystems.com Fri Feb 21 20:29:40 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 21 Feb 2014 15:29:40 -0500 Subject: [LAU] [LAD] JACK latency API clarifications In-Reply-To: <20140221202701.GB566@satellite> References: <20140221185221.GA566@satellite> <5307AA38.3080603@stackingdwarves.net> <20140221202701.GB566@satellite> Message-ID: On Fri, Feb 21, 2014 at 3:27 PM, Lieven Moors wrote: > Aren't most clients checking for > the sample rate in the process callback anyway? if they are, then they are doing it wrong. -------------- next part -------------- An HTML attachment was scrubbed... URL: From lievenmoors at gmail.com Fri Feb 21 20:31:39 2014 From: lievenmoors at gmail.com (Lieven Moors) Date: Fri, 21 Feb 2014 21:31:39 +0100 Subject: [LAU] [LAD] JACK latency API clarifications In-Reply-To: <20140221202701.GB566@satellite> References: <20140221185221.GA566@satellite> <5307AA38.3080603@stackingdwarves.net> <20140221202701.GB566@satellite> Message-ID: <20140221203139.GD566@satellite> > > Yes, I see... > I got into the habbit of using the same sample rate for all my projects > as well. And I can remember a few times I wished to change the sample rate > on the fly. > > Now I wonder if this would be difficult to implement. Do many clients > expect the sample rate to remain stable? Aren't most clients checking for > the sample rate in the process callback anyway? Of course, clients depending > on samples or IR's would have to play back at the wrong rate... > > lieven Sorry, sent this to the wrong list... From jeremy at autostatic.com Fri Feb 21 20:54:09 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 21 Feb 2014 21:54:09 +0100 Subject: [LAU] Low latency audio interface for the Beagle Bone Black? In-Reply-To: References: Message-ID: <5307BCF1.9070402@autostatic.com> On 02/20/2014 09:34 PM, Rafael Vega wrote: > Also wondering how hard it would be to directly connect some ADCs and DACs. > Has anyone tried? Hi Rafael, There are so-called capes available for BeagleBoard products like the BBB. Audio capes are available too: http://elinux.org/CircuitCo:Audio_Cape_RevB Best, 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 angelv at iac.es Fri Feb 21 21:30:45 2014 From: angelv at iac.es (Angel de Vicente) Date: Fri, 21 Feb 2014 21:30:45 +0000 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: <87y514smoh.fsf@bek.no> (anders vinjar's message of "Fri, 21 Feb 2014 12:57:34 +0100") References: <87y514smoh.fsf@bek.no> Message-ID: Hi, anders.vinjar at bek.no writes: > A> Before I go to the effort of starting to learn it, I wanted to > A> know if it is possible to somehow import audio files with it... > > Hi Angel. This should be rather straightforward w. CM3/Grace. > > When starting Grace (CM3's front-end), you get access to piles of > examples for working with soundfiles, processing or synthesis. Try > looking in the menu "Audio"->"Instrument Browser" (or hit Ctrl-i), and > load something which looks close. I've almost given up (for the moment) on CM, as SuperCollider was so easy for this. In any case, I would like to try CM later on, but what I gather from your mail is that I cannot easily import my own audio clips (and instead you are suggesting to look for stuff that looks similar to what I intended to import)? Am I correct? Then, I guess CM is still not suited to what I wanted to do, becuase the audio clips that I want to import are parts of songs, voices that I have recorded, etc. I guess that there has to be some way of impoting new audio clips, but for the time being I think I will try with SC. Cheers, -- ??ngel de Vicente http://www.iac.es/galeria/angelv/ --------------------------------------------------------------------------------------------- ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci?n de Datos, acceda a http://www.iac.es/disclaimer.php WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en From grekimj at acousticrefuge.com Fri Feb 21 21:50:17 2014 From: grekimj at acousticrefuge.com (Grekim Jennings) Date: Fri, 21 Feb 2014 16:50:17 -0500 Subject: [LAU] [OT] Free Course online about the Higgs Boson Message-ID: <5307CA19.9070709@acousticrefuge.com> Hi dear all. Just wanted to let you know about this amazing course that the University of Edimburg has online during 7 weeks, and there's still days to sign in: https://www.futurelearn.com/courses/higgs As maybe you've notice, I'm quite eclectic and curious about anything, and Science (Physics in this case) is one of my favourite matters, so I'm in the course. It's now in the thick of the 2nd week and I tell you that it is amazing the quality and also the simplicity they've achieved given the complexity of the subjects, with a lot of instructional videos and some articles and texts, and even Mr. Higgs is there himself. Kindest Regards. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es Thank you! This looks fascinating! Grekim From lists at parisson.com Fri Feb 21 22:20:23 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Fri, 21 Feb 2014 23:20:23 +0100 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <20140130141433.GB4676@ordinator> References: <20140129024408.GA3961@ordinator> <52EA15A5.8070107@autostatic.com> <20140130141433.GB4676@ordinator> Message-ID: <5307D127.2080206@parisson.com> On 30/01/2014 15:14, Lewis Pike wrote: > On Thu, Jan 30, 2014 at 10:04:37AM +0100, Jeremy Jongepier wrote: >> Maybe resetting the JACK buffersize helps, you can do this with >> jack_bufsize on the command line. Without an option it will prompt >> the current buffer size, if you then run jack_bufsize with that >> value the situation might improve. Or you could first use a higher >> value and then switch back to the former lower value. > > Hi Jeremy, > > Thanks for your help here; it's much appreciated. I actually don't > have JACK installed. I've really only been using a single application > at a time so my needs at this stage are rather modest. I figured the > added layer of abstraction offered by JACK would only complicate my > troubleshooting. > > I'm really only speculating here, but it seems like the ALSA drivers > are sometimes improperly initializing the Fast Track Pro. Why this > happens only some of the time is strange indeed. > Trying with JACK could help us a lot about this bug. Thanks G From lists at parisson.com Fri Feb 21 22:20:55 2014 From: lists at parisson.com (Guillaume Pellerin) Date: Fri, 21 Feb 2014 23:20:55 +0100 Subject: [LAU] M-Audio Fast Track Pro: unreliable, distorted recording In-Reply-To: <20140130141433.GB4676@ordinator> References: <20140129024408.GA3961@ordinator> <52EA15A5.8070107@autostatic.com> <20140130141433.GB4676@ordinator> Message-ID: <5307D147.6030509@parisson.com> 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 From nettings at stackingdwarves.net Sat Feb 22 01:32:03 2014 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sat, 22 Feb 2014 02:32:03 +0100 Subject: [LAU] [OT] Free Course online about the Higgs Boson In-Reply-To: <5307CA19.9070709@acousticrefuge.com> References: <5307CA19.9070709@acousticrefuge.com> Message-ID: <5307FE13.6040206@stackingdwarves.net> On 02/21/2014 10:50 PM, Grekim Jennings wrote: > Hi dear all. > > Just wanted to let you know about this amazing course that the > University of Edimburg has online during 7 weeks, and there's still > days to sign in: > https://www.futurelearn.com/courses/higgs > > As maybe you've notice, I'm quite eclectic and curious about anything, > and Science (Physics in this case) is one of my favourite matters, so > I'm in the course. It's now in the thick of the 2nd week and I tell > you that it is amazing the quality and also the simplicity they've > achieved given the complexity of the subjects, with a lot of > instructional videos and some articles and texts, and even Mr. Higgs > is there himself. thanks for posting this, however OT :) i hopped on right away, and it's a very welcome refresher so far. there are even exercises, which can then be discussed with other students. nice! i hope i can find the time to keep following it in the next few weeks. catching up with the first one and a half weeks is not a big deal if you have some physics knowledge left over from school, should be doable in just 2 or 3 hours. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From anders.vinjar at bek.no Sat Feb 22 11:05:41 2014 From: anders.vinjar at bek.no (anders.vinjar at bek.no) Date: Sat, 22 Feb 2014 12:05:41 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: (Angel de Vicente's message of "Fri, 21 Feb 2014 21:30:45 +0000") References: <87y514smoh.fsf@bek.no> Message-ID: <87ios7quey.fsf@bek.no> >>>>> "A" == Angel de Vicente writes: A> I've almost given up (for the moment) on CM, as SuperCollider was A> so easy for this. In any case, I would like to try CM later on, A> but what I gather from your mail is that I cannot easily import A> my own audio clips (and instead you are suggesting to look for A> stuff that looks similar to what I intended to import)? Am I A> correct? Not so. There are examples distributed with CM/Grace doing what you seem to be asking for. Cheers. -anders From rm at mh-freiburg.de Sat Feb 22 15:00:14 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Sat, 22 Feb 2014 16:00:14 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? In-Reply-To: <87ios7quey.fsf@bek.no> References: <87y514smoh.fsf@bek.no> <87ios7quey.fsf@bek.no> Message-ID: <20140222145320.M5187@mh-freiburg.de> On Sat, 22 Feb 2014 12:05:41 +0100, anders.vinjar wrote > >>>>> "A" == Angel de Vicente writes: > A> I've almost given up (for the moment) on CM, as SuperCollider > was A> so easy for this. In any case, I would like to try CM > later on, A> but what I gather from your mail is that I cannot > easily import A> my own audio clips (and instead you are > suggesting to look for A> stuff that looks similar to what I > intended to import)? Am I A> correct? > > Not so. There are examples distributed with CM/Grace doing what you > seem to be asking for. Playing audio loaded from files (kind of a file-streaming sampler)? What example are you thinking of? I couldn't find any example for playing samples from files. The only code I could find was in vkey.scm (wich is _not_ in the example section, you need to open the instrument browser, then select 'vkey' and then click 'edit instrument'). But that doesn't come with any example and seems to be broken - et requires expandn.scm, but that doesn't work (at least the examples don't work). Cheers, RalfD From anders.vinjar at bek.no Sat Feb 22 17:31:59 2014 From: anders.vinjar at bek.no (anders.vinjar at bek.no) Date: Sat, 22 Feb 2014 18:31:59 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? References: <87y514smoh.fsf@bek.no> <87ios7quey.fsf@bek.no> <20140222145320.M5187@mh-freiburg.de> Message-ID: <877g8nqcj4.fsf@bek.no> >>>>> "R" == R Mattes writes: R> Playing audio loaded from files (kind of a file-streaming R> sampler)? What example are you thinking of? try fullmix.scm From atte at youmail.dk Sat Feb 22 17:44:01 2014 From: atte at youmail.dk (Atte) Date: Sat, 22 Feb 2014 18:44:01 +0100 Subject: [LAU] ps3 controller with bloetooth Message-ID: <5308E1E1.8040206@youmail.dk> Hi I have "some success" with getting input from a ps3 controller over usb: This shows up in lsusb: atte at skagen:~$ lsusb | grep Sony Bus 003 Device 013: ID 054c:0268 Sony Corp. Batoh Device / PlayStation 3 Controller "cat /dev/input/js0" (as regular user) show the expected garbage when moving the controller around or touching buttons. I also get input in chuck opened with Hid.openJoystick(0), so I'm pretty sure the controller is recognized and sending stuff into the system. Now, how do I get it running over bluetooth (wireless). Google led me to lot's of outdated info, so I'm hoping someone here with hands-on experience on a recent system could provide a few starting points, hints, links or clues. Thanks in advance! -- Atte http://atte.dk http://modlys.dk From anders.vinjar at bek.no Sat Feb 22 18:27:19 2014 From: anders.vinjar at bek.no (anders.vinjar at bek.no) Date: Sat, 22 Feb 2014 19:27:19 +0100 Subject: [LAU] Common Music - Importing audio files, and pointers to learning material? References: <87y514smoh.fsf@bek.no> <87ios7quey.fsf@bek.no> <20140222145320.M5187@mh-freiburg.de> Message-ID: <8738jbq9yw.fsf@bek.no> >>>>> "R" == R Mattes writes: R> (...) requires expandn.scm, but that doesn't work (at least the R> examples don't work). Seems wave.ins uses the symbol 'env as a local variable for an envelope. Try replacing with aenv or some other name not used already. -anders From lmemsm at gmail.com Sat Feb 22 19:41:20 2014 From: lmemsm at gmail.com (LM) Date: Sat, 22 Feb 2014 14:41:20 -0500 Subject: [LAU] sound sample resources Message-ID: I posted some links related to sound samples to the FreePats mailing list and Carlos suggested I share them here as well. Was recently researching a multimedia topic and happened to run across some sound sample related web sites. Thought I'd share what I found in case others might be interested. I found a cross-platform soundfont editor for the sf2 format. Looks like it's built using Qt. http://sourceforge.net/projects/polyphone/ Thousands of free sound samples specially recorded by Philharmonia Orchestra players licensed under a Creative Commons Attribution-ShareAlike 3.0 Unported License: http://www.philharmonia.co.uk/explore/make_music/ The LinuxSampler project was founded in 2002 with the goal of producing a free, streaming capable open source pure software audio sampler with high stability: http://www.linuxsampler.org/about.html This project aims to create a full orchestral sample library in the .gig format for use with the linuxsampler: http://sourceforge.net/projects/openorchestra/ List of sample libraries: http://bb.linuxsampler.org/viewtopic.php?f=8&t=11 musescore soundfonts http://musescore.org/en/handbook/soundfont http://www.linuxmusicians.com/ linuxmusician samples: http://www.linuxmusicians.com/viewforum.php?f=50&sid=bd65ce094deb26d5c09255dbaafeea3e Recommended Sound Libraries and Where to Get Them... http://www.remastersys.com/forums/index.php?topic=1740.0 http://www.bandshed.net/sounds/sfz/ Web site to share Free music: http://freemusicpush.blogspot.com/ Best wishes. Laura http://www.distasis.com/recipes/music.htm From ralf.mardorf at rocketmail.com Sat Feb 22 20:30:57 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 22 Feb 2014 21:30:57 +0100 Subject: [LAU] sound sample resources In-Reply-To: References: Message-ID: <1393101057.1133.13.camel@archlinux> On Sat, 2014-02-22 at 14:41 -0500, LM wrote: > Carlos suggested I share them here as well. Thank you Laura and Carlos, it never harms to share information like this :). Regards, Ralf From rob at rektau.ukfsn.org Sat Feb 22 20:59:35 2014 From: rob at rektau.ukfsn.org (Rob) Date: Sat, 22 Feb 2014 20:59:35 +0000 Subject: [LAU] sound sample resources In-Reply-To: References: Message-ID: <94b48834-7041-46f5-9679-0ed4dd7c83bb@email.android.com> Thanks Rob -- Sent from my Android device with K-9 Mail. Please excuse my brevity. From temps.jo at gmail.com Sun Feb 23 07:28:32 2014 From: temps.jo at gmail.com (temps) Date: Sat, 22 Feb 2014 23:28:32 -0800 (PST) Subject: [LAU] lm3jo en deb version 1.1.1-4 Message-ID: <1393140512381-89556.post@n7.nabble.com> announcement I posted this weekend, lm3jo.deb . For download here: http://www.letime.net/vocale/paquet_deb/lm3jo.deb reminder lm3jo : Project synthesizer sounds lm3jo -- View this message in context: http://linux-audio.4202.n7.nabble.com/lm3jo-en-deb-version-1-1-1-4-tp89556.html Sent from the linux-audio-user mailing list archive at Nabble.com. From hamish.low.net at gmx.com Sun Feb 23 16:52:50 2014 From: hamish.low.net at gmx.com (sub_acoustic) Date: Sun, 23 Feb 2014 08:52:50 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <1386262286208-88128.post@n7.nabble.com> References: <1384701581408-87925.post@n7.nabble.com> <5288EE64.1010004@youmail.dk> <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <1386015225436-88069.post@n7.nabble.com> <529CF0B8.6050304@youmail.dk> <1386262286208-88128.post@n7.nabble.com> Message-ID: <1393174369381-89557.post@n7.nabble.com> Hi, Excuse my ignorance, but how to I recompile a kernel with the quirk? Or if you could direct me to the information I can't find any straightforward instructions anywhere... Just bought an R16, using Ubuntu Studio 64-bit - 12.04 but will upgrade shortly (and hopefully have this quirk in the kernel... cheers -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89557.html Sent from the linux-audio-user mailing list archive at Nabble.com. From espiritocz at gmail.com Sun Feb 23 18:09:06 2014 From: espiritocz at gmail.com (Milan Lazecky) Date: Sun, 23 Feb 2014 19:09:06 +0100 Subject: [LAU] raspi as midi synth Message-ID: hi folks, i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb cable to plug into raspberry pi which would use some fluidsynth soundfonts to synthesize music realtime. do you know about such project existing, or should i work on it from beginning? (i.e. find some rt kernel, install sw, rearrange system for no gui etc...) do you think raspberry can handle it well? thank you. milan -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Sun Feb 23 18:18:27 2014 From: atte at youmail.dk (Atte) Date: Sun, 23 Feb 2014 19:18:27 +0100 Subject: [LAU] re Zoom R16 In-Reply-To: <1393174369381-89557.post@n7.nabble.com> References: <1384701581408-87925.post@n7.nabble.com> <5288EE64.1010004@youmail.dk> <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <1386015225436-88069.post@n7.nabble.com> <529CF0B8.6050304@youmail.dk> <1386262286208-88128.post@n7.nabble.com> <1393174369381-89557.post@n7.nabble.com> Message-ID: <530A3B73.6010406@youmail.dk> On 02/23/2014 05:52 PM, sub_acoustic wrote: > Hi, > > Excuse my ignorance, but how to I recompile a kernel with the quirk? Or if > you could direct me to the information > I can't find any straightforward instructions anywhere... Here's what I did: 1) placed the quirks in a file (/usr/src/zoom_quirks.h) 2) included it in the build by adding a line to /usr/src/linux/sound/usb/quirks_table.h: #include "../../../zoom_quirks.h" It might be more direct to add the the quirk intry directly in /usr/src/linux/sound/usb/quirks_table.h, but this way it survives across different kernel builds and is easier to fiddle with. Here's the content of zoom_quirks.h (between ---begin--- and ---end---, remember the first character of the file is "{" the last is ",": ---begin--- { /* 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_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_MIDI_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .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 }, } } }, ---end--- Hope that helps! -- Atte http://atte.dk http://modlys.dk From fero.kiraly at gmail.com Sun Feb 23 18:40:46 2014 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Sun, 23 Feb 2014 19:40:46 +0100 Subject: [LAU] sound sample resources Message-ID: really great sources. thank you. -- Fero Kiraly www.ferokiraly.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From hamish.low.net at gmx.com Sun Feb 23 19:48:55 2014 From: hamish.low.net at gmx.com (sub_acoustic) Date: Sun, 23 Feb 2014 11:48:55 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <530A3B73.6010406@youmail.dk> References: <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <1393184935653-89561.post@n7.nabble.com> Thanks Atte, That's useful, but I don't actually know how to recompile the kernel...does this mean that the quirk will be picked up when I start up the computer, or will I need to remake the kernel in order to add it... https://help.ubuntu.com/community/Kernel/Compile tells me that if I don't know what I'm doing I might wreck my computer... */Reasons for NOT compiling a custom kernel > You merely need to compile a special driver. For this, you only need to > install the linux-headers packages/* Also another quick question...I understand that once I can capture via the R16, I'll need another device for playback from the laptop/ardour/jack etc. Would the laptop soundcard be sufficient for monitoring?, or should I get another device...? Would this do...http://www.behringer.com/EN/Products/UCA202.aspx I've been using/trying to use a Presonus Firebox for a couple of years, intermittently it worked well, but overall Firewire and FFADO and xruns have been a major major major headache and waste of time so I've given it away...this is also the reason while I'll do the bulk of my recording directly in the Zoom R16 and then use my laptop primarily for mixing and mastering... ...I've learnt to be wary of an enthusiastic developers who know way more than I do, and so I can never hope to emulate the performance they claim they have...with the R16 I can't go wrong as I'll have recorded something regardless... Cheers Hamish -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89561.html Sent from the linux-audio-user mailing list archive at Nabble.com. From rob at rektau.ukfsn.org Sun Feb 23 22:20:53 2014 From: rob at rektau.ukfsn.org (Rob) Date: Sun, 23 Feb 2014 22:20:53 +0000 Subject: [LAU] re Zoom R16 In-Reply-To: <1393184935653-89561.post@n7.nabble.com> References: <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: On 23 February 2014 19:48:55 GMT+00:00, sub_acoustic wrote: >Thanks Atte, > >That's useful, but I don't actually know how to recompile the >kernel...does >this mean that the quirk will be picked up when I start up the >computer, or >will I need to remake the kernel in order to add it... >https://help.ubuntu.com/community/Kernel/Compile > tells me that if I >don't know what I'm doing I might wreck my computer... > >*/Reasons for NOT compiling a custom kernel >> You merely need to compile a special driver. For this, you only need >to >> install the linux-headers packages/* > >Also another quick question...I understand that once I can capture via >the >R16, I'll need another device for playback from the laptop/ardour/jack >etc. >Would the laptop soundcard be sufficient for monitoring?, or should I >get >another device...? Would this >do...http://www.behringer.com/EN/Products/UCA202.aspx > >I've been using/trying to use a Presonus Firebox for a couple of years, >intermittently it worked well, but overall Firewire and FFADO and xruns >have >been a major major major headache and waste of time so I've given it >away...this is also the reason while I'll do the bulk of my recording >directly in the Zoom R16 and then use my laptop primarily for mixing >and >mastering... >...I've learnt to be wary of an enthusiastic developers who know way >more >than I do, and so I can never hope to emulate the performance they >claim >they have...with the R16 I can't go wrong as I'll have recorded >something >regardless... > >Cheers > >Hamish > > > > >-- >View this message in context: >http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89561.html >Sent from the linux-audio-user mailing list archive at Nabble.com. >_______________________________________________ >Linux-audio-user mailing list >Linux-audio-user at lists.linuxaudio.org >http://lists.linuxaudio.org/listinfo/linux-audio-user Check out: http://wiki.linuxaudio.org/wiki/start -- Sent from my Android device with K-9 Mail. Please excuse my brevity. From simonzwise at gmail.com Sun Feb 23 23:03:35 2014 From: simonzwise at gmail.com (Simon Wise) Date: Mon, 24 Feb 2014 10:03:35 +1100 Subject: [LAU] raspi as midi synth In-Reply-To: References: Message-ID: <530A7E47.5080401@gmail.com> On 24/02/14 05:09, Milan Lazecky wrote: > hi folks, > i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb > cable to plug into raspberry pi which would use some fluidsynth soundfonts > to synthesize music realtime. > do you know about such project existing, or should i work on it from > beginning? (i.e. find some rt kernel, install sw, rearrange system for no > gui etc...) > do you think raspberry can handle it well? USB is probably your biggest issue here, provided the raspberry can handle the synthesis you want to do, you can help by assigning only a minimum of memory to the GPU. No need to do any other rearranging of the system, with raspbian it starts out headless (with X installed if you want to launch it) and capable of quite low latencies. It has a basic minimum of processes running and ready to add what you want from the extensive repository with apt-get. The settings to work headless are mostly the default. These are great advantages in terms of time required to get it running. USB implementation is poor and only some audio cards work, plus something else on USB like a midi input or networking will make it trickier. In your case, needing only stereo out, it is certainly easier. Seriously consider an audio solution that can use HDMI audio out. Simon From atte at youmail.dk Mon Feb 24 06:44:17 2014 From: atte at youmail.dk (Atte) Date: Mon, 24 Feb 2014 07:44:17 +0100 Subject: [LAU] re Zoom R16 In-Reply-To: <1393184935653-89561.post@n7.nabble.com> References: <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <530AEA41.1090909@youmail.dk> On 02/23/2014 08:48 PM, sub_acoustic wrote: > Thanks Atte, > > That's useful, but I don't actually know how to recompile the kernel...does > this mean that the quirk will be