From abhayadevs at gmail.com Mon Jul 1 09:10:41 2013 From: abhayadevs at gmail.com (Abhayadev S) Date: Mon, 1 Jul 2013 14:40:41 +0530 Subject: [LAU] Linux Firewire Users Message-ID: Hi All, Anybody out there who is successful and full-time users of any Focusrite Firewire products (preferably Liquid Saffire 56) under Linux? Regards, Abhayadev S ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Jul 1 09:19:39 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 11:19:39 +0200 Subject: [LAU] Linux Firewire Users In-Reply-To: References: Message-ID: <51D149AB.3080903@autostatic.com> On 07/01/2013 11:10 AM, Abhayadev S wrote: > Hi All, > > Anybody out there who is successful and full-time users of any Focusrite > Firewire products (preferably Liquid Saffire 56) under Linux? > > Regards, > Abhayadev S Yes, Focusrite Saffire Pro 40 user here. Did you already check http://ffado.org/ ? Regards, Jeremy From abhayadevs at gmail.com Mon Jul 1 09:24:05 2013 From: abhayadevs at gmail.com (Abhayadev S) Date: Mon, 1 Jul 2013 14:54:05 +0530 Subject: [LAU] Linux Firewire Users In-Reply-To: <51D149AB.3080903@autostatic.com> References: <51D149AB.3080903@autostatic.com> Message-ID: yeah i checked and it says experimental ! how is the feature support ? can we get features of Saffire MixControl software? Regards, Abhayadev S http://sites.google.com/site/abhayadevs On Mon, Jul 1, 2013 at 2:49 PM, Jeremy Jongepier wrote: > On 07/01/2013 11:10 AM, Abhayadev S wrote: > >> Hi All, >> >> Anybody out there who is successful and full-time users of any Focusrite >> Firewire products (preferably Liquid Saffire 56) under Linux? >> >> Regards, >> Abhayadev S >> > > Yes, Focusrite Saffire Pro 40 user here. Did you already check > http://ffado.org/ ? > > Regards, > > Jeremy > ______________________________**_________________ > Linux-audio-user mailing list > Linux-audio-user at lists.**linuxaudio.org > http://lists.linuxaudio.org/**listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Jul 1 09:33:34 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 11:33:34 +0200 Subject: [LAU] Linux Firewire Users In-Reply-To: References: <51D149AB.3080903@autostatic.com> Message-ID: <51D14CEE.9030903@autostatic.com> On 07/01/2013 11:24 AM, Abhayadev S wrote: > yeah i checked and it says experimental ! how is the feature support ? can > we get features of Saffire MixControl software? Everything should work, except the DSP part. As I don't use the mixer software myself I can't tell what features are supported. But a lot of work has been done recently to create mixer templates for almost all Focusrite Saffire Pro devices. You should definitely ask on the FFADO user mailinglist about the status of your Pro 56. Regards, Jeremy From jeremy at autostatic.com Mon Jul 1 11:08:34 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 13:08:34 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <50772.188.26.169.14.1372620159.squirrel@boosthardware.com> References: <5194EBA0.6060204@kudla.org> <1368767257692-85191.post@n7.nabble.com> <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D08448.9060004@autostatic.com> <50772.188.26.169.14.1372620159.squirrel@boosthardware.com> Message-ID: <51D16332.6090203@autostatic.com> On 06/30/2013 09:22 PM, Patrick Shirkey wrote: > > On Mon, July 1, 2013 5:17 am, Jeremy Jongepier wrote: >> On 06/29/2013 11:57 PM, Patrick Shirkey wrote: >>> According to this thread it's an ARM specific debug method. >>> >>> http://www.spinics.net/lists/arm-kernel/msg19006.html >>> >>> Someone here might be able to help if you pastebin the revised patch >> >> Then I have to clean up first and diff the whole source with a clean >> source. Made a bit of a mess out of it so this could take some time. > > It may not be necessary. Just put the entire build into a tarball and > point out the location that that code fails. > >> And >> with 'here' you mean this list? The last lists I will ever post on will >> be kernel dev lists, I'm just a mere user you know ;) >> > > I'm sure that people on this list will be able to offer some advices. > > > -- > Patrick Shirkey > Boost Hardware Ltd Got it to compile but it doesn't boot. I've uploaded a tarball of my modified source: http://rk.autostatic.com/linux-rt This directory also contains a modified RT patch file. Jeremy From jeremy at autostatic.com Mon Jul 1 11:30:32 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 13:30:32 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> References: <5194EBA0.6060204@kudla.org> <1368767257692-85191.post@n7.nabble.com> <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> Message-ID: <51D16858.8090908@autostatic.com> On 06/29/2013 11:57 PM, Patrick Shirkey wrote: > Just for reference sake, if you try it with just -P enabled does it > perform any better? It does actually. Have to use lower frames/period setting(-p64) and sample rate (-r44100) settings though. Jeremy From pshirkey at boosthardware.com Mon Jul 1 11:50:26 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Mon, 1 Jul 2013 21:50:26 +1000 (EST) Subject: [LAU] Google TV device w/ AV out In-Reply-To: <51D16858.8090908@autostatic.com> References: <5194EBA0.6060204@kudla.org> <1368767257692-85191.post@n7.nabble.com> <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> Message-ID: <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> On Mon, July 1, 2013 9:30 pm, Jeremy Jongepier wrote: > On 06/29/2013 11:57 PM, Patrick Shirkey wrote: >> Just for reference sake, if you try it with just -P enabled does it >> perform any better? > > It does actually. Have to use lower frames/period setting(-p64) and > sample rate (-r44100) settings though. > This reminded me that if the audio device is communicating over the usb bus it should perform better if the sample rate is divisible by the period size. So if you use 48000 try a period size of -p48 or -p96, etc... Another possible assist is documented here: http://alsa.opensrc.org/Usb-audio#Tuning_USB_devices_for_minimal_latencies -- Patrick Shirkey Boost Hardware Ltd From jeremy at autostatic.com Mon Jul 1 12:00:16 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 14:00:16 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> References: <5194EBA0.6060204@kudla.org> <1368767257692-85191.post@n7.nabble.com> <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> Message-ID: <51D16F50.2080206@autostatic.com> On 07/01/2013 01:50 PM, Patrick Shirkey wrote: > This reminded me that if the audio device is communicating over the usb > bus it should perform better if the sample rate is divisible by the period > size. > > So if you use 48000 try a period size of -p48 or -p96, etc... > > Another possible assist is documented here: > > http://alsa.opensrc.org/Usb-audio#Tuning_USB_devices_for_minimal_latencies http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-January/066765.html Jeremy From pshirkey at boosthardware.com Mon Jul 1 14:12:43 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Tue, 2 Jul 2013 00:12:43 +1000 (EST) Subject: [LAU] Google TV device w/ AV out In-Reply-To: <51D16F50.2080206@autostatic.com> References: <1368767257692-85191.post@n7.nabble.com> <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> Message-ID: <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> On Mon, July 1, 2013 10:00 pm, Jeremy Jongepier wrote: > On 07/01/2013 01:50 PM, Patrick Shirkey wrote: >> This reminded me that if the audio device is communicating over the usb >> bus it should perform better if the sample rate is divisible by the >> period >> size. >> >> So if you use 48000 try a period size of -p48 or -p96, etc... >> >> Another possible assist is documented here: >> >> http://alsa.opensrc.org/Usb-audio#Tuning_USB_devices_for_minimal_latencies > > http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-January/066765.html > IIUC the potential issues with period sizes with non powers of two are at the application level. JACK is agnostic about it. Anyway it's a little spooky that you get better performance with -p64 than -p128 even in playback only mode. -- Patrick Shirkey Boost Hardware Ltd From jeremy at autostatic.com Mon Jul 1 14:44:13 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 01 Jul 2013 16:44:13 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> References: <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> Message-ID: <51D195BD.5000209@autostatic.com> On 07/01/2013 04:12 PM, Patrick Shirkey wrote: > IIUC the potential issues with period sizes with non powers of two are at > the application level. JACK is agnostic about it. > Yes, guitarix for instance doesn't like period sizes that are not a power of two. > Anyway it's a little spooky that you get better performance with -p64 than > -p128 even in playback only mode. Well I'm testing and haven't done any real stress-tests yet. Maybe it was a one-off. At least I get undistorted sound out of this device. Now if I could only figure out why JACK bails out every once so often. And why full-duplex doesn't work, well, it works but the audio coming out is distorted. Jeremy From pshirkey at boosthardware.com Mon Jul 1 15:16:12 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Tue, 2 Jul 2013 01:16:12 +1000 (EST) Subject: [LAU] Google TV device w/ AV out In-Reply-To: <51D195BD.5000209@autostatic.com> References: <51A27F1F.8010304@autostatic.com> <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> <51D195BD.5000209@autostatic.com> Message-ID: <62544.188.26.169.14.1372691772.squirrel@boosthardware.com> On Tue, July 2, 2013 12:44 am, Jeremy Jongepier wrote: > On 07/01/2013 04:12 PM, Patrick Shirkey wrote: >> IIUC the potential issues with period sizes with non powers of two are >> at >> the application level. JACK is agnostic about it. >> > > Yes, guitarix for instance doesn't like period sizes that are not a > power of two. > >> Anyway it's a little spooky that you get better performance with -p64 >> than >> -p128 even in playback only mode. > > Well I'm testing and haven't done any real stress-tests yet. Maybe it > was a one-off. At least I get undistorted sound out of this device. Now > if I could only figure out why JACK bails out every once so often. Does this happen with or without wifi enabled? > And > why full-duplex doesn't work, well, it works but the audio coming out is > distorted. > System load causing distortion on the Audio device in duplex mode. It suggests something is wrong with the bus and or interrupts but I don't see why it would be ok in playback only mode. Not sure if it is related but I have noticed that my old usb quattro often takes a while to warm up. Sometimes I have to run it for a few hours and start/stop jack several times to get undistorted output. Another issue is the Rockchip devs probably haven't tested the device at low latency with duplex mode so they haven't picked up that issue. We can try flagging it with them. I forgot what the audio chipset is. Can you post the output of cat /proc/asound/cards? -- Patrick Shirkey Boost Hardware Ltd From fons at linuxaudio.org Mon Jul 1 21:29:53 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 1 Jul 2013 21:29:53 +0000 Subject: [LAU] Behringer ADA8000 phase Message-ID: <20130701212952.GA25646@linuxaudio.org> Hello all, The incredible happens. The electronics of the 'Lampadario' at the Casa del Suono in Parma consists of a rack with an RME ADI468 converting MADI to 8 ADAT outputs, 8 Behringer ADA8000 converters, and 8 QSC amplifiers of 8 channels each. The rack was wired (very neatly) by a firm specialising in this sort of work. When I installed the software four years ago, I found out that 25 of the 64 channels had their phase inverted. For one of those it was an error in the speaker wiring, which was easy to correct. The other 24 corresponded exactly to 3 groups of 8, and the speaker wiring was OK. I assumed that the cables between the ADA8000 and the amps were to blame - this is a non-standard cable which had to be hand-made by the whoever did the wiring. If two technicians had worked on that, they could have had different ideas of what were the correct connections. Since I didn't want to take the rack apart, resolder 24 wires and put it all back, and since there was only one SW app driving the installation at that time, those 24 inversions were corrected for by that software. So far so good. Recently I re-measured the IRs of the whole thing. There were again 24 channels out of phase. But not the same ones. One of the groups of 8 had turned in-phase, and another one was now inverted. The only thing that has happened to the installation over the last years is that some of the Behringers failed (power supply blown up, one per year on average) and were replaced. So I checked those separately. And yes, some of them had their output phase inverted w.r.t. the others. Apparently the thing exists in two versions, but apart from measuring there's no way to tell which is which. So I'll have to recheck things each time any of them are replaced again. Thank $GOD we didn't use those for the WFS system. The incredible happens. 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 wizardofgosz at gmail.com Mon Jul 1 22:24:40 2013 From: wizardofgosz at gmail.com (Ricardus Vincente) Date: Mon, 01 Jul 2013 18:24:40 -0400 Subject: [LAU] Behringer ADA8000 phase In-Reply-To: <20130701212952.GA25646@linuxaudio.org> References: <20130701212952.GA25646@linuxaudio.org> Message-ID: <51D201A8.8030707@gmail.com> On 07/01/2013 05:29 PM, Fons Adriaensen wrote: > Hello all, > > The incredible happens. > > The electronics of the 'Lampadario' at the Casa del Suono in > Parma consists of a rack with an RME ADI468 converting MADI > to 8 ADAT outputs, 8 Behringer ADA8000 converters, and 8 QSC > amplifiers of 8 channels each. The rack was wired (very neatly) > by a firm specialising in this sort of work. > > When I installed the software four years ago, I found out > that 25 of the 64 channels had their phase inverted. For one > of those it was an error in the speaker wiring, which was easy > to correct. The other 24 corresponded exactly to 3 groups of 8, > and the speaker wiring was OK. I assumed that the cables between > the ADA8000 and the amps were to blame - this is a non-standard > cable which had to be hand-made by the whoever did the wiring. > If two technicians had worked on that, they could have had > different ideas of what were the correct connections. > > Since I didn't want to take the rack apart, resolder 24 wires > and put it all back, and since there was only one SW app driving > the installation at that time, those 24 inversions were corrected > for by that software. So far so good. > > Recently I re-measured the IRs of the whole thing. There > were again 24 channels out of phase. But not the same ones. > One of the groups of 8 had turned in-phase, and another > one was now inverted. > > The only thing that has happened to the installation over > the last years is that some of the Behringers failed (power > supply blown up, one per year on average) and were replaced. > So I checked those separately. And yes, some of them had their > output phase inverted w.r.t. the others. Apparently the thing > exists in two versions, but apart from measuring there's no > way to tell which is which. So I'll have to recheck things > each time any of them are replaced again. Thank $GOD we didn't > use those for the WFS system. > > The incredible happens. > > Ciao, Wow. Tricky. I think the original Black-faced ADATs would record the audio out of phase to tape, and then correct it on the way out. So if you had tapes recorded in a Black-faced ADAT, and played it in a silver XT, or had tapes recorded in a Silver XT, and played it in a Black-faced unit, you were 180 degrees out of phase on playback. Rich... From kvutter at frii.com Tue Jul 2 01:19:00 2013 From: kvutter at frii.com (Kevin Utter) Date: Mon, 1 Jul 2013 19:19:00 -0600 Subject: [LAU] setBfree Rotary Speaker sounds strange! In-Reply-To: <51CA8E64.7090709@autostatic.com> References: <77304259-314B-455C-9E62-24402115C074@frii.com> <51CA8E64.7090709@autostatic.com> Message-ID: <5039F30E-17CC-4B55-BA92-88C51A9A527A@frii.com> On Jun 26, 2013, at 12:47 AM, Jeremy Jongepier wrote: > Hello Kevin, > > You might want to report that issue here: https://github.com/pantherb/setBfree/issues Thanks much. I did that about 3 days ago. I'm hoping I did it correctly. I've seen nothing about anyone even seeing it, as far as I understand it. I think I filed it under Questions, but since I haven't done this before, I'm not quite sure of the procedure. Thanks again. Kevin From kvutter at frii.com Tue Jul 2 01:33:00 2013 From: kvutter at frii.com (Kevin Utter) Date: Mon, 1 Jul 2013 19:33:00 -0600 Subject: [LAU] Roland Octa-Capture In-Reply-To: <519733F0.4010502@ladisch.de> References: <0C7991E8-C23C-4D83-AF59-0A3527E41C62@frii.com> <5195DD19.40509@ladisch.de> <519733F0.4010502@ladisch.de> Message-ID: On May 18, 2013, at 1:55 AM, Clemens Ladisch wrote: >>> Please show the output of "lsusb -v" for this device. >> >> Its listed here. > > That was without "-v". > > Please run "lsusb -v -d 0582:0120". > Hi! I'm wondering if you got the attachment with that output, and if it provides any further information? Thanks much. Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: From derJonnyB at web.de Tue Jul 2 03:33:42 2013 From: derJonnyB at web.de (JonnyB) Date: Tue, 02 Jul 2013 05:33:42 +0200 Subject: [LAU] Midi controller keeps disconnecting Message-ID: <51D24A16.3010002@web.de> Hi all, I recently bought this midi controller: http://www.musicstore.de/en_EN/GBP/Fame-Tweak-49-Controller-Masterkeyboard/art-SYN0003694-000 It seems to be this Chinese thing: http://www.worlde.com.cn/products_detail/&productId=1496ebdb-c1c5-401b-b57c-6201634388b3&comp_stats=comp-FrontProducts_list01-1305628296986.html I was hoping midi is standardized enough that it just works in Linux, it doesn't ... But I'd rather keep it, 'cause i don't find anything comparable suiting my needs for this price. Now here's the problem: It basically gets recognized all the way down to alsa $aconnect -li client 24: 'USB Device 0x218:0x303' [type=kernel] 0 'USB Device 0x218:0x303 MIDI 1' but it keeps disconnecting/reconnecting. The syslog keeps cycling like this (about twice a second): Jul 2 03:22:03 pc kernel: [13863.876029] usb 6-1: new full-speed USB device number 102 using uhci_hcd Jul 2 03:22:03 pc pulseaudio[2458]: [pulseaudio] module-alsa-card.c: Failed to find a working profile. Jul 2 03:22:03 pc pulseaudio[2458]: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-WORLDE_FAME_Tweak_49-00-U0x2180x303" card_name="alsa_card.usb-WORLDE_FAME_Tweak_49-00-U0x2180x303" namereg_fail=false tsched=yes ignore_dB=no deferred_volume=yes card_properties="module-udev-detect.discovered=1""): initialization failed. Jul 2 03:22:03 pc kernel: [13864.288055] usb 6-1: USB disconnect, device number 102 (I just recognized) That it sometimes even disconnects, before it is handled by udev Jul 2 03:26:21 pc kernel: [14122.272020] usb 6-1: new full-speed USB device number 98 using uhci_hcd Jul 2 03:26:21 pc kernel: [14122.442212] usb 6-1: USB disconnect, device number 98 so this is some problem with the drivers/firmware?? Any suggestions are highly appreciated :) Thanks in advance, JonnyB N.B.: My OS is a Linux pc 3.2.0-45-generic-pae #70-Ubuntu SMP Wed May 29 20:31:05 UTC 2013 i686 i686 i386 GNU/Linux And yes, it's working in windoze. Problems with a different device from the same vendor were discussed here (but seems rather unrelated): http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-March/091142.html From espiritocz at gmail.com Tue Jul 2 04:04:28 2013 From: espiritocz at gmail.com (Milan Lazecky) Date: Tue, 2 Jul 2013 12:04:28 +0800 Subject: [LAU] Midi controller keeps disconnecting In-Reply-To: <51D24A16.3010002@web.de> References: <51D24A16.3010002@web.de> Message-ID: Hi. I cannot help more than by guessing that a possible direction would be to use windows drivers in linux...? I know it was possible for my SIM card device in notebook, so probably it can work here as well. Google.. Good luck. Milan 2013/7/2 JonnyB > Hi all, > > I recently bought this midi controller: > > http://www.musicstore.de/en_EN/GBP/Fame-Tweak-49-Controller-Masterkeyboard/art-SYN0003694-000 > It seems to be this Chinese thing: > > http://www.worlde.com.cn/products_detail/&productId=1496ebdb-c1c5-401b-b57c-6201634388b3&comp_stats=comp-FrontProducts_list01-1305628296986.html > > I was hoping midi is standardized