picked up when I start up the computer, or > will I need to remake the kernel in order to add it... What I do is: download a kernel from kernel.org, apply realtime patch, apply the zoom_quirks.txt, compile the kernel, install it and reboot into it... It's not that easy, although there are lots of good info outthere... -- Atte http://atte.dk http://modlys.dk From espiritocz at gmail.com Mon Feb 24 08:19:18 2014 From: espiritocz at gmail.com (Milan Lazecky) Date: Mon, 24 Feb 2014 09:19:18 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <530A7E47.5080401@gmail.com> References: <530A7E47.5080401@gmail.com> Message-ID: Thank you Simon for positive reaction, I start to look forward to prepare such synthesizer, will think about HDMI audio, but at this moment I will be just happy with RPi stereo line-out. Kind regards Milan 2014-02-24 0:03 GMT+01:00 Simon Wise : > On 24/02/14 05:09, Milan Lazecky wrote: > >> hi folks, >> i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb >> cable to plug into raspberry pi which would use some fluidsynth soundfonts >> to synthesize music realtime. >> do you know about such project existing, or should i work on it from >> beginning? (i.e. find some rt kernel, install sw, rearrange system for no >> gui etc...) >> do you think raspberry can handle it well? >> > > USB is probably your biggest issue here, provided the raspberry can handle > the synthesis you want to do, you can help by assigning only a minimum of > memory to the GPU. No need to do any other rearranging of the system, with > raspbian it starts out headless (with X installed if you want to launch it) > and capable of quite low latencies. It has a basic minimum of processes > running and ready to add what you want from the extensive repository with > apt-get. The settings to work headless are mostly the default. These are > great advantages in terms of time required to get it running. > > USB implementation is poor and only some audio cards work, plus something > else on USB like a midi input or networking will make it trickier. In your > case, needing only stereo out, it is certainly easier. > > Seriously consider an audio solution that can use HDMI audio out. > > > Simon > _______________________________________________ > 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 Feb 24 08:52:57 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 24 Feb 2014 09:52:57 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: References: Message-ID: <530B0869.60604@autostatic.com> On 02/23/2014 07:09 PM, Milan Lazecky wrote: > hi folks, > i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb > cable to plug into raspberry pi which would use some fluidsynth soundfonts > to synthesize music realtime. > do you know about such project existing, or should i work on it from > beginning? (i.e. find some rt kernel, install sw, rearrange system for no > gui etc...) > do you think raspberry can handle it well? > > thank you. > > milan Hi Milan, http://wiki.linuxaudio.org/wiki/raspberrypi Also check my Youtube channel for some examples: http://www.youtube.com/autostatic3000/raspberrypi I've had a brief occasion where I could hook up an Akai EWI to my RPI, it worked but I couldn't test it extensively. Best, 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 gerhard.zintel at web.de Mon Feb 24 11:01:32 2014 From: gerhard.zintel at web.de (Gerhard Zintel) Date: Mon, 24 Feb 2014 12:01:32 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <530A7E47.5080401@gmail.com> References: <530A7E47.5080401@gmail.com> Message-ID: <201402241201.32230.gerhard.zintel@web.de> On Monday 24 February 2014, Simon Wise wrote: > On 24/02/14 05:09, Milan Lazecky wrote: > > i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb > > cable to plug into raspberry pi which would use some fluidsynth soundfonts > > to synthesize music realtime. > > do you know about such project existing, or should i work on it from > > beginning? (i.e. find some rt kernel, install sw, rearrange system for no > > gui etc...) > > do you think raspberry can handle it well? > > USB implementation is poor and only some audio cards work, plus something else > on USB like a midi input or networking will make it trickier. In your case, > needing only stereo out, it is certainly easier. > I have very good results on a Raspi with an USB Audio DAC from here: http://hifimediy.com/index.php?route=product/product&product_id=83 Description from the site: "HiFimeDIY Sabre USB DAC. Digital to Analog Converter 96khz/24bit (incl USB to optical converter feature)" Gerhard From simonzwise at gmail.com Mon Feb 24 11:06:27 2014 From: simonzwise at gmail.com (Simon Wise) Date: Mon, 24 Feb 2014 22:06:27 +1100 Subject: [LAU] raspi as midi synth In-Reply-To: References: <530A7E47.5080401@gmail.com> Message-ID: <530B27B3.2030103@gmail.com> On 24/02/14 19:19, Milan Lazecky wrote: > Thank you Simon for positive reaction, > I start to look forward to prepare such synthesizer, will think about HDMI > audio, but at this moment I will be just happy with RPi stereo line-out. > Kind regards > You might not be so happy with the built in line out, it is a very basic output with very low quality audio. The hardware design is quite heavily focussed on playing media (preferably directly, on an otherwise headless system) through the HDMI output, and I've found the supplied libraries and code for doing this quite workable. The xbmc lot did a fair bit of work on this, and as a small device to attach to a projector they are great. I've also used some small usb audio cards with good results, but did not need low latency or other usb stuff so I can't say how they'd go with your project. Here is a link to one we used ... http://www.ebay.co.uk/itm/UK-PCM2704-USB-to-S-PDIF-Sound-Card-DAC-3-5mm-analog-digital-output-volume-funct-/120940457164 but I think we got them a bit cheaper than that, and used a little amp about that size also ... for a bunch of sound sources. There are lots of little cards based on that chip, they probably all work well. Simon From espiritocz at gmail.com Mon Feb 24 12:13:15 2014 From: espiritocz at gmail.com (Milan Lazecky) Date: Mon, 24 Feb 2014 13:13:15 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <530B27B3.2030103@gmail.com> References: <530A7E47.5080401@gmail.com> <530B27B3.2030103@gmail.com> Message-ID: Thank you all, now I get enough information - I will order some simple usb soundcard/DAC and will approach to tune up the system using LAU wiki, thanks for the links and ideas! Will inform about some progress once I will get the card etc. Thank you! Milan 2014-02-24 12:06 GMT+01:00 Simon Wise : > On 24/02/14 19:19, Milan Lazecky wrote: > >> Thank you Simon for positive reaction, >> I start to look forward to prepare such synthesizer, will think about HDMI >> audio, but at this moment I will be just happy with RPi stereo line-out. >> Kind regards >> >> > You might not be so happy with the built in line out, it is a very basic > output with very low quality audio. The hardware design is quite heavily > focussed on playing media (preferably directly, on an otherwise headless > system) through the HDMI output, and I've found the supplied libraries and > code for doing this quite workable. The xbmc lot did a fair bit of work on > this, and as a small device to attach to a projector they are great. > > I've also used some small usb audio cards with good results, but did not > need low latency or other usb stuff so I can't say how they'd go with your > project. > > Here is a link to one we used ... > > http://www.ebay.co.uk/itm/UK-PCM2704-USB-to-S-PDIF-Sound- > Card-DAC-3-5mm-analog-digital-output-volume-funct-/120940457164 > > but I think we got them a bit cheaper than that, and used a little amp > about that size also ... for a bunch of sound sources. There are lots of > little cards based on that chip, they probably all work well. > > > Simon > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From itarozzi at gmail.com Mon Feb 24 15:58:36 2014 From: itarozzi at gmail.com (Ivan Tarozzi) Date: Mon, 24 Feb 2014 16:58:36 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: References: Message-ID: <530B6C2C.7080305@gmail.com> Hi Carlos, I'm developing an industrial application for a client using APC Rock. I'm using Android, so I can confirm that linux runs on it and you can find kernel and uboot here: https://github.com/apc-io/apc-rock I never tried a "standard" linux ditributions, so I can't tell about it, but I fear the VIA support against linux is absent. I suggest you look at some better supported board. just my2c Ivan Il 21/02/2014 09:35, Carlos sanchiavedraz ha scritto: > Hello dear all. > > Lately I've found this devices by APC that seem quite interesting to > me given my interest and projects around Raspberry PI and Linux Audio: > > http://apc.io/products/rock/ > http://apc.io/products/8750a/ > > I love this presentation as a book,, > http://apc.io/products/paper/ > > ... and that, presenting a device in a nice and not PC-like way, is > something I've been thinkinf of for a long time, but I couldn't do it > myself without a lot of building and DIY. > > Does anybody have or tried one as it is? And with Linux? > From jason at mancine.net Mon Feb 24 17:29:44 2014 From: jason at mancine.net (jmancine) Date: Mon, 24 Feb 2014 09:29:44 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <530A3B73.6010406@youmail.dk> References: <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <1393262984491-89575.post@n7.nabble.com> Atte Andr? Jensen wrote > On 02/23/2014 05:52 PM, sub_acoustic wrote: >> Hi, >> >> Excuse my ignorance, but how to I recompile a kernel with the quirk? Or >> if >> you could direct me to the information >> I can't find any straightforward instructions anywhere... > > Here's what I did: > > 1) placed the quirks in a file (/usr/src/zoom_quirks.h) > > 2) included it in the build by adding a line to > /usr/src/linux/sound/usb/quirks_table.h: > #include "../../../zoom_quirks.h" > > It might be more direct to add the the quirk intry directly in > /usr/src/linux/sound/usb/quirks_table.h, but this way it survives across > different kernel builds and is easier to fiddle with. Hi Atte, Am I right that you can make a change to your zoom_quirks.h file and it will be included when you reboot (without rebuilding the kernel)? I was looking for a way to do this... so simple. :) -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89575.html Sent from the linux-audio-user mailing list archive at Nabble.com. From rm at mh-freiburg.de Mon Feb 24 17:36:53 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Mon, 24 Feb 2014 18:36:53 +0100 Subject: [LAU] re Zoom R16 In-Reply-To: <1393262984491-89575.post@n7.nabble.com> References: <1384707481221-87927.post@n7.nabble.com> <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> <1393262984491-89575.post@n7.nabble.com> Message-ID: <20140224173514.M71899@mh-freiburg.de> On Mon, 24 Feb 2014 09:29:44 -0800 (PST), jmancine wrote > Hi Atte, Not being Atte, but I think I can answer this. > Am I right that you can make a change to your zoom_quirks.h file and > it will be included when you reboot (without rebuilding the kernel)? No way. The .h files are read by the C compiler (actually, the C preprocessor). Changes are only picked up during a recomile. HTH Ralf Mattes From jason at mancine.net Mon Feb 24 17:41:51 2014 From: jason at mancine.net (jmancine) Date: Mon, 24 Feb 2014 09:41:51 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <20140224173514.M71899@mh-freiburg.de> References: <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> <1393262984491-89575.post@n7.nabble.com> <20140224173514.M71899@mh-freiburg.de> Message-ID: <1393263711325-89577.post@n7.nabble.com> R. Mattes wrote > No way. The .h files are read by the C compiler (actually, the C > preprocessor). Changes are only picked up during a recomile. Makes sense...Thanks for the response. The constant recompiling is what has kept me from testing more solutions to playback on the R16...wish there was a way to set a dynamic path to the R16 quirk that was picked up on reboot, but it sounds as if it needs to be compiled each time. -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89577.html Sent from the linux-audio-user mailing list archive at Nabble.com. From gordonjcp at gjcp.net Mon Feb 24 17:46:21 2014 From: gordonjcp at gjcp.net (Gordon JC Pearce) Date: Mon, 24 Feb 2014 17:46:21 +0000 Subject: [LAU] re Zoom R16 In-Reply-To: <1393263711325-89577.post@n7.nabble.com> References: <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> <1393262984491-89575.post@n7.nabble.com> <20140224173514.M71899@mh-freiburg.de> <1393263711325-89577.post@n7.nabble.com> Message-ID: <20140224174621.GB28990@gjcp.net> On Mon, Feb 24, 2014 at 09:41:51AM -0800, jmancine wrote: > R. Mattes wrote > > No way. The .h files are read by the C compiler (actually, the C > > preprocessor). Changes are only picked up during a recomile. > > Makes sense...Thanks for the response. The constant recompiling is what has > kept me from testing more solutions to playback on the R16...wish there was > a way to set a dynamic path to the R16 quirk that was picked up on reboot, > but it sounds as if it needs to be compiled each time. Compile as a module, unload, compile, reload. -- Gordonjcp MM0YEQ From jason at mancine.net Mon Feb 24 18:07:33 2014 From: jason at mancine.net (jmancine) Date: Mon, 24 Feb 2014 10:07:33 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <20140224174621.GB28990@gjcp.net> References: <1384883176341-87971.post@n7.nabble.com> <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> <1393262984491-89575.post@n7.nabble.com> <20140224173514.M71899@mh-freiburg.de> <1393263711325-89577.post@n7.nabble.com> <20140224174621.GB28990@gjcp.net> Message-ID: <1393265253705-89581.post@n7.nabble.com> gordonjcp wrote > Compile as a module, unload, compile, reload. Not sure exactly what you mean...I would still have to recompile the whole kernel (in order to pass the changes to all the other modules like snd_usb_audio, soundcore, snd_usbmidi, etc, etc) right??? -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89581.html Sent from the linux-audio-user mailing list archive at Nabble.com. From espiritocz at gmail.com Mon Feb 24 19:57:23 2014 From: espiritocz at gmail.com (Milan Lazecky) Date: Mon, 24 Feb 2014 20:57:23 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <530ba115.a43ac20a.23c6.737dSMTPIN_ADDED_MISSING@mx.google.com> References: <530ba115.a43ac20a.23c6.737dSMTPIN_ADDED_MISSING@mx.google.com> Message-ID: actually, you are right Ben. I intended to prepare RPi to use some external battery to play "on street", or somewhere more mobile, instead of taking my precious VL70m synthesizer all the time. Thank you for reminder. You say "rapid stuff gets laggy", I will try it anyway, using minimalistic raspbian and some performance tuning, but thank you for your "kind warning", will keep it in mind. Will try to use some more simple soundfonts (but probably not any "cheap" 8-bit like synthesizers..) Regards Milan 2014-02-24 20:44 GMT+01:00 Ben Bell : > On Sun, Feb 23, 2014 at 07:09:06PM +0100, Milan Lazecky wrote: > > i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb > > cable to plug into raspberry pi which would use some fluidsynth > soundfonts > > to synthesize music realtime. > > There are some examples of people trying to do this and I think the general > impression is that the latency is touch and go depending on what you're > playing. I have one here running fluidsynth, alsa midi using an Evolution > USB keyboard, and a set of mellotron soundfonts. It's OK for chords and > single note runs, but if I try anything rapid, it feels laggy. Of course, > compared with a real mellotron that's not so bad, but playing a WX5 may be > sore. > > As others have pointed out the audio out isn't audiophile quality, but I'd > have thought if you were in a studio you'd use proper hardware and this > would be for live use? In which case, factory in an amp, an audience > talking > and so on, and I don't think it's as big an issue as people make out. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bjb-linux-audio-user at deus.net Mon Feb 24 19:44:20 2014 From: bjb-linux-audio-user at deus.net (Ben Bell) Date: Mon, 24 Feb 2014 19:44:20 +0000 Subject: [LAU] raspi as midi synth In-Reply-To: References: Message-ID: <20140224205256.2F8D162BFC@lists.linuxaudio.org> On Sun, Feb 23, 2014 at 07:09:06PM +0100, Milan Lazecky wrote: > i have a yamaha wx5 (midi saxophone). i was thinking to use my midi2usb > cable to plug into raspberry pi which would use some fluidsynth soundfonts > to synthesize music realtime. There are some examples of people trying to do this and I think the general impression is that the latency is touch and go depending on what you're playing. I have one here running fluidsynth, alsa midi using an Evolution USB keyboard, and a set of mellotron soundfonts. It's OK for chords and single note runs, but if I try anything rapid, it feels laggy. Of course, compared with a real mellotron that's not so bad, but playing a WX5 may be sore. As others have pointed out the audio out isn't audiophile quality, but I'd have thought if you were in a studio you'd use proper hardware and this would be for live use? In which case, factory in an amp, an audience talking and so on, and I don't think it's as big an issue as people make out. From hamish.low.net at gmx.com Mon Feb 24 21:50:43 2014 From: hamish.low.net at gmx.com (sub_acoustic) Date: Mon, 24 Feb 2014 13:50:43 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <530AEA41.1090909@youmail.dk> References: <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <1393278643930-89584.post@n7.nabble.com> Thanks Atte, Does that mean that I could download the latest UbuntuStudio distro, including kernel then apply the zoom_quirks.txt, compile the kernel, install it and reboot into it...? sounds tricky...perhaps the Ubuntu Studio developers would be so kind as to add the quirk to the next distro... why are manufacturers so hesitant to support linux?, or at least share information with Linux developers? they would sell so many more units... Just got my Zoom R16...time to start making some sound! -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89584.html Sent from the linux-audio-user mailing list archive at Nabble.com. From gianfranco at portalmod.com.br Mon Feb 24 21:56:07 2014 From: gianfranco at portalmod.com.br (Gianfranco Ceccolini) Date: Mon, 24 Feb 2014 18:56:07 -0300 Subject: [LAU] simple command line player for JACK Message-ID: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> Hi everybody I?ve just stumbled on a very simple issue that I can?t solve. Is there a simple command line player for JACK that plays .ogg files? My first option would be the mplayer but it has got loads of dependencies and , as I?m working on a embedded device, I?d like to keep things as lean as possible. Any help or suggestion is appreciated Kind regards Gianfranco The MOD Team From leoave at gmail.com Mon Feb 24 21:57:59 2014 From: leoave at gmail.com (Leonardo Palomares) Date: Mon, 24 Feb 2014 13:57:59 -0800 Subject: [LAU] simple command line player for JACK In-Reply-To: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> Message-ID: mocp (from moc packages) it runs on jack or alsa On Mon, Feb 24, 2014 at 1:56 PM, Gianfranco Ceccolini < gianfranco at portalmod.com.br> wrote: > Hi everybody > > I?ve just stumbled on a very simple issue that I can?t solve. > > Is there a simple command line player for JACK that plays .ogg files? > > My first option would be the mplayer but it has got loads of dependencies > and , as I?m working on a embedded device, I?d like to keep things as lean > as possible. > > Any help or suggestion is appreciated > > Kind regards > > Gianfranco > The MOD Team > _______________________________________________ > 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 Feb 24 23:23:02 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 24 Feb 2014 15:23:02 -0800 (PST) Subject: [LAU] simple command line player for JACK In-Reply-To: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> Message-ID: On Mon, 24 Feb 2014, Gianfranco Ceccolini wrote: > I?ve just stumbled on a very simple issue that I can?t solve. > Is there a simple command line player for JACK that plays .ogg files? > My first option would be the mplayer but it has got loads of dependencies and , as I?m working on a embedded device, I?d like to keep things as lean as possible. > Any help or suggestion is appreciated Have you tried jack.play from jack.tools? ogg123 will send it's output to stdout as a wav if there is a jack player that will accept that as input. -- Len Ovens www.ovenwerks.net From fons at linuxaudio.org Mon Feb 24 23:34:17 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 24 Feb 2014 23:34:17 +0000 Subject: [LAU] simple command line player for JACK In-Reply-To: References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> Message-ID: <20140224233417.GB2394@linuxaudio.org> On Mon, Feb 24, 2014 at 03:23:02PM -0800, Len Ovens wrote: > ogg123 will send it's output to stdout as a wav if there is a jack > player that will accept that as input. If find it incredible that after all those years programs like ogg123 still don't support Jack, while they will happily output to a number of archaic systems like esd, arts, irix,... 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 lau at kudla.org Mon Feb 24 23:54:10 2014 From: lau at kudla.org (Rob Kudla) Date: Mon, 24 Feb 2014 18:54:10 -0500 Subject: [LAU] simple command line player for JACK In-Reply-To: <20140224233417.GB2394@linuxaudio.org> References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> <20140224233417.GB2394@linuxaudio.org> Message-ID: <530BDBA2.3020504@kudla.org> On 02/24/14 18:34, Fons Adriaensen wrote: > If find it incredible that after all those years programs like ogg123 > still don't support Jack, while they will happily output to a number of > archaic systems like esd, arts, irix,... Well, it's a hell of a lot easier to merely refrain from deleting old code than it is to write new code. Just saying... Rob From joelz at pobox.com Tue Feb 25 01:56:59 2014 From: joelz at pobox.com (Joel Roth) Date: Mon, 24 Feb 2014 15:56:59 -1000 Subject: [LAU] simple command line player for JACK In-Reply-To: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> Message-ID: <20140225015659.GA24668@sprite> On Mon, Feb 24, 2014 at 06:56:07PM -0300, Gianfranco Ceccolini wrote: > Hi everybody > > I?ve just stumbled on a very simple issue that I can?t solve. > > Is there a simple command line player for JACK that plays .ogg files? This might work (not tested) ogg123 -d raw test.ogg | ecasound -i stdin -o jack,system > My first option would be the mplayer but it has got loads of dependencies and , as I?m working on a embedded device, I?d like to keep things as lean as possible. > > Any help or suggestion is appreciated > > Kind regards > > Gianfranco > The MOD Team > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Joel Roth From paul at linuxaudiosystems.com Tue Feb 25 02:21:20 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Mon, 24 Feb 2014 21:21:20 -0500 Subject: [LAU] simple command line player for JACK In-Reply-To: <20140225015659.GA24668@sprite> References: <052E0D94-E8CC-4698-8C84-8A1B24D92F25@portalmod.com.br> <20140225015659.GA24668@sprite> Message-ID: On Mon, Feb 24, 2014 at 8:56 PM, Joel Roth wrote: > On Mon, Feb 24, 2014 at 06:56:07PM -0300, Gianfranco Ceccolini wrote: > > Hi everybody > > > > I've just stumbled on a very simple issue that I can't solve. > > > > Is there a simple command line player for JACK that plays .ogg files? > > This might work (not tested) > > ogg123 -d raw test.ogg | ecasound -i stdin -o jack,system > > if your distribution was sanely built, then you just need sndfile-jackplayer which plays every format that libsndfile can handle (including ogg). gstreamer command line utilities can also be used. -------------- next part -------------- An HTML attachment was scrubbed... URL: From csanchezgs at gmail.com Tue Feb 25 08:10:59 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 25 Feb 2014 09:10:59 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: <530B6C2C.7080305@gmail.com> References: <530B6C2C.7080305@gmail.com> Message-ID: 2014-02-24 16:58 GMT+01:00 Ivan Tarozzi : > Hi Carlos, > I'm developing an industrial application for a client using APC Rock. > > I'm using Android, so I can confirm that linux runs on it and you can > find kernel and uboot here: > https://github.com/apc-io/apc-rock > > I never tried a "standard" linux ditributions, so I can't tell about it, > but I fear the VIA support against linux is absent. > > I suggest you look at some better supported board. > > just my2c > > Ivan > > Thanks so much, Ivan. It's really helpful to have such a warn before wanting to purchase one of those and make my life trickier, most of all when surely I would try to put Musix inside that would have some driver problems (it's 100% free/libre based on Debian). I liked the book-style one, but now maybe I'll go some other way. Regards. > > > Il 21/02/2014 09:35, Carlos sanchiavedraz ha scritto: >> Hello dear