enough that it just works in Linux, it > doesn't ... > But I'd rather keep it, 'cause i don't find anything comparable suiting > my needs for this price. > > Now here's the problem: > It basically gets recognized all the way down to alsa > > $aconnect -li > client 24: 'USB Device 0x218:0x303' [type=kernel] > 0 'USB Device 0x218:0x303 MIDI 1' > > but it keeps disconnecting/reconnecting. The syslog keeps cycling like > this (about twice a second): > > Jul 2 03:22:03 pc kernel: [13863.876029] usb 6-1: new full-speed USB > device number 102 using uhci_hcd > Jul 2 03:22:03 pc pulseaudio[2458]: [pulseaudio] module-alsa-card.c: > Failed to find a working profile. > Jul 2 03:22:03 pc pulseaudio[2458]: [pulseaudio] module.c: Failed to > load module "module-alsa-card" (argument: "device_id="2" > name="usb-WORLDE_FAME_Tweak_49-00-U0x2180x303" > card_name="alsa_card.usb-WORLDE_FAME_Tweak_49-00-U0x2180x303" > namereg_fail=false tsched=yes ignore_dB=no deferred_volume=yes > card_properties="module-udev-detect.discovered=1""): initialization failed. > Jul 2 03:22:03 pc kernel: [13864.288055] usb 6-1: USB disconnect, > device number 102 > > > (I just recognized) That it sometimes even disconnects, before it is > handled by udev > > Jul 2 03:26:21 pc kernel: [14122.272020] usb 6-1: new full-speed USB > device number 98 using uhci_hcd > Jul 2 03:26:21 pc kernel: [14122.442212] usb 6-1: USB disconnect, > device number 98 > > so this is some problem with the drivers/firmware?? > Any suggestions are highly appreciated :) > > Thanks in advance, > JonnyB > > > N.B.: > My OS is a Linux pc 3.2.0-45-generic-pae #70-Ubuntu SMP Wed May 29 > 20:31:05 UTC 2013 i686 i686 i386 GNU/Linux > And yes, it's working in windoze. > Problems with a different device from the same vendor were discussed > here (but seems rather unrelated): > > http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-March/091142.html > > > _______________________________________________ > 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 clemens at ladisch.de Tue Jul 2 06:44:15 2013 From: clemens at ladisch.de (Clemens Ladisch) Date: Tue, 02 Jul 2013 08:44:15 +0200 Subject: [LAU] Midi controller keeps disconnecting In-Reply-To: <51D24A16.3010002@web.de> References: <51D24A16.3010002@web.de> Message-ID: <51D276BF.6030907@ladisch.de> JonnyB wrote: > it keeps disconnecting/reconnecting. The syslog keeps cycling like > this (about twice a second): > > Jul 2 03:22:03 pc kernel: [13863.876029] usb 6-1: new full-speed USB device number 102 using uhci_hcd > [...] > Jul 2 03:22:03 pc kernel: [13864.288055] usb 6-1: USB disconnect, device number 102 "USB disconnect" typically is a hardware problem (controller/cable/ power), but in this case, this sounds like a power management bug in the device's firmware. Many MIDI devices have similar bugs, so automatic PM was disabled ... > My OS is a Linux pc 3.2.0-45-generic-pae #70-Ubuntu ... in version 3.2.0-47. Regards, Clemens From robin at gareus.org Tue Jul 2 16:20:52 2013 From: robin at gareus.org (Robin Gareus) Date: Tue, 02 Jul 2013 18:20:52 +0200 Subject: [LAU] setBfree Rotary Speaker sounds strange! In-Reply-To: <5039F30E-17CC-4B55-BA92-88C51A9A527A@frii.com> References: <77304259-314B-455C-9E62-24402115C074@frii.com> <51CA8E64.7090709@autostatic.com> <5039F30E-17CC-4B55-BA92-88C51A9A527A@frii.com> Message-ID: <51D2FDE4.4080602@gareus.org> On 07/02/2013 03:19 AM, Kevin Utter wrote: > > On Jun 26, 2013, at 12:47 AM, Jeremy Jongepier wrote: >> Hello Kevin, >> >> You might want to report that issue here: https://github.com/pantherb/setBfree/issues > > > Thanks much. I did that about 3 days ago. I'm hoping I did it correctly. I've seen nothing about anyone even seeing it, as far as I understand it. I think I filed it under Questions, but since I haven't done this before, I'm not quite sure of the procedure. Thanks again. > > Kevin Sorry for the delay. I did not yet have the time to look into it. robin From rustys.lists at gmail.com Tue Jul 2 16:40:50 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Tue, 2 Jul 2013 09:40:50 -0700 Subject: [LAU] command line cricket removal tool? Message-ID: Hi folks, Does any one know of a simple tool for removing crickets from my home studio? :-) Hahaha, my "studio" is now in the garage. and, I guess, when I opened the main door for some ventilation, someone crept in hoping that I would make them a star!!!! :-) I haven't actually caught the offending creature on "tape" but if I do, I guess I'll just have to try some phase reversal or something. :) Of course, most of this is in fun, I hope that by the time I make my first recording, he'll learn some studio manners, or died, or moved out, :-) but I hadn't thought of the possibility of a cricket taking up residence in the studio. :-) Rusty From fero.kiraly at gmail.com Tue Jul 2 16:53:46 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Tue, 2 Jul 2013 18:53:46 +0200 Subject: [LAU] linux-rt & presonus 1818VSL Message-ID: Dear colleagues ;) I have some troubles - read xruns - with this setup: archlinux - linux-rt(various versions)-jack-presonus1818VSL My aim is to get as low latency as possibile. I have tried various kernel versions (rt) and various JACK setups (48000/(256,128,64),3) but none of them was without xruns when running: pd - plugins - ardour now I am compiling time by time kernel from series 3.2 - because I've read they are good for this purpose - and I am trying... Does anybody have some experience with similar setup and with no xruns ? thanks fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at quirq.ukfsn.org Tue Jul 2 17:11:08 2013 From: lists at quirq.ukfsn.org (Q) Date: Tue, 02 Jul 2013 18:11:08 +0100 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: <51D309AC.3040309@quirq.ukfsn.org> On 02/07/13 17:40, Rusty Perez wrote: > Hi folks, > Does any one know of a simple tool for removing crickets from my home > studio? :-) > Hahaha, my "studio" is now in the garage. and, I guess, when I opened > the main door for some ventilation, someone crept in hoping that I > would make them a star!!!! :-) > I haven't actually caught the offending creature on "tape" but if I > do, I guess I'll just have to try some phase reversal or something. :) > > Of course, most of this is in fun, I hope that by the time I make my > first recording, he'll learn some studio manners, or died, or moved > out, :-) but I hadn't thought of the possibility of a cricket taking > up residence in the studio. :-) > > Rusty Move to the UK, the rain usually stops play ;-) Q From julien at mail.upb.de Tue Jul 2 17:12:09 2013 From: julien at mail.upb.de (Julien Claassen) Date: Tue, 2 Jul 2013 19:12:09 +0200 (CEST) Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: Hello Rusty! The best cricket removal tool of course is someone, who captures or kills it. The next best thing could be playing with phase inverted signals. If you have two microphones, you could connect them both, use one for your recording and direct another away from your own noise production. If possible as striahgt as you can towards the offender. I'm not absolutely sure about all the mechanics of such a process, but it shold work, if the signals are synchronous. Others might have to say more about that. the other option is filtering and EQ'ing. If th cricket doesn't cover too much sonic space or doesn't interfere too much with your intended frequencies, you could try an equaliser. You can sweep the bands and find the sweet spots for the cricket and lower them as much as possible. If the cricket offers a lot of high frequencies you could probably even work with some more drastic lowpass fitlering. that doesn't work for every instrument. The third - and from a technical point easiest - method would be to compose a lullaby or other nature-friendly music and claim, that the cricket is intentional and from your personal sound library. :-) Chirping regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From jh at brainiac.com Tue Jul 2 17:25:38 2013 From: jh at brainiac.com (Joe Hartley) Date: Tue, 2 Jul 2013 13:25:38 -0400 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: <20130702132538.1acb84ae4a5172aac3f92fd8@brainiac.com> On Tue, 2 Jul 2013 09:40:50 -0700 Rusty Perez wrote: > Hi folks, > Does any one know of a simple tool for removing crickets from my home > studio? :-) If you can't get rid of the cricket, try using him as part of the studio. Since the number of chirps per minute (cpm) is dependent on temperature, figure out how many cpm you need to mesh with the bpm of the music you're working on, and set the studio temp accordingly. Voila! Instant chirp track! To convert cricket chirps to degrees Fahrenheit, count number of chirps in 14 seconds then add 40 to get temperature. Example: 30 chirps + 40 = 70? F To convert cricket chirps to degrees Celsius, count number of chirps in 25 seconds, divide by 3, then add 4 to get temperature. Example: 48 chirps /(divided by) 3 + 4 = 20? C So if you're working on a piece in 4/4 that is 120 BPM, a 70? F studio will give you 1 chirp per measure. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From louigi.verona at gmail.com Tue Jul 2 17:41:18 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Tue, 2 Jul 2013 21:41:18 +0400 Subject: [LAU] Linux Audio podcast, episode001 Message-ID: Hey fellas! Just some thoughts to share, get the podcast here: http://www.louigiverona.ru/?page=projects&s=writings&t=linux&a=linux_podcast -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Tue Jul 2 17:49:17 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 2 Jul 2013 18:49:17 +0100 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: On Tue, Jul 2, 2013 at 5:53 PM, Fero Kiraly wrote: > I have some troubles - read xruns - with this setup: There's a lot more to removing Xruns from a system than just the kernel and JACK settings (although they are very important :) Known things that need to be done: -Interrupt requests of your hardware must be prioritized (rtirq script helps here) -Interrupt thread handlers (on RT kernel) must be prioritized -Thread priorities must be checked for JACK itself Some other things to look into are: -CPU scaling (causes Xruns here, although others say it works fine) -WiFi (must be disabled here: "iwconfig wlan0 txpower off", not just disconnected) -Graphics chip drivers (I'm using a standard xf86-video driver for my ATI HD2400XT) That's a good start anyway: If I find some time, I'll write a quick post on tuning an Arch system for maximum low-lat RT performance. > My aim is to get as low latency as possible. Please specify... How low is low :D I've achieved 0.3ms at 96kHz, running a CPU stress test on each core without Xruns, so I'm pretty happy with that... since the hardware (Echo Indigo DJ PCMCIA) can't handle any lower! > but none of them was without xruns when running: > pd - plugins - ardour PD is a bit of a xrun hog on my machine here: not bashing PD, I probably don't have it set up with ideally. But perhaps try running eg Ardour on its own for testing purposes? Or another simple app which isn't as dependent on the "patch" that's loaded..? HTH, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From dplist at free.fr Tue Jul 2 18:10:10 2013 From: dplist at free.fr (David) Date: Tue, 2 Jul 2013 20:10:10 +0200 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: <20130702201010.e07ade8570890334417f09da@free.fr> On Tue, 2 Jul 2013 19:12:09 +0200 (CEST) Julien Claassen wrote: > The best cricket removal tool of course is someone, who captures or > kills it. Why on Earth would you want to *kill* it ? -- David From robin at linuxaudio.org Tue Jul 2 18:27:32 2013 From: robin at linuxaudio.org (Robin Gareus) Date: Tue, 02 Jul 2013 20:27:32 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: <51D31B94.90800@linuxaudio.org> On 07/02/2013 07:49 PM, Harry van Haaren wrote: > On Tue, Jul 2, 2013 at 5:53 PM, Fero Kiraly wrote: >> I have some troubles - read xruns - with this setup: > There's a lot more to removing Xruns from a system than just the kernel and > JACK settings (although they are very important :) * linux 3.2.35-2 PREEMPT RT (debian's RT kernel recompiled with VSL1818 clock selector fix added) * jackd 1.9.10 (git) * Rui's rtirq script * jackd with -p64 -n2 (actually jackdbus) * presonus 1818VSL -> very reliable no x-runs > Known things that need to be done: > -Interrupt requests of your hardware must be prioritized (rtirq script > helps here) > -Interrupt thread handlers (on RT kernel) must be prioritized > -Thread priorities must be checked for JACK itself +1, +1, +1 > Some other things to look into are: > -CPU scaling (causes Xruns here, although others say it works fine) no problem here w/freq scaling. I have disabled C1E halt states and EIST in the Bios, though. > -WiFi (must be disabled here: "iwconfig wlan0 txpower off", not just > disconnected) Nor does wifi + networkmanager cause issues here. but in either case the 1818VSL here is on a dedicated hardware IRQ - not shared with any other devices. > -Graphics chip drivers (I'm using a standard xf86-video driver for my ATI > HD2400XT) Intel graphics here. 2c, robin From diego.simak at gmail.com Tue Jul 2 18:21:34 2013 From: diego.simak at gmail.com (Diego Simak) Date: Tue, 2 Jul 2013 15:21:34 -0300 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: 2013/7/2 Louigi Verona : > Hey fellas! > > Just some thoughts to share, get the podcast here: > http://www.louigiverona.ru/?page=projects&s=writings&t=linux&a=linux_podcast > > > > -- > Louigi Verona > http://www.louigiverona.ru/ > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Very interesting podcast I must say. I agree in most of your thoughts, sadly. ;-) I imagine some kind of automation editor reading the parameters values (OSC? CV?) from the plugin or standalone app, creating a graphical representation of the parameter value. Without any serious thinking I could say that if NON Mixer allow LV2 and DSSI it will be an enormeous approach for that "automation editor" just controlling the plugin parameter from NON Timeline. Any option is to have other plugin to automate exiting parameter under Ingen o Carla... since I'm not a coder I cannot start any other thought in how to do this. Thanks for sharing your thoughts. Diego From bearcat at feline-soul.com Tue Jul 2 18:41:33 2013 From: bearcat at feline-soul.com (=?UTF-8?B?QmVhcmNhdCDFnsOhbmRvcg==?=) Date: Tue, 2 Jul 2013 12:41:33 -0600 Subject: [LAU] DRC and mics ..little idea of what i'm doing.. Message-ID: Folks, I just moved into a new house and have my stereo setup. Granted some things will be moved around the room and some quilts will be going onto the walls, but it's sitting basically where it will sit. Something i've always wanted to do is apply DRC to my audio system. I just started looking into this and i know very little about mics etc. I think i'm going to get something along these lines: http://www.parts-express.com/pe/showdetl.cfm?partnumber=390-801 I have an asus xonar essence ST soundcard found here: http://www.asus.com/Sound_Cards_and_DigitaltoAnalog_Converters/Xonar_Essence_ST/ First off, i assume that even though it's got an input that *looks* like a TRS jack i can't just get an xlr-trs converter and plug it in and go? If you want to point my nose at "understanding mics and phantom power and why you're a doofus for thinking this will work" please do. Thanks, -- Bearcat M. ??ndor Feline Soul Systems LLC Voice: 872.CAT.SOUL (872.228.7685) Fax: 406.235.7070 Jabber/xmpp/gtalk/email: bearcat at feline-soul.net MSN: bearcatsandor at hotmail.com Yahoo: bearcatsandor AIM: bearcatmsandor -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Tue Jul 2 18:42:54 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Tue, 2 Jul 2013 20:42:54 +0200 Subject: [LAU] linux-rt & presonus 1818VSL Message-ID: I say hallo for Harry ! ;) F: I have some troubles - read xruns - with this setup: H: There's a lot more to removing Xruns from a system than just the kernel and JACK settings (although they are very important :) F: actually I know realTimeConfigQuickScan, so my output looks: *[paum at bookes ~]$ realTimeConfigQuickScan * *== GUI-enabled checks ==* *Checking if you are root... no - good* *Checking filesystem 'noatime' parameter... 3.6.11 kernel - good* *(relatime is default since 2.6.30)* *Checking CPU Governors... CPU 0: 'performance' CPU 1: 'performance' CPU 2: 'performance' CPU 3: 'performance' - good* *Checking swappiness... 10 - good* *Checking for resource-intensive background processes... none found - good* *Checking checking sysctl inotify max_user_watches... >= 524288 - good* *Checking access to the high precision event timer... readable - good* *Checking access to the real-time clock... readable - good* *Checking whether you're in the 'audio' group... yes - good* *Checking for multiple 'audio' groups... no - good* *Checking the ability to prioritize processes with chrt... yes - good* *Checking kernel support for high resolution timers... found - good* *Kernel with Real-Time Preemption... not found - not good* *Kernel without real-time capabilities found* *For more information, see http://wiki.linuxmusicians.com/doku.php?id=system_configuration#installing_a_real-time_kernel * *Checking if kernel system timer is set to 1000 hz... found - good* *Checking kernel support for tickless timer... found - good* *== Other checks ==* *Checking filesystem types... ok.* *ok.* *** Set $SOUND_CARD_IRQ to the IRQ of your soundcard to enable more checks.* * Find your sound card's IRQ by looking at '/proc/interrupts' and lspci.* *[paum at bookes ~]$ * and I dont understand two things: *Kernel with Real-Time Preemption... not found - not good* *Kernel without real-time capabilities found* *BUT* *[paum at bookes ~]$ uname -a* *Linux bookes 3.6.11-rt33-1-rt #1 SMP PREEMPT RT Sun Apr 28 12:18:40 CEST 2013 x86_64 GNU/Linux* *[paum at bookes ~]$ * and the second, how to: ** Set $SOUND_CARD_IRQ to the IRQ of your soundcard to enable more checks. Find your sound card's IRQ by looking at '/proc/interrupts' and lspci. ?? H: Known things that need to be done: -Interrupt requests of your hardware must be prioritized (rtirq script helps here) F: my system says: rtirq.service loaded active exited Realtime IRQ thread system tuning H: -Interrupt thread handlers (on RT kernel) must be prioritized F: how to ? H: -Thread priorities must be checked for JACK itself F: How to ? H: Some other things to look into are: -CPU scaling (causes Xruns here, although others say it works fine) -WiFi (must be disabled here: "iwconfig wlan0 txpower off", not just disconnected) -Graphics chip drivers (I'm using a standard xf86-video driver for my ATI HD2400XT) F: CPU scaling seems good: *Checking CPU Governors... CPU 0: 'performance' CPU 1: 'performance' CPU 2: 'performance' CPU 3: 'performance' - good* WiFi - thats pity, because I often use my phone as controller (TouchOSC) Graphics - Nvidia - I am using noveau driver H: That's a good start anyway: If I find some time, I'll write a quick post on tuning an Arch system for maximum low-lat RT performance. F: appreciated ;) F: My aim is to get as low latency as possible. H :Please specify... How low is low :D I've achieved 0.3ms at 96kHz, running a CPU stress test on each core without Xruns, so I'm pretty happy with that... since the hardware (Echo Indigo DJ PCMCIA) can't handle any lower! F: oh, great !. it could be an ideal for me. I use various acoustic instruments for live playing, so yes I am talking about less than 4ms F: but none of them was without xruns when running: pd - plugins - ardour H: PD is a bit of a xrun hog on my machine here: not bashing PD, I probably don't have it set up with ideally. But perhaps try running eg Ardour on its own for testing purposes? Or another simple app which isn't as dependent on the "patch" that's loaded..? HTH, -Harry fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Tue Jul 2 18:55:38 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Tue, 2 Jul 2013 22:55:38 +0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: Hey Diego! NON is generally a nice approach. The problem I had with it, as far as I remember, is that it does not support MIDI. Which really kills the whole deal for me, since the rest of Linux Audio apps are MIDI apps. Somewhere on the NON site there was a lengthy instruction on how to setup a bridge between MIDI and OSC, but that was not my cup of tea, that kind of tech voodoo. The power of Linux Audio - the