all. >> >> Lately I've found this devices by APC that seem quite interesting to >> me given my interest and projects around Raspberry PI and Linux Audio: >> >> http://apc.io/products/rock/ >> http://apc.io/products/8750a/ >> >> I love this presentation as a book,, >> http://apc.io/products/paper/ >> >> ... and that, presenting a device in a nice and not PC-like way, is >> something I've been thinkinf of for a long time, but I couldn't do it >> myself without a lot of building and DIY. >> >> Does anybody have or tried one as it is? And with Linux? >> > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From jmckernon at gmail.com Tue Feb 25 15:37:15 2014 From: jmckernon at gmail.com (James Mckernon) Date: Tue, 25 Feb 2014 15:37:15 +0000 Subject: [LAU] compiling jack-osc? Message-ID: I'm trying to compile (or otherwise obtain) the jackd utilities found at http://rd.slavepianos.org/sw/rju/ , but I don't seem to be able to. In particular, I'm interested in using the jack-osc tool to convert jack transport messages (from Ardour) to osc messages. (Incidentally, if anyone knows of another way of doing this, please let me know.) Although the sources are there, I can't figure out how to compile them - in fact I don't even know if they're in a compileable state. I tried running make and playing around with gcc, but they seem to depend on some files that aren't present there. I'm running Arch Linux, but can't find a package for them in the AUR (Arch user repository). Are these tools still current, or have they succumbed to bit rot? Does anyone know how I might be able to compile them? Thanks, J From paul at linuxaudiosystems.com Tue Feb 25 15:51:39 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 25 Feb 2014 10:51:39 -0500 Subject: [LAU] compiling jack-osc? In-Reply-To: References: Message-ID: On Tue, Feb 25, 2014 at 10:37 AM, James Mckernon wrote: > I'm trying to compile (or otherwise obtain) the jackd utilities found > at http://rd.slavepianos.org/sw/rju/ , but I don't seem to be able to. > In particular, I'm interested in using the jack-osc tool to convert > jack transport messages (from Ardour) to osc messages. (Incidentally, > if anyone knows of another way of doing this, please let me know.) > Although the sources are there, I can't figure out how to compile them > - in fact I don't even know if they're in a compileable state. I tried > running make and playing around with gcc, but they seem to depend on > some files that aren't present there. I'm running Arch Linux, but > can't find a package for them in the AUR (Arch user repository). > > Are these tools still current, or have they succumbed to bit rot? Does > anyone know how I might be able to compile them? > how about some actual compile-time errors? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmckernon at gmail.com Tue Feb 25 16:02:05 2014 From: jmckernon at gmail.com (James Mckernon) Date: Tue, 25 Feb 2014 16:02:05 +0000 Subject: [LAU] compiling jack-osc? In-Reply-To: References: Message-ID: I think I've got it working. Apologies for the false alarm. I was missing freeglut, which appears to be necessary. I also didn't realize that the best way to get the files was with darcs CVS; instead I was just grabbing them with wget. So retrieving the files the correct way seemed to help, too. On Tue, Feb 25, 2014 at 3:51 PM, Paul Davis wrote: > > > > On Tue, Feb 25, 2014 at 10:37 AM, James Mckernon > wrote: >> >> I'm trying to compile (or otherwise obtain) the jackd utilities found >> at http://rd.slavepianos.org/sw/rju/ , but I don't seem to be able to. >> In particular, I'm interested in using the jack-osc tool to convert >> jack transport messages (from Ardour) to osc messages. (Incidentally, >> if anyone knows of another way of doing this, please let me know.) >> Although the sources are there, I can't figure out how to compile them >> - in fact I don't even know if they're in a compileable state. I tried >> running make and playing around with gcc, but they seem to depend on >> some files that aren't present there. I'm running Arch Linux, but >> can't find a package for them in the AUR (Arch user repository). >> >> Are these tools still current, or have they succumbed to bit rot? Does >> anyone know how I might be able to compile them? > > > how about some actual compile-time errors? > From rm at mh-freiburg.de Tue Feb 25 17:41:37 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Tue, 25 Feb 2014 18:41:37 +0100 Subject: [LAU] compiling jack-osc? In-Reply-To: References: Message-ID: <20140225173826.M88396@mh-freiburg.de> On Tue, 25 Feb 2014 16:02:05 +0000, James Mckernon wrote > I think I've got it working. Apologies for the false alarm. > > I was missing freeglut, which appears to be necessary. I also didn't > realize that the best way to get the files was with darcs CVS; > instead I was just grabbing them with wget. So retrieving the files the > correct way seemed to help, too. The main trick after checking out the rju darcs repository is to check out c-common from within the rju directory: darcs clone http://rd.slavepianos.org/sw/rju/ cd rju darcs clone http://rd.slavepianos.org/sw/c-common/ cd c-common make cd .. make HTH RalfD From rm at mh-freiburg.de Tue Feb 25 17:43:04 2014 From: rm at mh-freiburg.de (R. Mattes) Date: Tue, 25 Feb 2014 18:43:04 +0100 Subject: [LAU] compiling jack-osc? In-Reply-To: References: Message-ID: <20140225174304.M18120@mh-freiburg.de> On Tue, 25 Feb 2014 16:02:05 +0000, James Mckernon wrote > I think I've got it working. Apologies for the false alarm. > > I was missing freeglut, which appears to be necessary. I also didn't > realize that the best way to get the files was with darcs CVS; > instead I was just grabbing them with wget. So retrieving the files the > correct way seemed to help, too. The main trick after checking out the rju darcs repository is to check out c-common from within the rju directory: darcs clone http://rd.slavepianos.org/sw/rju/ cd rju darcs clone http://rd.slavepianos.org/sw/c-common/ cd c-common make cd .. make HTH RalfD From tweed at lollipopfactory.com Tue Feb 25 17:58:40 2014 From: tweed at lollipopfactory.com (Tweed) Date: Tue, 25 Feb 2014 12:58:40 -0500 Subject: [LAU] compiling jack-osc? In-Reply-To: <20140225173826.M88396@mh-freiburg.de> References: <20140225173826.M88396@mh-freiburg.de> Message-ID: <530CD9D0.1070801@lollipopfactory.com> On 02/25/2014 12:41 PM, R. Mattes wrote: > On Tue, 25 Feb 2014 16:02:05 +0000, James Mckernon wrote >> I think I've got it working. Apologies for the false alarm. >> >> I was missing freeglut, which appears to be necessary. I also didn't >> realize that the best way to get the files was with darcs CVS; >> instead I was just grabbing them with wget. So retrieving the files the >> correct way seemed to help, too. > The main trick after checking out the rju darcs repository is to > check out c-common from within the rju directory: > > darcs clone http://rd.slavepianos.org/sw/rju/ > cd rju > darcs clone http://rd.slavepianos.org/sw/c-common/ > cd c-common > make > cd .. > make > > HTH RalfD > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Thank You Ralf! was banging me head on that one me was :) I was even in rju's parent directory and missed c-common. doh. thanks again -- the-temp-agency.com/lollipop-factory From rmouneyres at gmail.com Tue Feb 25 18:10:28 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Tue, 25 Feb 2014 19:10:28 +0100 Subject: [LAU] compiling jack-osc? In-Reply-To: <530CD9D0.1070801@lollipopfactory.com> References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> Message-ID: hello, i'm interested in compiling on arch too. which darcs package are you using from the AUR ? 2014-02-25 18:58 UTC+01:00, Tweed : > On 02/25/2014 12:41 PM, R. Mattes wrote: >> On Tue, 25 Feb 2014 16:02:05 +0000, James Mckernon wrote >>> I think I've got it working. Apologies for the false alarm. >>> >>> I was missing freeglut, which appears to be necessary. I also didn't >>> realize that the best way to get the files was with darcs CVS; >>> instead I was just grabbing them with wget. So retrieving the files the >>> correct way seemed to help, too. >> The main trick after checking out the rju darcs repository is to >> check out c-common from within the rju directory: >> >> darcs clone http://rd.slavepianos.org/sw/rju/ >> cd rju >> darcs clone http://rd.slavepianos.org/sw/c-common/ >> cd c-common >> make >> cd .. >> make >> >> HTH RalfD >> >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> > Thank You Ralf! was banging me head on that one me was :) > I was even in rju's parent directory and missed c-common. doh. > thanks again > > -- > the-temp-agency.com/lollipop-factory > > _______________________________________________ > 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 Tue Feb 25 18:12:34 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Tue, 25 Feb 2014 19:12:34 +0100 Subject: [LAU] OT: Pierrot Lunaire (was: Re: Exam Cheating investigation) In-Reply-To: <20140119121757.3d8d630f@telecino> References: <1389820387.64828.YahooMailNeo@web122601.mail.ne1.yahoo.com> <52DA818C.7050302@gmx.net> <201401181324.22782.gheskett@wdtv.com> <20140118214833.GA11978@linuxaudio.org> <20140119121757.3d8d630f@telecino> Message-ID: <1393351954.2355.6.camel@archlinux> On Sun, 2014-01-19 at 12:17 -0500, Marc Lavall?e wrote: > Pierrot Lunaire is indeed easy to identify In the Internet I found a nice wallpaper for my Jwm desktop. http://picpaste.com/pics/Pierrot_lunaire-Melodrama_von_Sch__nberg.-zdSoXgwa.1393350545.png A few years later after Sch?nberg painted (IMO his music is better than his paintings are), he perhaps would have painted in a similar way. From tweed at lollipopfactory.com Tue Feb 25 18:18:38 2014 From: tweed at lollipopfactory.com (Tweed) Date: Tue, 25 Feb 2014 13:18:38 -0500 Subject: [LAU] compiling jack-osc? In-Reply-To: References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> Message-ID: <530CDE7E.1040504@lollipopfactory.com> On 02/25/2014 01:10 PM, Rapha?l Mouneyres wrote: > hello, i'm interested in compiling on arch too. > which darcs package are you using from the AUR ? > > > 2014-02-25 18:58 UTC+01:00, Tweed : >> On 02/25/2014 12:41 PM, R. Mattes wrote: >>> On Tue, 25 Feb 2014 16:02:05 +0000, James Mckernon wrote >>>> I think I've got it working. Apologies for the false alarm. >>>> >>>> I was missing freeglut, which appears to be necessary. I also didn't >>>> realize that the best way to get the files was with darcs CVS; >>>> instead I was just grabbing them with wget. So retrieving the files the >>>> correct way seemed to help, too. >>> The main trick after checking out the rju darcs repository is to >>> check out c-common from within the rju directory: >>> >>> darcs clone http://rd.slavepianos.org/sw/rju/ >>> cd rju >>> darcs clone http://rd.slavepianos.org/sw/c-common/ >>> cd c-common >>> make >>> cd .. >>> make >>> >>> HTH RalfD >>> >>> >>> _______________________________________________ >>> Linux-audio-user mailing list >>> Linux-audio-user at lists.linuxaudio.org >>> http://lists.linuxaudio.org/listinfo/linux-audio-user >>> >> Thank You Ralf! was banging me head on that one me was :) >> I was even in rju's parent directory and missed c-common. doh. >> thanks again >> >> -- >> the-temp-agency.com/lollipop-factory >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> I'm on debian squeeze. AvLinux. darcs 2.4.4-3. Don't know about aur packages for Arch. Here's dl info: http://darcs.net/Binaries -- the-temp-agency.com/lollipop-factory From jmckernon at gmail.com Tue Feb 25 18:42:37 2014 From: jmckernon at gmail.com (James Mckernon) Date: Tue, 25 Feb 2014 18:42:37 +0000 Subject: [LAU] compiling jack-osc? In-Reply-To: References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> Message-ID: On Tue, Feb 25, 2014 at 6:10 PM, Rapha?l Mouneyres wrote: > hello, i'm interested in compiling on arch too. > which darcs package are you using from the AUR ? I used darcs-bin as I had trouble with getting the dependencies to compile the other version. From rmouneyres at gmail.com Tue Feb 25 18:47:26 2014 From: rmouneyres at gmail.com (=?ISO-8859-1?Q?Rapha=EBl_Mouneyres?=) Date: Tue, 25 Feb 2014 19:47:26 +0100 Subject: [LAU] compiling jack-osc? In-Reply-To: References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> Message-ID: >I used darcs-bin as I had trouble with getting the dependencies to compile the other version. ok, i was worried about some outdated packages. it worked great with the precompiled binary on the darcs website pointed by Tweed. Plus it saved me from installing a one time use software. the tools are great, even the jack-plumbing is better than the jack.plumbing as it won't crash on unactivated ports (where you need a to patch jack.plumbing) and can use a specific plumbing file. I really like that ! 