diversity of solutions - is at the same time its weakness, as typically your setup at any given time will reliably have apps that fall out of your routine. Currently for me it is seq24 that does not work with JACK Transport. If this is going to be fixed someday, by that time some other app might have another problem. At the same time seq24 has no reliable automation (I was told it is there, I found nothing, tbh). And also seq24 has weird volume editing. As mentioned with NON, it does not support MIDI. Ardour 3 works great, but does not support DSSI, so no WhySynth or Nekobee, for instance. The list goes on. There is always a "but". And it is absolutely normal, as each app is a project in of itself. Even if it is supposed to be a tool that makes sense only with other applications, rarely apps are designed with the whole environment and/or too much time and effort is required to accommodate said environment. And so you end up with this shattered puzzle. You put one piece in - another falls out. This is why I believe a different approach would be great - basically, a modular all-in-one system, meaning modules that are designed to speak to each other. How realistic is that to make? Don't know. -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Tue Jul 2 19:14:44 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 2 Jul 2013 20:14:44 +0100 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: On Tue, Jul 2, 2013 at 6:41 PM, Louigi Verona wrote: > Just some thoughts to share Nice perspective, informative. I think it would be a great resource to get feedback like what you said from a wide range of linux-audio musicians, as a kind of "probe" to find the most significant lacking elements. Perhaps I'll be told "we know that already", but what harm in confirming our suspicions? Anybody know how to best gather up such info? (I'm not experienced in that kinda thing) Re workflow, if some type of OSC protocol could be established between a synth ("client"/"automate-able") and an automation program ("host"/"automater"), twisting the knob on the synth client could inform the host which parameter to show in the editor...? Perhaps its a bit of a hack, and it would require wide-spread implementation to become useful, but it is a *workable* solution. Perhaps not the neatest, but I think worth mentioning. -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Tue Jul 2 19:17:21 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Tue, 2 Jul 2013 23:17:21 +0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: An interesting discussion in the mailing list was at the end of 2012 when I wrote this article: http://www.louigiverona.ru/?page=projects&s=writings&t=linux&a=linux_progress http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-October/087434.html Many people say what they think. On Tue, Jul 2, 2013 at 11:14 PM, Harry van Haaren wrote: > On Tue, Jul 2, 2013 at 6:41 PM, Louigi Verona > wrote: > > Just some thoughts to share > Nice perspective, informative. I think it would be a great resource to get > feedback like what you said from a wide range of linux-audio musicians, as > a kind of "probe" to find the most significant lacking elements. Perhaps > I'll be told "we know that already", but what harm in confirming our > suspicions? > Anybody know how to best gather up such info? (I'm not experienced in that > kinda thing) > > Re workflow, if some type of OSC protocol could be established between a > synth ("client"/"automate-able") and an automation program > ("host"/"automater"), twisting the knob on the synth client could inform > the host which parameter to show in the editor...? > Perhaps its a bit of a hack, and it would require wide-spread > implementation to become useful, but it is a *workable* solution. Perhaps > not the neatest, but I think worth mentioning. > > > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From diego.simak at gmail.com Tue Jul 2 19:25:52 2013 From: diego.simak at gmail.com (Diego Simak) Date: Tue, 2 Jul 2013 16:25:52 -0300 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: 2013/7/2 Louigi Verona : > Hey Diego! > > NON is generally a nice approach. The problem I had with it, as far as I > remember, is that it does not support MIDI. Which really kills the whole > deal for me, since the rest of Linux Audio apps are MIDI apps. Somewhere on > the NON site there was a lengthy instruction on how to setup a bridge > between MIDI and OSC, but that was not my cup of tea, that kind of tech > voodoo. Yes I understand it. In fact I kept thinking in how to control hydrogen or yoshimi from NON for instance. I can not see how to do this at the moment. > > The power of Linux Audio - the diversity of solutions - is at the same time > its weakness, as typically your setup at any given time will reliably have > apps that fall out of your routine. Currently for me it is seq24 that does > not work with JACK Transport. If this is going to be fixed someday, by that > time some other app might have another problem. At the same time seq24 has > no reliable automation (I was told it is there, I found nothing, tbh). I don't want to focus in seq24 since the topic is more general and more important but I remember this tutorial from Leigh Dyer showing MIDI CC automation in seq24 http://wootangent.net/2010/11/linux-music-tutorial-seq24-part-2/ And > also seq24 has weird volume editing. > As mentioned with NON, it does not support MIDI. Ardour 3 works great, but > does not support DSSI, so no WhySynth or Nekobee, for instance. > > The list goes on. There is always a "but". And it is absolutely normal, as > each app is a project in of itself. Even if it is supposed to be a tool that > makes sense only with other applications, rarely apps are designed with the > whole environment and/or too much time and effort is required to accommodate > said environment. And so you end up with this shattered puzzle. You put one > piece in - another falls out. > > This is why I believe a different approach would be great - basically, a > modular all-in-one system, meaning modules that are designed to speak to > each other. > > How realistic is that to make? Don't know. > > > > -- > Louigi Verona > http://www.louigiverona.ru/ From harryhaaren at gmail.com Tue Jul 2 19:29:56 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 2 Jul 2013 20:29:56 +0100 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: On Tue, Jul 2, 2013 at 7:42 PM, Fero Kiraly wrote: > I say hallo for Harry ! ;) Thanks! Your email client seems to mash up things a little: at least, the email I'm replying to is quite garbled here... please send plain text emails in future, and just reply under the text, no need for H: F: etc :) Also, when pasting output from commands use a "pastebin" like http://pastebin.com/ and provide a link, it makes for easier reading. Thanks! > > H: That's a good start anyway: If I find some time, I'll write a quick post on tuning an Arch system for maximum low-lat RT performance. > F: appreciated ;) A lot of your questions will be answered in this post: I'll link it here when it is finished. > F: oh, great !. it could be an ideal for me. I use various acoustic instruments for live playing, so yes I am talking about less than 4ms I'm also using soft-synths / effects live, so indeed latency should be minimal. Understood. -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Tue Jul 2 19:41:16 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 2 Jul 2013 20:41:16 +0100 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: On Tue, Jul 2, 2013 at 8:14 PM, Harry van Haaren wrote: > Re workflow, if some type of OSC protocol could be established between a > synth ("client"/"automate-able") and an automation program > ("host"/"automater") > I've been made aware (thanks FalkTX) that this type of OSC protocol is available, although I believe some details are still gritty. I'm not one of the devs, so can't reliably comment. Links: https://github.com/original-male/non/issues/61 https://github.com/falkTX/Carla/issues/44 -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Tue Jul 2 20:29:05 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 2 Jul 2013 20:29:05 +0000 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: <20130702202905.GA26869@linuxaudio.org> On Tue, Jul 02, 2013 at 09:40:50AM -0700, Rusty Perez wrote: > Of course, most of this is in fun, I hope that by the time I make my > first recording, he'll learn some studio manners, or died, or moved > out, :-) but I hadn't thought of the possibility of a cricket taking > up residence in the studio. :-) Open the door and shout 'Extra omnes !' If your cricket is Roman Catholic he will leave. In the other case there's nothing you can do. 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 dlphillips at woh.rr.com Tue Jul 2 20:38:07 2013 From: dlphillips at woh.rr.com (Dave Phillips) Date: Tue, 02 Jul 2013 16:38:07 -0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: <51D33A2F.6090109@woh.rr.com> On 07/02/2013 02:55 PM, Louigi Verona wrote: > ... I believe a different approach would be great - basically, a > modular all-in-one system, meaning modules that are designed to speak > to each other. > > How realistic is that to make? Don't know. Well, PyDAW seems to fit the bill. Of course, everyone hates the developer so there's no enlightened discussion possible here. (It would not be amiss to suggest that he made himself less than likeable). AFAIK I'm the only one on this list who looked beyond the personality to actually use the man's software. And it is exactly what you describe, a closed-modular system of audio/MIDI recording and editing tools, with effects and instruments prepackaged and immediately available for use. (His synths are really nice, btw). Jeff has continued to develop the program and I look forward to testing his latest additions. Yes, he despises JACK and I'm pretty sure he doesn't feel all that kindly towards ALSA. So what ? Are these things sacred cows, never to be disturbed from their current positions ? And if he's offensive, maybe he takes it far more seriously than we think, and maybe just maybe he has thought through to a very different Tao of Linux audio. In another example, the OSS/Linux guys think differently about it, and they have a viable product. When I asked "What sucks about Linux audio ?" I got a lot of replies that support the contention that all is NOT well in the Linux audio world. At the same time it occurred to me that that's just how it is here, and if you (the impersonal "you", not you specifically, Louigi) want to influence the way things go - in the same way that a specific group of people determined the current path beginning in the late 1990s - then you'd best get up and start moving, because there's talking and walking. Talk's cheap. And the required walking skills are not inconsiderable. Developers of Win/Mac music software listen to users because they pay them for a product. Can't escape the economics of the thing, it matters whether you like it or not. If I want to keep my customer base then I attend to their needs, else I don't get paid, and I might as well pass the hat at Louigi Verona gigs. So we can keep talking or we can start paying. Paul Davis has developed Ardour thanks to contributions. I highly doubt it would have reached its current level without the money coming in. Rui has a good day gig, and he's taken his own sweet time to develop QTractor. (Is it out of alpha yet, Rui ? :) Not to speak for them, but I suggest that both of these gentlemen react positively to cash injections. OTOH I think things are likely to be just swell for people like myself. My use cases are not so machine-centric, and I don't need the standard tools for EDM. Selfishly speaking, things are looking great from my POV. But if I have a concern for the continued development of Linux audio software then I must consider the needs of my colleagues who do require those other tools. Their immediate concerns may not matter to me, but their longer-term involvement is critically determined by the availability of their needed tools. Too much stasis, and sooner or later even Louigi is going back to Windows. And I'd rather not lose colleagues with his capabilities. Rant over. Peace out. dp From arnold at arnoldarts.de Tue Jul 2 20:48:43 2013 From: arnold at arnoldarts.de (Arnold Krille) Date: Tue, 2 Jul 2013 22:48:43 +0200 Subject: [LAU] command line cricket removal tool? In-Reply-To: <20130702202905.GA26869@linuxaudio.org> References: <20130702202905.GA26869@linuxaudio.org> Message-ID: <20130702224843.3c584445@xingu.arnoldarts.de> On Tue, 2 Jul 2013 20:29:05 +0000 Fons Adriaensen wrote: > On Tue, Jul 02, 2013 at 09:40:50AM -0700, Rusty Perez wrote: > > > Of course, most of this is in fun, I hope that by the time I make my > > first recording, he'll learn some studio manners, or died, or moved > > out, :-) but I hadn't thought of the possibility of a cricket taking > > up residence in the studio. :-) > > Open the door and shout 'Extra omnes !' > > If your cricket is Roman Catholic he will leave. In the other > case there's nothing you can do. Announcing "free beer outside" should work for everything else? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: not available URL: From allcoms at gmail.com Tue Jul 2 20:53:52 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Tue, 2 Jul 2013 21:53:52 +0100 Subject: [LAU] JACK on Haiku Message-ID: A member of the Haiku community has made an early prototype port of JACK2 to the fledgling, media-centric FLOSS OS https://www.haiku-os.org/blog/barrett/2013-07-02_jack2_personal_analysis_part_1 TL;DR Haiku currently lacks RT scheduling and shm.h and this proto port lacks a backend to produce any output. The author proposes it may be possible to create a bridge similar to JackRouter on OSX to pipe JACK to Haiku's media_kit. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Tue Jul 2 21:03:14 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 2 Jul 2013 21:03:14 +0000 Subject: [LAU] command line cricket removal tool? In-Reply-To: <20130702224843.3c584445@xingu.arnoldarts.de> References: <20130702202905.GA26869@linuxaudio.org> <20130702224843.3c584445@xingu.arnoldarts.de> Message-ID: <20130702210314.GA6346@linuxaudio.org> On Tue, Jul 02, 2013 at 10:48:43PM +0200, Arnold Krille wrote: > > Open the door and shout 'Extra omnes !' > > > > If your cricket is Roman Catholic he will leave. In the other > > case there's nothing you can do. > > Announcing "free beer outside" should work for everything else? Should work for snails. Don't know about crickets... :-) -- 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 tito.01beta at gmail.com Tue Jul 2 21:16:43 2013 From: tito.01beta at gmail.com (Tito Latini) Date: Tue, 2 Jul 2013 23:16:43 +0200 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: Message-ID: <20130702211643.GA1832@rhk.homenet.telecomitalia.it> On Tue, Jul 02, 2013 at 09:40:50AM -0700, Rusty Perez wrote: > Hi folks, > Does any one know of a simple tool for removing crickets from my home > studio? :-) There are four types of cricket song: The calling song attracts females and repels other males, and is fairly loud. The courting song is used when a female cricket is near, and is a very quiet song. An aggressive song is triggered by chemoreceptors on the antennae that detect the near presence of another male cricket and a copulatory song is produced for a brief period after a successful mating. -- Wikipedia Algorithm to remove the cricket noise (pseudo-code): - put a cricket-female near the door of the studio - cricketto starts the calling song - when you hear the courting song, push the crickets out and close the door of the studio - cricket noise is removed if the copulatory song is out of the studio HTH tito From rncbc at rncbc.org Tue Jul 2 21:43:37 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Tue, 02 Jul 2013 22:43:37 +0100 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: <51D33A2F.6090109@woh.rr.com> References: <51D33A2F.6090109@woh.rr.com> Message-ID: <51D34989.7000808@rncbc.org> On 07/02/2013 09:38 PM, Dave Phillips wrote: > [...] Rui has a good day gig, and he's taken his own sweet time to > develop QTractor. (Is it out of alpha yet, Rui ? :) Not to speak for > them, but I suggest that both of these gentlemen react positively to > cash injections. thanks Dave (hal9k talk:) everybody reacts positively to positive incentives (just an economist lapalissade, but yeah;) whether it's still alpha? well, yes. on my call and personal stance there's only two phases to software life cycle: alpha and omega. anything in between is just marketing gibberish alpha is when some software "entity" is actually running to its users benefit. whether saying it does it well or not--in my experience it often does not--it's strictly a user's prerogative, not a developer's one. while in this so-called alpha phase, all software lingers in a more or less broken state. yes, broken. nothing is perfect and never will, until... then, paraphrasing Exup?ry's, only when the last user leaves the building then you know it reached ultimate perfection and that my friend, it's when you know it reached omega :) cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From fons at linuxaudio.org Tue Jul 2 21:49:25 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 2 Jul 2013 21:49:25 +0000 Subject: [LAU] Updates Message-ID: <20130702214925.GB6346@linuxaudio.org> Hello all, Some maintenance updates are available on * libclxclient 3.9.0: bugfixes * aeolus, aliki, jaaa, japa: all now use zita-alsa-pcmi instead of clalsadrv. * The aliki package now includes the manual. That means that clalsadrv is now deprecated. It will remain available for a few months and then disappear forever. Note to AMS devs: zita-alsa-pcmi is a near drop-in replacement for clalsadrv-2.0.0: * Change the library name in the build files * s/#include/#include/ * s/Alsa_driver/Alsa_pcmi/ * s/->stat()/->state()/ Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From harryhaaren at gmail.com Tue Jul 2 22:19:41 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 2 Jul 2013 23:19:41 +0100 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: On Tue, Jul 2, 2013 at 8:29 PM, Harry van Haaren wrote: > Also, when pasting output from commands use a "pastebin" like http://pastebin.com/ and provide a link, it makes for easier reading. I'd like to take this back: its better to have the information neatly available on the mailing list. > If I find some time, I'll write a quick post on tuning an Arch system for maximum low-lat RT performance. Link to article, (note it is geared towards Arch linux, but the same concepts apply across distro flavours) http://openavproductions.com/real-time-latency-tuning/ Thanks for providing the incentive to write this article, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Tue Jul 2 22:26:47 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Wed, 3 Jul 2013 00:26:47 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: Thank you. great. i am gonna to read. all the best 2013/7/3 Harry van Haaren > Thanks for providing the incentive to write this article -- Fero Kiraly www.ferokiraly.com www.cluster-ensemble.com T -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien at mail.upb.de Tue Jul 2 22:56:42 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 3 Jul 2013 00:56:42 +0200 (CEST) Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: Message-ID: Hello Louigi! Your podcast was very intersting. It seems, that in the GUI there are the same difficulties as in the commandline, only on a higher level. We have the same problem of combining different apps in a simple setup. Nama is on a good road there to do all the magic in the background. But somehow it makes me feel a little bett3er about the state of CLI problematics. Thanks for your podcast. However relevant or true it might be for the current state of GUI applications. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From falktx at gmail.com Wed Jul 3 00:36:33 2013 From: falktx at gmail.com (Filipe Coelho) Date: Wed, 03 Jul 2013 01:36:33 +0100 Subject: [LAU] Carla Plugin Host 1.0.0 released! Message-ID: <51D37211.4020508@gmail.com> Hi there everyone, just passing the info ;) I'm not much of a mailing list guy, so if possible check the LM forum link in this message for replies, thanks! -------------------------------------------------------------------- After more than 2 years in development, here it is - the first stable release of Carla is now available! [Screenshot: http://kxstudio.sourceforge.net/screenshots/carla.png] Carla Plugin Host, or just Carla, is an audio plugin host, with support for many audio drivers and plugin formats. It has some nice features like automation of parameters via MIDI CC and full OSC control. Carla currently supports LADSPA (including LRDF), DSSI, LV2, and VST plugin formats, plus GIG, SF2 and SFZ file support. It uses JACK as the default and preferred audio driver but also supports native drivers like ALSA, DirectSound or CoreAudio. We currently release source code plus Windows and Linux binaries (focus goes into the Linux version). Downloads available at the usual place, http://kxstudio.sourceforge.net/Downloads#Binaries. For more information about Carla check its own application page here. We'll be updating that page in the next few days, with some tutorials and workflow videos as well, so stay tuned! You can follow the release discussion over at LinuxMusicians forums, https://linuxmusicians.com/viewtopic.php?f=24&t=11249. If you find this application useful and/or want to help further development, please consider a donation: http://kxstudio.sourceforge.net/Donations From paul at linuxaudiosystems.com Wed Jul 3 07:03:11 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Wed, 3 Jul 2013 03:03:11 -0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: <51D33A2F.6090109@woh.rr.com> References: <51D33A2F.6090109@woh.rr.com> Message-ID: On Tue, Jul 2, 2013 at 4:38 PM, Dave Phillips wrote: > > Yes, he despises JACK and I'm pretty sure he doesn't feel all that kindly > towards ALSA. So what ? Are these things sacred cows, never to be disturbed > from their current positions ? And if he's offensive, maybe he takes it far > more seriously than we think, and maybe just maybe he has thought through > to a very different Tao of Linux audio. In another example, the OSS/Linux > guys think differently about it, and they have a viable product. > far too busy to get involved in one of these threads, but i thought i should repeat something i've said many times before, in various ways. you will NOT find the developers of audio apps on OS X or Windows wasting much time cursing out CoreAudio or WDM or WaveRT, certainly not in public. These features of their landscape are fixed and not subject to change through any efforts they can make (other than bug reports and so forth). and yes, those systems are far from perfect, but they are THE systems you get to use on their respective platforms, whether you like it or not. linux has always had the problem that nobody really wields a big enough stick - it is always possible to image and more importantly to actually implement an entirely different approach to a particular aspect of the kernel and userspace subsystems. as a result of this, we get benefits (some really innovative stuff happens first on linux) and we get disadvantages (because expertise and hours are spent "fixing" systems that you'd just have to deal with on other platforms). one final small note for louigi regarding things like "why ardour doesn't support DSSI". logic on OS X doesn't support VST. the proprietary world is not so diffferent :) of course, there is one important difference: nobody but apple could ever add VST support to logic. anyone could add DSSI support to ardour, if they had the knowledge and desire. i'm just not that person. -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Wed Jul 3 07:04:18 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Wed, 3 Jul 2013 11:04:18 +0400 Subject: [LAU] Carla Plugin Host 1.0.0 released! In-Reply-To: <51D37211.4020508@gmail.com> References: <51D37211.4020508@gmail.com> Message-ID: This is great news indeed! I've been using Carla for the past few months and I can say it is a great plugin host and you are able to save the project and reopen it later. The GUI is very good, comfortable to use, easy to assign MIDI control. All in all, Carla is there as a permanent element of my workflow. Thanks Filipe! -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Wed Jul 3 07:12:05 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Wed, 3 Jul 2013 11:12:05 +0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: <51D33A2F.6090109@woh.rr.com> Message-ID: Sure, Paul, my comment on Ardour was not meant to say that you should support DSSI. It simply pointed out the facts. I feel great respect towards developers of Linux Audio and I totally understand that each developer has his reasons for doing what he does. So I was not even asking "why Ardour doesn't support DSSI", I just said that it doesn't. Obviously, you have good reasons for what you are doing. So, I hope my tone did not turn out as demanding as it was not meant to be at all. You made an interesting point about Windows that people just work with what they have when on Linux devs spend lots of time fixing things. I wonder how big is this effect. Do many people engage in this kind of activity? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Wed Jul 3 07:37:54 2013 From: gnome at hawaii.rr.com (david) Date: Tue, 02 Jul 2013 21:37:54 -1000 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: <51D33A2F.6090109@woh.rr.com> Message-ID: <51D3D4D2.706@hawaii.rr.com> On 07/02/2013 09:12 PM, Louigi Verona wrote: > Sure, Paul, my comment on Ardour was not meant to say that you should > support DSSI. It simply pointed out the facts. > > I feel great respect towards developers of Linux Audio and I totally > understand that each developer has his reasons for doing what he does. > So I was not even asking "why Ardour doesn't support DSSI", I just said > that it doesn't. Obviously, you have good reasons for what you are doing. > > So, I hope my tone did not turn out as demanding as it was not meant to > be at all. > > You made an interesting point about Windows that people just work with > what they have when on Linux devs spend lots of time fixing things. I > wonder how big is this effect. Do many people engage in this kind of > activity? Well, ages ago, I was one of the first users of CorelDRAW. Fed them many bug reports and UI feedback. Won't claim to make a big difference, but I did isolate a serious bug sufficiently that they could finally fix it. They rewarded me with a free upgrade to v2. So it's pretty cool when a developer rewards (me) by putting out a better version! I know a lot of Windows users just put up with problems, maybe grumble a little bit, but figure it's just the way computers are. I work with a number of them. My biggest complaint about Microsoft is that they have convinced users to accept computer and application crashes as "normal". Reminds me, sometime I must start learning Ardour. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From jeremy at autostatic.com Wed Jul 3 07:43:20 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 03 Jul 2013 09:43:20 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: <51D3D618.2@autostatic.com> On 07/03/2013 12:19 AM, Harry van Haaren wrote: > On Tue, Jul 2, 2013 at 8:29 PM, Harry van Haaren > wrote: >> Also, when pasting output from commands use a "pastebin" like > http://pastebin.com/ and provide a link, it makes for easier reading. > I'd like to take this back: its better to have the information neatly > available on the mailing list. > >> If I find some time, I'll write a quick post on tuning an Arch system for > maximum low-lat RT performance. > Link to article, (note it is geared towards Arch linux, but the same > concepts apply across distro flavours) > http://openavproductions.com/real-time-latency-tuning/ > > Thanks for providing the incentive to write this article, -Harry > Nice article Harry! Some remarks though: * Is /dev/rtc the kernel timer? I thought it was the device node for access to the real-time clock. I don't even think it's necessary to prioritize it anymore because most applications use the kernel timer: http://wiki.linuxaudio.org/wiki/system_configuration#hardware_timers I'm far from an expert on timers so I could be wrong, I still find this a bit of esoteric stuff. * Using setpci in this context is deprecated unless you're still using am machine with a PCI bus which I highly doubt: http://wiki.linuxaudio.org/wiki/system_configuration#pci_bus_latency Regards, Jeremy From jeremy at autostatic.com Wed Jul 3 08:03:51 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 03 Jul 2013 10:03:51 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: <51D3DAE7.9020402@autostatic.com> On 07/02/2013 08:42 PM, Fero Kiraly wrote: > I say hallo for Harry ! ;) > > F: I have some troubles - read xruns - with this setup: > H: There's a lot more to removing Xruns from a system than just the kernel > and JACK settings (although they are very important :) > F: actually I know realTimeConfigQuickScan, so my output looks: > > *[paum at bookes ~]$ realTimeConfigQuickScan * > *== GUI-enabled checks ==* > *Checking if you are root... no - good* > *Checking filesystem 'noatime' parameter... 3.6.11 kernel - good* > *(relatime is default since 2.6.30)* > *Checking CPU Governors... CPU 0: 'performance' CPU 1: 'performance' CPU 2: > 'performance' CPU 3: 'performance' - good* > *Checking swappiness... 10 - good* > *Checking for resource-intensive background processes... none found - good* > *Checking checking sysctl inotify max_user_watches... >= 524288 - good* > *Checking access to the high precision event timer... readable - good* > *Checking access to the real-time clock... readable - good* > *Checking whether you're in the 'audio' group... yes - good* > *Checking for multiple 'audio' groups... no - good* > *Checking the ability to prioritize processes with chrt... yes - good* > *Checking kernel support for high resolution timers... found - good* > *Kernel with Real-Time Preemption... not found - not good* > *Kernel without real-time capabilities found* > *For more information, see > http://wiki.linuxmusicians.com/doku.php?id=system_configuration#installing_a_real-time_kernel > * > *Checking if kernel system timer is set to 1000 hz... found - good* > *Checking kernel support for tickless timer... found - good* > *== Other checks ==* > *Checking filesystem types... ok.* > *ok.