2014-02-25 19:42 UTC+01:00, James Mckernon : > On Tue, Feb 25, 2014 at 6:10 PM, Rapha?l Mouneyres > wrote: >> hello, i'm interested in compiling on arch too. >> which darcs package are you using from the AUR ? > > I used darcs-bin as I had trouble with getting the dependencies to > compile the other version. > From mail at peterodoherty.net Wed Feb 26 08:21:31 2014 From: mail at peterodoherty.net (Peter O'Doherty) Date: Wed, 26 Feb 2014 09:21:31 +0100 Subject: [LAU] jack + other audio Message-ID: <530DA40B.2000903@peterodoherty.net> Hi list, I'm know this has been asked a thousand times already but could someone please point me in the direction of instructions to enable non-jack applications (like vlc) to work concurrently with jack? The well-known situation is that jack "hogs" audio and makes other applications unusable. How to get around this? Many thanks, Peter -- //============================= -> Peter O'Doherty -> http://www.peterodoherty.net -> mail at peterodoherty.net //============================= From pshirkey at boosthardware.com Wed Feb 26 08:56:07 2014 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Wed, 26 Feb 2014 19:56:07 +1100 (EST) Subject: [LAU] jack + other audio In-Reply-To: <530DA40B.2000903@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> Message-ID: <56165.86.105.95.182.1393404967.squirrel@boosthardware.com> On Wed, February 26, 2014 7:21 pm, Peter O'Doherty wrote: > Hi list, > > I'm know this has been asked a thousand times already but could someone > please point me in the direction of instructions to enable non-jack > applications (like vlc) to work concurrently with jack? The well-known > situation is that jack "hogs" audio and makes other applications > unusable. How to get around this? > There are three options: 1: use pulse audio with jack-sink - easy 2: use the jack-alsa plugin for alsa - not so easy 3: get a second audio device just for normal playback - $$$ (also pretty easy) -- Patrick Shirkey Boost Hardware Ltd From joelz at pobox.com Wed Feb 26 09:25:53 2014 From: joelz at pobox.com (Joel Roth) Date: Tue, 25 Feb 2014 23:25:53 -1000 Subject: [LAU] jack + other audio In-Reply-To: <530DA40B.2000903@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> Message-ID: <20140226092553.GA24056@sprite> On Wed, Feb 26, 2014 at 09:21:31AM +0100, Peter O'Doherty wrote: > Hi list, > > I'm know this has been asked a thousand times already but could > someone please point me in the direction of instructions to enable > non-jack applications (like vlc) to work concurrently with jack? The > well-known situation is that jack "hogs" audio and makes other > applications unusable. How to get around this? Today I used these instructions successfully to route ALSA and Pulseaudio outputs through JACK. http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack For Skype to work via Pulseaudio on my amd64 Debian system I needed to install an additional library: libpulse0:i386 Greetings, Joel > Many thanks, > Peter > > > -- > //============================= > -> Peter O'Doherty > -> http://www.peterodoherty.net > -> mail at peterodoherty.net > //============================= > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Joel Roth From mail at peterodoherty.net Wed Feb 26 09:44:49 2014 From: mail at peterodoherty.net (Peter O'Doherty) Date: Wed, 26 Feb 2014 10:44:49 +0100 Subject: [LAU] jack + other audio In-Reply-To: <56165.86.105.95.182.1393404967.squirrel@boosthardware.com> References: <530DA40B.2000903@peterodoherty.net> <56165.86.105.95.182.1393404967.squirrel@boosthardware.com> Message-ID: <530DB791.1050703@peterodoherty.net> Thanks. I've installed pulseaudio-module-jack but "Pulseaudio JACK Sink" is not showing up in the connections in qjackctl or in volume control. What am I missing? In case it's relevant I'm running Ubuntu 12.04 and qjackctl 0.3.8. On 02/26/2014 09:56 AM, Patrick Shirkey wrote: > On Wed, February 26, 2014 7:21 pm, Peter O'Doherty wrote: >> Hi list, >> >> I'm know this has been asked a thousand times already but could someone >> please point me in the direction of instructions to enable non-jack >> applications (like vlc) to work concurrently with jack? The well-known >> situation is that jack "hogs" audio and makes other applications >> unusable. How to get around this? >> > > There are three options: > > 1: use pulse audio with jack-sink - easy > > 2: use the jack-alsa plugin for alsa - not so easy > > 3: get a second audio device just for normal playback - $$$ (also pretty > easy) > > > > > > > > -- > 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 > -- //============================= -> Peter O'Doherty -> http://www.peterodoherty.net -> mail at peterodoherty.net //============================= From mail at peterodoherty.net Wed Feb 26 09:55:00 2014 From: mail at peterodoherty.net (Peter O'Doherty) Date: Wed, 26 Feb 2014 10:55:00 +0100 Subject: [LAU] jack + other audio In-Reply-To: <530DB791.1050703@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> <56165.86.105.95.182.1393404967.squirrel@boosthardware.com> <530DB791.1050703@peterodoherty.net> Message-ID: <530DB9F4.7070809@peterodoherty.net> Please disregard this last email. It's working as it should now. Thanks again. Peter On 02/26/2014 10:44 AM, Peter O'Doherty wrote: > Thanks. > > I've installed pulseaudio-module-jack but "Pulseaudio JACK Sink" is > not showing up in the connections in qjackctl or in volume control. > What am I missing? > > In case it's relevant I'm running Ubuntu 12.04 and qjackctl 0.3.8. > > > On 02/26/2014 09:56 AM, Patrick Shirkey wrote: >> On Wed, February 26, 2014 7:21 pm, Peter O'Doherty wrote: >>> Hi list, >>> >>> I'm know this has been asked a thousand times already but could someone >>> please point me in the direction of instructions to enable non-jack >>> applications (like vlc) to work concurrently with jack? The well-known >>> situation is that jack "hogs" audio and makes other applications >>> unusable. How to get around this? >>> >> >> There are three options: >> >> 1: use pulse audio with jack-sink - easy >> >> 2: use the jack-alsa plugin for alsa - not so easy >> >> 3: get a second audio device just for normal playback - $$$ (also pretty >> easy) >> >> >> >> >> >> >> >> -- >> 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 >> > > -- //============================= -> Peter O'Doherty -> http://www.peterodoherty.net -> mail at peterodoherty.net //============================= From simonzwise at gmail.com Wed Feb 26 11:39:42 2014 From: simonzwise at gmail.com (Simon Wise) Date: Wed, 26 Feb 2014 22:39:42 +1100 Subject: [LAU] jack + other audio In-Reply-To: <530DA40B.2000903@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> Message-ID: <530DD27E.300@gmail.com> On 26/02/14 19:21, Peter O'Doherty wrote: > Hi list, > > I'm know this has been asked a thousand times already but could someone please > point me in the direction of instructions to enable non-jack applications (like > vlc) to work concurrently with jack? The well-known situation is that jack > "hogs" audio and makes other applications unusable. How to get around this? The situation is ALSA only allows for one connection at a time, so anything connected directly locks everything else out. So you need some other layer to share your device. Either pulse or jack will share your ALSA device for you, but only one of them can be connected to ALSA at a time. Pulse does not support the connection jack needs since jack provides types of access to the device that pulse does not. So jack cannot connect via pulse, hence you have to connect pulse via jack instead. This was difficult some time ago, but now the solutions offered earlier in the thread exist. Pulse and jack cater for very different needs. Simon From itarozzi at gmail.com Wed Feb 26 17:24:59 2014 From: itarozzi at gmail.com (Ivan Tarozzi) Date: Wed, 26 Feb 2014 18:24:59 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: References: <530B6C2C.7080305@gmail.com> Message-ID: <530E236B.70608@gmail.com> Il 25/02/2014 09:10, Carlos sanchiavedraz ha scritto: > 2014-02-24 16:58 GMT+01:00 Ivan Tarozzi : >> Hi Carlos, >> I'm developing an industrial application for a client using APC Rock. >> >> I'm using Android, so I can confirm that linux runs on it and you can >> find kernel and uboot here: >> https://github.com/apc-io/apc-rock >> >> I never tried a "standard" linux ditributions, so I can't tell about it, >> but I fear the VIA support against linux is absent. >> >> I suggest you look at some better supported board. >> >> just my2c >> >> Ivan >> >> > Thanks so much, Ivan. > It's really helpful to have such a warn before wanting to purchase one > of those and make my life trickier, most of all when surely I would > try to put Musix inside that would have some driver problems (it's > 100% free/libre based on Debian). > > I liked the book-style one, but now maybe I'll go some other way. > > Regards. > Hi Carlos, I fear you can't put musix on any ARM board. Pay attention to the arch when you download an iso. I can't find any musix ARM version in the musix mirrors. Of course you can start from a debian (or other supported distro) and then add the programs that you want. If a packet is absent in arm repo, you can of course compile yourself. So, consider all that said when you plan to buy a board and download software for it. Now I have a beaglebone black on my desk. I haven't found spare time to test it, but next days I hope to try the debian for BBB. Here some references: http://beagleboard.org/Products/BeagleBone+Black http://elinux.org/Beagleboard:Debian_On_BeagleBone_Black I have no idea about debian repo for arm, I don't know if all packages that I found in my amd64 repo are present in arm repository. Or I could be for the Ubuntu way (I dislike a bit, but here the UBUNTU ARM wiki page: https://wiki.ubuntu.com/ARM) I will report you when I completed some test, if you want. Cheers! Ivan From csanchezgs at gmail.com Wed Feb 26 19:12:01 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Wed, 26 Feb 2014 20:12:01 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E236B.70608@gmail.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> Message-ID: 2014-02-26 18:24 GMT+01:00 Ivan Tarozzi : > > Il 25/02/2014 09:10, Carlos sanchiavedraz ha scritto: >> 2014-02-24 16:58 GMT+01:00 Ivan Tarozzi : >>> Hi Carlos, >>> I'm developing an industrial application for a client using APC Rock. >>> >>> I'm using Android, so I can confirm that linux runs on it and you can >>> find kernel and uboot here: >>> https://github.com/apc-io/apc-rock >>> >>> I never tried a "standard" linux ditributions, so I can't tell about it, >>> but I fear the VIA support against linux is absent. >>> >>> I suggest you look at some better supported board. >>> >>> just my2c >>> >>> Ivan >>> >>> >> Thanks so much, Ivan. >> It's really helpful to have such a warn before wanting to purchase one >> of those and make my life trickier, most of all when surely I would >> try to put Musix inside that would have some driver problems (it's >> 100% free/libre based on Debian). >> >> I liked the book-style one, but now maybe I'll go some other way. >> >> Regards. >> > > Hi Carlos, > I fear you can't put musix on any ARM board. Pay attention to the arch > when you download an iso. > I can't find any musix ARM version in the musix mirrors. > Yes, I know, I'm part of Musix team ;). But I think I could copy the main differential things of Musix in a Debian ARM. > > Of course you can start from a debian (or other supported distro) and > then add the programs that you want. > If a packet is absent in arm repo, you can of course compile yourself. > > So, consider all that said when you plan to buy a board and download > software for it. Good advices. > > Now I have a beaglebone black on my desk. I haven't found spare time to > test it, but next days I hope to try the debian for BBB. > > Here some references: > http://beagleboard.org/Products/BeagleBone+Black > http://elinux.org/Beagleboard:Debian_On_BeagleBone_Black > Our dear co-lister here, Jeremy Jongepier (http://autostatic.com/) , I think he wrote a post about Beagleboard on his blog time ago. He is very active trying this kind of mini-PCs. I think it gave him some headaches. > I have no idea about debian repo for arm, I don't know if all packages > that I found in my amd64 repo are present in arm repository. > Or I could be for the Ubuntu way (I dislike a bit, but here the UBUNTU > ARM wiki page: https://wiki.ubuntu.com/ARM) > > I will report you when I completed some test, if you want. It would be great! > > Cheers! > Ivan > > > > Thanks, Ivan. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From simonzwise at gmail.com Wed Feb 26 20:12:06 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 27 Feb 2014 07:12:06 +1100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E236B.70608@gmail.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> Message-ID: <530E4A96.9000402@gmail.com> On 27/02/14 04:24, Ivan Tarozzi wrote: > > Il 25/02/2014 09:10, Carlos sanchiavedraz ha scritto: >> 2014-02-24 16:58 GMT+01:00 Ivan Tarozzi: >>> Hi Carlos, >>> I'm developing an industrial application for a client using APC Rock. >>> >>> I'm using Android, so I can confirm that linux runs on it and you can >>> find kernel and uboot here: >>> https://github.com/apc-io/apc-rock >>> >>> I never tried a "standard" linux ditributions, so I can't tell about it, >>> but I fear the VIA support against linux is absent. >>> >>> I suggest you look at some better supported board. >>> >>> just my2c >>> >>> Ivan >>> >>> >> Thanks so much, Ivan. >> It's really helpful to have such a warn before wanting to purchase one >> of those and make my life trickier, most of all when surely I would >> try to put Musix inside that would have some driver problems (it's >> 100% free/libre based on Debian). >> >> I liked the book-style one, but now maybe I'll go some other way. >> >> Regards. >> > > Hi Carlos, > I fear you can't put musix on any ARM board. Pay attention to the arch > when you download an iso. > I can't find any musix ARM version in the musix mirrors. > > > Of course you can start from a debian (or other supported distro) and > then add the programs that you want. > If a packet is absent in arm repo, you can of course compile yourself. > > So, consider all that said when you plan to buy a board and download > software for it. > > Now I have a beaglebone black on my desk. I haven't found spare time to > test it, but next days I hope to try the debian for BBB. > > Here some references: > http://beagleboard.org/Products/BeagleBone+Black > http://elinux.org/Beagleboard:Debian_On_BeagleBone_Black > > I have no idea about debian repo for arm, I don't know if all packages > that I found in my amd64 repo are present in arm repository. > Or I could be for the Ubuntu way (I dislike a bit, but here the UBUNTU > ARM wiki page: https://wiki.ubuntu.com/ARM) ARM is a very variable target ... debian has two ARM versions which will suit some devices .... and are a reasonable starting point for trying to get a new device working. There is a raspberry specific repo based on one of them, where a huge amount of work has been put in to make it run well on a raspberry ... that work depended in part on support from broadcom and is possibly the biggest thing that makes the raspberries an interesting platform for me. Getting GNU/linux running on a specific ARM device is serious work, the manufacturers have often put work into running Android/linux but that is a very different platform. Ubuntu is working on supporting some phones with its GNU/linux, which could be nice to have. Simon From pablo.fbus at gmail.com Wed Feb 26 20:30:09 2014 From: pablo.fbus at gmail.com (=?ISO-8859-1?Q?Pablo_Fern=E1ndez?=) Date: Wed, 26 Feb 2014 21:30:09 +0100 Subject: [LAU] jack + other audio In-Reply-To: <530DA40B.2000903@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> Message-ID: <530E4ED1.2070105@gmail.com> El 26/02/14 09:21, Peter O'Doherty escribi?: > Hi list, > > I'm know this has been asked a thousand times already but could > someone please point me in the direction of instructions to enable > non-jack applications (like vlc) to work concurrently with jack? The > well-known situation is that jack "hogs" audio and makes other > applications unusable. How to get around this? > > Many thanks, > Peter > As you mention vlc, note that many multimedia players feature a jack audio output plugin. In the case of vlc (and in debian-based distros at least) this doesn't work by default but you have to install a separate package, "vlc-plugin-jack" (and then enable jack in the audio preferences). From jeremy at autostatic.com Wed Feb 26 20:48:25 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 26 Feb 2014 21:48:25 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E4A96.9000402@gmail.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> <530E4A96.9000402@gmail.com> Message-ID: <530E5319.7090405@autostatic.com> On 02/26/2014 09:12 PM, Simon Wise wrote: > ARM is a very variable target ... debian has two ARM versions which will > suit some devices .... armel and armhf should cover most devices. Haven't come across an ARM based device (except phones) yet that doesn't run Debian. and are a reasonable starting point for trying to > get a new device working. There is a raspberry specific repo based on > one of them, where a huge amount of work has been put in to make it run > well on a raspberry ... that work depended in part on support from > broadcom and is possibly the biggest thing that makes the raspberries an > interesting platform for me. > Afaik Raspbian was and still is a community project unrelated to the The Foundation itself. Rasbian was needed because Debian armel couldn't benefit from the floating point hardware of the RPi. > Getting GNU/linux running on a specific ARM device is serious work, the > manufacturers have often put work into running Android/linux but that is > a very different platform. Ubuntu is working on supporting some phones > with its GNU/linux, which could be nice to have. > True but once folks got Linux running on specific ARM devices and documented the installation steps this can make things a lot easier :) Bye, Jeremy > Simon -------------- 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 Wed Feb 26 20:50:31 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 26 Feb 2014 21:50:31 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> Message-ID: <530E5397.40805@autostatic.com> On 02/26/2014 08:12 PM, Carlos sanchiavedraz wrote: > Our dear co-lister here, Jeremy Jongepier (http://autostatic.com/) , I > think he wrote a post about Beagleboard on his blog time ago. He is > very active trying this kind of mini-PCs. I think it gave him some > headaches. I did not have a decent enough power adapter for the BBB so I had issues getting USB devices, like audio interfaces, detected. I blogged about it here: http://autostatic.com/2013/09/17/exit-beaglebone-black-hello-cubieboard2 But that's just a personal experience. Maybe I did something wrong or I've looked over something obvious. At that time I also received a Cubieboard2 and that board immediately absorbed all my interest as it's more powerful and easier to set up. And it has onboard audio IO. 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 jeremy at autostatic.com Wed Feb 26 20:56:35 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 26 Feb 2014 21:56:35 +0100 Subject: [LAU] compiling jack-osc? In-Reply-To: <530CDE7E.1040504@lollipopfactory.com> References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> <530CDE7E.1040504@lollipopfactory.com> Message-ID: <530E5503.1080409@autostatic.com> On 02/25/2014 07:18 PM, Tweed wrote: > I'm on debian squeeze. Isn't jack-osc part of the Debian jack-tools package? Or is there a difference between jack-osc and jack.osc? 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 simonzwise at gmail.com Wed Feb 26 21:27:19 2014 From: simonzwise at gmail.com (Simon Wise) Date: Thu, 27 Feb 2014 08:27:19 +1100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E5319.7090405@autostatic.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> <530E4A96.9000402@gmail.com> <530E5319.7090405@autostatic.com> Message-ID: <530E5C37.4040207@gmail.com> On 27/02/14 07:48, Jeremy Jongepier wrote: > On 02/26/2014 09:12 PM, Simon Wise wrote: >> get a new device working. There is a raspberry specific repo based on >> one of them, where a huge amount of work has been put in to make it run >> well on a raspberry ... that work depended in part on support from >> broadcom and is possibly the biggest thing that makes the raspberries an >> interesting platform for me. >> > > Afaik Raspbian was and still is a community project unrelated to the The > Foundation itself. Rasbian was needed because Debian armel couldn't > benefit from the floating point hardware of the RPi. but very importantly it contains all the libraries and sample code, the broadcom code with community additions, to make use of the hardware in a GNU/linux context... which I happen to be immersed in this week. > >> Getting GNU/linux running on a specific ARM device is serious work, the >> manufacturers have often put work into running Android/linux but that is >> a very different platform. Ubuntu is working on supporting some phones >> with its GNU/linux, which could be nice to have. >> > > True but once folks got Linux running on specific ARM devices and > documented the installation steps this can make things a lot easier :) certainly, and the sooner the better. > > Bye, > > Jeremy > >> Simon > > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From tweed at lollipopfactory.com Wed Feb 26 22:13:00 2014 From: tweed at lollipopfactory.com (Tweed) Date: Wed, 26 Feb 2014 17:13:00 -0500 Subject: [LAU] compiling jack-osc? In-Reply-To: <530E5503.1080409@autostatic.com> References: <20140225173826.M88396@mh-freiburg.de> <530CD9D0.1070801@lollipopfactory.com> <530CDE7E.1040504@lollipopfactory.com> <530E5503.1080409@autostatic.com> Message-ID: <530E66EC.9080506@lollipopfactory.com> On 02/26/2014 03:56 PM, Jeremy Jongepier wrote: > On 02/25/2014 07:18 PM, Tweed wrote: >> I'm on debian squeeze. > Isn't jack-osc part of the Debian jack-tools package? Or is there a > difference between jack-osc and jack.osc? > > Bye, > > Jeremy > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Yes. they're both by rohan drape. outwardly they seem identical. jack-osc is jack.clock not sure what the differences are. having issues wiht jack-udp actually, "buffer underflow" and "out of order packet arrival". tried varoius buffers. doesn't help. probably just keep using netjack. I do like jack-transport though. -- the-temp-agency.com/lollipop-factory -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Wed Feb 26 22:19:44 2014 From: len at ovenwerks.net (Len Ovens) Date: Wed, 26 Feb 2014 14:19:44 -0800 (PST) Subject: [LAU] jack + other audio In-Reply-To: <530DB791.1050703@peterodoherty.net> References: <530DA40B.2000903@peterodoherty.net> <56165.86.105.95.182.1393404967.squirrel@boosthardware.com> <530DB791.1050703@peterodoherty.net> Message-ID: On Wed, 26 Feb 2014, Peter O'Doherty wrote: > Thanks. > > I've installed pulseaudio-module-jack but "Pulseaudio JACK Sink" is not > showing up in the connections in qjackctl or in volume control. What am I > missing? > > In case it's relevant I'm running Ubuntu 12.04 and qjackctl 0.3.8. One of two things have to happen. Either jack has to be run as jackdbus or the jack sink module has to be loaded manually after jackd is running. -- Len Ovens www.ovenwerks.net From atte at youmail.dk Thu Feb 27 07:29:25 2014 From: atte at youmail.dk (Atte) Date: Thu, 27 Feb 2014 08:29:25 +0100 Subject: [LAU] re Zoom R16 In-Reply-To: <1393278643930-89584.post@n7.nabble.com> References: <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <530EE955.40107@youmail.dk> On 02/24/2014 10:50 PM, sub_acoustic wrote: > Thanks Atte, > > Does that mean that I could download the latest UbuntuStudio distro, > including kernel then apply the zoom_quirks.txt, compile the kernel, install > it and reboot > into it...? Yeah, just make sure you have the kernel source, not sure if this is provided with unbuntustudio. > sounds tricky...perhaps the Ubuntu Studio developers would be so kind as to > add the quirk to the next distro... I think it should be included in the main linux kernel. Did anyone contact clemens at ladisch.de to ask for zoom r16/r24 to be added to quirks-table.h (just downloaded kernel 3.13.5 and AFAICT the zoom quirks are not in there)? > why are manufacturers so hesitant to support linux?, or at least share > information with Linux developers? they would sell so many more units... It's annoying, but I think it all boils down to the relatively small number of linux users... -- Atte http://atte.dk http://modlys.dk From gnome at hawaii.rr.com Thu Feb 27 07:45:37 2014 From: gnome at hawaii.rr.com (david) Date: Wed, 26 Feb 2014 21:45:37 -1000 Subject: [LAU] re Zoom R16 In-Reply-To: <530EE955.40107@youmail.dk> References: <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> <530EE955.40107@youmail.dk> Message-ID: <530EED21.5020506@hawaii.rr.com> On 02/26/2014 09:29 PM, Atte wrote: > On 02/24/2014 10:50 PM, sub_acoustic wrote: >> why are manufacturers so hesitant to support linux?, or at least share >> information with Linux developers? they would sell so many more units... > > It's annoying, but I think it all boils down to the relatively small > number of linux users... And the even smaller number of pro-audio Linux users. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From csanchezgs at gmail.com Thu Feb 27 08:35:16 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 27 Feb 2014 09:35:16 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E5397.40805@autostatic.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> <530E5397.40805@autostatic.com> Message-ID: 2014-02-26 21:50 GMT+01:00 Jeremy Jongepier : > On 02/26/2014 08:12 PM, Carlos sanchiavedraz wrote: >> Our dear co-lister here, Jeremy Jongepier (http://autostatic.com/) , I >> think he wrote a post about Beagleboard on his blog time ago. He is >> very active trying this kind of mini-PCs. I think it gave him some >> headaches. > > I did not have a decent enough power adapter for the BBB so I had issues > getting USB devices, like audio interfaces, detected. I blogged about it > here: > > http://autostatic.com/2013/09/17/exit-beaglebone-black-hello-cubieboard2 > > But that's just a personal experience. Maybe I did something wrong or > I've looked over something obvious. At that time I also received a > Cubieboard2 and that board immediately absorbed all my interest as it's > more powerful and easier to set up. And it has onboard audio IO. > > Bye, > > Jeremy > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Onboard audio IO on RPi is the first thing I miss in it, I liked the bookish APC just for that (and the look of it).Now I have to plug my USB audio interface for audio stuff (although distros like RaspBMC have good inner audio and HDMI support). That's what would make a really powerful headless portable platform, one thing to take with you to jam or when you don't need pro-audio (i.e. recording yourself or a band with an audio interface). One of my projects arose around this idea, is about some kind of Semantic Musical Dashboard to control presets (done) and (in the future and even with your voice) connecting things in Jackd. I started developing it on the necessity or preference of focusing on playing and jamming using FX and a looper and no screen. It is developed using HTML5/REST enabling control from the phone (or whatever thing that has a web browser) and a really light "server" that performs the actions on the RPi (it can be whatever SO) connected to a net; the rest is hands, instruments, a MIDI pedalboard, guitar/instrument and maybe an amp. Lately I'm sticking with my moto, +Muso -Tech. I'm experimenting with phones and tablets just because they have already a touch interface and IO (although maybe mic is not for pro stuff) and you almost always carry them with you. But this is another story right now until we have proper pro-audio on Linux based phones. But things seems are moving forward on this matter. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Thu Feb 27 08:37:00 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 27 Feb 2014 09:37:00 +0100 Subject: [LAU] APC devices for Audio In-Reply-To: <530E5C37.4040207@gmail.com> References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> <530E4A96.9000402@gmail.com> <530E5319.7090405@autostatic.com> <530E5C37.4040207@gmail.com> Message-ID: 2014-02-26 22:27 GMT+01:00 Simon Wise : > On 27/02/14 07:48, Jeremy Jongepier wrote: >> >> On 02/26/2014 09:12 PM, Simon Wise wrote: > > >>> get a new device working. There is a raspberry specific repo based on >>> one of them, where a huge amount of work has been put in to make it run >>> well on a raspberry ... that work depended in part on support from >>> broadcom and is possibly the biggest thing that makes the raspberries an >>> interesting platform for me. >>> >> >> Afaik Raspbian was and still is a community project unrelated to the The >> Foundation itself. Rasbian was needed because Debian armel couldn't >> benefit from the floating point hardware of the RPi. > > > but very importantly it contains all the libraries and sample code, the > broadcom code with community additions, to make use of the hardware in a > GNU/linux context... which I happen to be immersed in this week. > Please let me/us know about you're progress. > >> >>> Getting GNU/linux running on a specific ARM device is serious work, the >>> manufacturers have often put work into running Android/linux but that is >>> a very different platform. Ubuntu is working on supporting some phones >>> with its GNU/linux, which could be nice to have. >>> >> >> True but once folks got Linux running on specific ARM devices and >> documented the installation steps this can make things a lot easier :) > > > certainly, and the sooner the better. >> >> Bye, >> >> Jeremy >> >> >>> Simon >> >> >> >> >> >> _______________________________________________ >> 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 -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From atte at youmail.dk Thu Feb 27 10:42:28 2014 From: atte at youmail.dk (Atte) Date: Thu, 27 Feb 2014 11:42:28 +0100 Subject: [LAU] re Zoom R16 In-Reply-To: <1393278643930-89584.post@n7.nabble.com> References: <20131118162157.GA12034@linuxaudio.org> <1384792948823-87956.post@n7.nabble.com> <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <530F1694.7090205@youmail.dk> On 02/24/2014 10:50 PM, sub_acoustic wrote: > Thanks Atte, > > Does that mean that I could download the latest UbuntuStudio distro, > including kernel then apply the zoom_quirks.txt, compile the kernel, install > it and reboot > into it...? > > sounds tricky... Here's the script I use for building kernels, save this into a text file in ~/bin/kompile, make it executable with "chmod +x ~/bin/kompile", cd to /usr/src/ where you have your kernel source (make sure you have a symlink called "linux" to the kernel source "ln -s /usr/src/linux-3.13.5 /usr/src/linux") and that you have write permission to /usr/src (on debian add yourself to the src group), and run "kompile". Then you should get a .deb in /usr/src you can install with "sudo dpkg -i". Hope it helps... ---------------------------------------------------------------- #!/bin/bash cores=$(getconf _NPROCESSORS_ONLN) if [ `pwd` != '/usr/src' ]; then echo This script must be run from /usr/src exit fi if [ ! -d linux ]; then echo No symlink or directory '"linux"' found exit fi revision=0 cd linux pushd . V=`grep -e ^VERSION Makefile | sed -e 's/.*=//g'` P=`grep -e ^PATCHLEVEL Makefile | sed -e 's/.*=//g'` S=`grep -e ^SUBLEVEL Makefile | sed -e 's/.*=//g'` E=`grep -e ^EXTRAVERSION Makefile | sed -e 's/.*=//g'` version=`echo $V.$P.$S$E | sed 's/ //g'` branch=`echo $V.$P | sed 's/ //g'` running=`uname -r` echo version: $version echo branch: $branch echo running: $running sed -rie 's/echo "\+"/#echo "\+"/' scripts/setlocalversion make-kpkg clean fakeroot make-kpkg -j$cores --initrd --revision=$revision kernel_image --------------------------------------------------------------------- -- Atte http://atte.dk http://modlys.dk From hanswil at notam02.no Thu Feb 27 17:40:49 2014 From: hanswil at notam02.no (Hans Wilmers) Date: Thu, 27 Feb 2014 18:40:49 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <20140224205256.2F8D162BFC@lists.linuxaudio.org> References: <20140224205256.2F8D162BFC@lists.linuxaudio.org> Message-ID: <530F78A1.1040200@notam02.no> On 02/24/2014 08:44 PM, Ben Bell wrote: > > As others have pointed out the audio out isn't audiophile quality, but I'd > have thought if you were in a studio you'd use proper hardware and this > would be for live use? In which case, factory in an amp, an audience talking > and so on, and I don't think it's as big an issue as people make out. > Has anybody here tried out this hardware? http://www.element14.com/community/community/raspberry-pi/raspberry-pi-accessories/wolfson_pi / Hans From jeremy at autostatic.com Thu Feb 27 19:23:52 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 27 Feb 2014 20:23:52 +0100 Subject: [LAU] raspi as midi synth In-Reply-To: <530F78A1.1040200@notam02.no> References: <20140224205256.2F8D162BFC@lists.linuxaudio.org> <530F78A1.1040200@notam02.no> Message-ID: <530F90C8.8040709@autostatic.com> On 02/27/2014 06:40 PM, Hans Wilmers wrote: > Has anybody here tried out this hardware? > http://www.element14.com/community/community/raspberry-pi/raspberry-pi-accessories/wolfson_pi > > > / Hans Is it finally available? The I'm going to order one right away! 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 hamish.low.net at gmx.com Thu Feb 27 22:07:44 2014 From: hamish.low.net at gmx.com (sub_acoustic) Date: Thu, 27 Feb 2014 14:07:44 -0800 (PST) Subject: [LAU] re Zoom R16 In-Reply-To: <530F1694.7090205@youmail.dk> References: <1384883176341-87971.post@n7.nabble.com> <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> Message-ID: <1393538864605-89649.post@n7.nabble.com> Atte, you're a legend Much appreciated! -- View this message in context: http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p89649.html Sent from the linux-audio-user mailing list archive at Nabble.com. From len at ovenwerks.net Thu Feb 27 22:18:26 2014 From: len at ovenwerks.net (Len Ovens) Date: Thu, 27 Feb 2014 14:18:26 -0800 (PST) Subject: [LAU] raspi as midi synth In-Reply-To: <530F78A1.1040200@notam02.no> References: <20140224205256.2F8D162BFC@lists.linuxaudio.org> <530F78A1.1040200@notam02.no> Message-ID: On Thu, 27 Feb 2014, Hans Wilmers wrote: > Has anybody here tried out this hardware? > http://www.element14.com/community/community/raspberry-pi/raspberry-pi-accessories/wolfson_pi Looks interesting. It does limit what other HW can be used with the board, but I would guess almost anything does that. They also warn against using a USB hub at the same time, but I guess a USB MIDI (with no USB mouse) would be ok. It does not say what the latency is or can be set to. No mention of jackd at all :) Their kernel has to be used right now, but the driver bits are available if you wish to roll your own kernel. -- Len Ovens www.ovenwerks.net From simonzwise at gmail.com Fri Feb 28 00:56:23 2014 From: simonzwise at gmail.com (Simon Wise) Date: Fri, 28 Feb 2014 11:56:23 +1100 Subject: [LAU] APC devices for Audio In-Reply-To: References: <530B6C2C.7080305@gmail.com> <530E236B.70608@gmail.com> <530E4A96.9000402@gmail.com> <530E5319.7090405@autostatic.com> <530E5C37.4040207@gmail.com> Message-ID: <530FDEB7.90401@gmail.com> On 27/02/14 19:37, Carlos sanchiavedraz wrote: > 2014-02-26 22:27 GMT+01:00 Simon Wise: >> On 27/02/14 07:48, Jeremy Jongepier wrote: >>> >>> On 02/26/2014 09:12 PM, Simon Wise wrote: >> >> >>>> get a new device working. There is a raspberry specific repo based on >>>> one of them, where a huge amount of work has been put in to make it run >>>> well on a raspberry ... that work depended in part on support from >>>> broadcom and is possibly the biggest thing that makes the raspberries an >>>> interesting platform for me. >>>> >>> >>> Afaik Raspbian was and still is a community project unrelated to the The >>> Foundation itself. Rasbian was needed because Debian armel couldn't >>> benefit from the floating point hardware of the RPi. >> >> >> but very importantly it contains all the libraries and sample code, the >> broadcom code with community additions, to make use of the hardware in a >> GNU/linux context... which I happen to be immersed in this week. >> > > Please let me/us know about you're progress. I'm following on from Antoine earlier work, playing videos with fading, masking, cueing, timing and such .. getting closer, it is in OSC controlled form but will be a pd object soon enough. when I get it running properly, which must be in the next week or so, I'll post the git. Simon From k.s.matheussen at gmail.com Fri Feb 28 07:31:52 2014 From: k.s.matheussen at gmail.com (Kjetil Matheussen) Date: Fri, 28 Feb 2014 08:31:52 +0100 Subject: [LAU] jack + other audio Message-ID: "Patrick Shirkey": > 1: use pulse audio with jack-sink - easy > But have you gotten decent latency this way? (i.e. at least less than 40ms) > 2: use the jack-alsa plugin for alsa - not so easy > Why is this not easy? It's just pasting a few lines into ~/.asoundrc, and then you're done... Pretty decent latency too. I'm pretty sure this solution is generally the best. http://jackaudio.org/routing_alsa