* > *** Set $SOUND_CARD_IRQ to the IRQ of your soundcard to enable more checks.* > * Find your sound card's IRQ by looking at '/proc/interrupts' and lspci.* > *[paum at bookes ~]$ * > > > and I dont understand two things: > > *Kernel with Real-Time Preemption... not found - not good* > *Kernel without real-time capabilities found* > *BUT* > *[paum at bookes ~]$ uname -a* > *Linux bookes 3.6.11-rt33-1-rt #1 SMP PREEMPT RT Sun Apr 28 12:18:40 CEST > 2013 x86_64 GNU/Linux* > *[paum at bookes ~]$ * > Hello Fero, Where did you get the realTimeConfigQuickScan script? The author updated it a month ago so it also checks for the right CONFIG option for 3.x kernels. And did you compile the kernel yourself? > and the second, how to: > ** Set $SOUND_CARD_IRQ to the IRQ of your soundcard to enable more checks. > Find your sound card's IRQ by looking at '/proc/interrupts' and lspci. > ?? > This only applies for onboard or PCI audio interfaces, so not for USB and FireWire interfaces. Regards, Jeremy From jeremy at autostatic.com Wed Jul 3 08:09:08 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 03 Jul 2013 10:09:08 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: <51D31B94.90800@linuxaudio.org> References: <51D31B94.90800@linuxaudio.org> Message-ID: <51D3DC24.9050906@autostatic.com> On 07/02/2013 08:27 PM, Robin Gareus wrote: > On 07/02/2013 07:49 PM, Harry van Haaren wrote: >> On Tue, Jul 2, 2013 at 5:53 PM, Fero Kiraly wrote: >>> I have some troubles - read xruns - with this setup: >> There's a lot more to removing Xruns from a system than just the kernel and >> JACK settings (although they are very important :) > > * linux 3.2.35-2 PREEMPT RT > (debian's RT kernel recompiled with VSL1818 clock selector fix added) > * jackd 1.9.10 (git) > * Rui's rtirq script > * jackd with -p64 -n2 (actually jackdbus) Hi Robin, So with and USB2.0 audio interface like the 181VSL using -n2 yields a stable setup too, no need anymore to use -n3? > * presonus 1818VSL > > -> very reliable no x-runs > >> Known things that need to be done: >> -Interrupt requests of your hardware must be prioritized (rtirq script >> helps here) >> -Interrupt thread handlers (on RT kernel) must be prioritized >> -Thread priorities must be checked for JACK itself > > +1, +1, +1 > >> Some other things to look into are: >> -CPU scaling (causes Xruns here, although others say it works fine) > > no problem here w/freq scaling. I have disabled C1E halt states and EIST > in the Bios, though. > I have yet to encounter a BIOS that has these options. Regards, Jeremy From robin at gareus.org Wed Jul 3 08:31:55 2013 From: robin at gareus.org (Robin Gareus) Date: Wed, 03 Jul 2013 10:31:55 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: <51D3DC24.9050906@autostatic.com> References: <51D31B94.90800@linuxaudio.org> <51D3DC24.9050906@autostatic.com> Message-ID: <51D3E17B.5070600@gareus.org> On 07/03/2013 10:09 AM, Jeremy Jongepier wrote: > On 07/02/2013 08:27 PM, Robin Gareus wrote: >> On 07/02/2013 07:49 PM, Harry van Haaren wrote: >>> On Tue, Jul 2, 2013 at 5:53 PM, Fero Kiraly >>> wrote: >>>> I have some troubles - read xruns - with this setup: >>> There's a lot more to removing Xruns from a system than just >>> the kernel and JACK settings (although they are very important >>> :) >> >> * linux 3.2.35-2 PREEMPT RT (debian's RT kernel recompiled with >> VSL1818 clock selector fix added) * jackd 1.9.10 (git) * Rui's >> rtirq script * jackd with -p64 -n2 (actually jackdbus) > > Hi Robin, > > So with and USB2.0 audio interface like the 181VSL using -n2 yields > a stable setup too, no need anymore to use -n3? > I don't know if that's USB-2.0 or something else.. - The same goes for the statement that "USB devices should perform better if the sample rate is divisible by the period size". I cannot find any evidence that supports that statement. As with all complex systems, I trust measurements more than theory: http://robin.linuxaudio.org/tmp/vsl1818latency.png (x-axis are permutations of jackd's -p, -n, --sync parameters, no x-runs with either configuration but it's just jackd + jack_delay, no load). [Surprisingly]? the latency increments are not quantized to milliseconds (as the USB protocol implementation would suggest it should). From mista.tapas at gmx.net Wed Jul 3 09:03:59 2013 From: mista.tapas at gmx.net (Florian Paul Schmidt) Date: Wed, 03 Jul 2013 11:03:59 +0200 Subject: [LAU] Carla Plugin Host 1.0.0 released! In-Reply-To: References: <51D37211.4020508@gmail.com> Message-ID: <51D3E8FF.2080301@gmx.net> On 03.07.2013 09:04, Louigi Verona wrote: > This is great news indeed! > > I've been using Carla for the past few months and I can say it is a > great plugin host and you are able to save the project and reopen it > later. The GUI is very good, comfortable to use, easy to assign MIDI > control. > > All in all, Carla is there as a permanent element of my workflow. > > Thanks Filipe! > I second this opinion. It's a great work! Flo > > > > -- > Louigi Verona > http://www.louigiverona.ru/ > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Wed Jul 3 09:49:52 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Wed, 3 Jul 2013 11:49:52 +0200 Subject: [LAU] linux-rt & presonus 1818VSL Message-ID: Hi Jeremy I gt get realtimeconfigQuickscan from AUR https://aur.archlinux.org/packages/realtimeconfigquickscan/ I just updated to new version and what I have 'not good' is: ... Checking kernel support for tickless timer... not found - not good Try enabling tickless timer support (CONFIG_NO_HZ) ... kernel I have compiled by myself, but I did no config changes. should I ? fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Wed Jul 3 10:05:18 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 03 Jul 2013 12:05:18 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: <51D3F75E.90303@autostatic.com> On 07/03/2013 11:49 AM, Fero Kiraly wrote: > Hi Jeremy > > I gt get realtimeconfigQuickscan from AUR > https://aur.archlinux.org/packages/realtimeconfigquickscan/ > > I just updated to new version and what I have 'not good' is: > ... > Checking kernel support for tickless timer... not found - not good > Try enabling tickless timer support (CONFIG_NO_HZ) > ... > If you have CONFIG_HZ=1000 then it should be ok assuming this is about a desktop/notebook. Afaik CONFIG_NO_HZ is a power-consumption related setting, setting this option makes the kernel timer only wake up when requested while setting it at 1000Hz activates it all the time. Not sure if I worded that right, I'm not a kernel expert. With respect to performance implications I can't say anything sensible. I'm using kernels with CONFIG_HZ=1000 and CONFIG_NO_HZ=y side by side and haven't noticed any real differences so far. > kernel I have compiled by myself, but I did no config changes. > should I ? > Afaik that's not necessary. Jeremy > > fero > From fero.kiraly at gmail.com Wed Jul 3 10:31:01 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Wed, 3 Jul 2013 12:31:01 +0200 Subject: [LAU] linux-rt & presonus 1818VSL Message-ID: Thanks Jeremy, yes, my timer should be ok: ... Checking if kernel system timer is set to 1000 hz... found - good ... now I have configured rtirq due to harry's article and seems good 48000/64/2 I am gonna to testing.. great forum, great people, great software. thank you all. fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Wed Jul 3 12:27:38 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Wed, 3 Jul 2013 13:27:38 +0100 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: <51D3D618.2@autostatic.com> References: <51D3D618.2@autostatic.com> Message-ID: On Wed, Jul 3, 2013 at 8:43 AM, Jeremy Jongepier wrote: > On 07/03/2013 12:19 AM, Harry van Haaren wrote: > Nice article Harry! Some remarks though: > * Is /dev/rtc the kernel timer? I thought it was the device node for > access to the real-time clock. I don't even think it's necessary to > prioritize it anymore because most applications use the kernel timer: > http://wiki.linuxaudio.org/wiki/system_configuration#hardware_timers > I'm far from an expert on timers so I could be wrong, I still find this > a bit of esoteric stuff. > Same here: I wrote that because its in my setup, and AFAIK its the kernel timer... Just found this article: https://www.kernel.org/doc/Documentation/rtc.txt It suggests that /dev/rtc is the "old" RTC, and /dev/rtc0 /dev/rtc1 .. rtcX is the new system. I have got /dev/rtc0 present on my system. I'll investigate into whether JACK / the ALSA backend use /dev/rtc, or if it really isn't used anymore. > * Using setpci in this context is deprecated unless you're still using > am machine with a PCI bus which I highly doubt: > http://wiki.linuxaudio.org/wiki/system_configuration#pci_bus_latency > I was advised on IRC that the PCI bandwidth reset would help, and it didn't negatively influence my machine: hence I added it. lspci -v -t shows that my firewire, cardbus ("CardBus bridge: Texas Instruments PCIxx12 Cardbus Controller") SD card and multimedia card readers are on the same branch, connected to the root node. I suppose rm-modding those kernel modules would ensure better performance (not using them probably suffices). The graphics chip is on its own node, also connected directly to the root. I don't think it'll get in the way.. Setting the latencies won't hurt, I'll leave it there for now but add a note that on modern hardware its not necessary. Thanks for the feedback! -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From willgodfrey at musically.me.uk Wed Jul 3 13:26:44 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Wed, 3 Jul 2013 14:26:44 +0100 Subject: [LAU] Slightly O/T was: linux-rt & presonus 1818VSL In-Reply-To: References: <51D3D618.2@autostatic.com> Message-ID: <20130703142644.2330f647@debian> On Wed, 3 Jul 2013 13:27:38 +0100 Harry van Haaren wrote: > On Wed, Jul 3, 2013 at 8:43 AM, Jeremy Jongepier wrote: > > > On 07/03/2013 12:19 AM, Harry van Haaren wrote: > > Nice article Harry! Some remarks though: > > * Is /dev/rtc the kernel timer? I thought it was the device node for > > access to the real-time clock. I don't even think it's necessary to > > prioritize it anymore because most applications use the kernel timer: > > http://wiki.linuxaudio.org/wiki/system_configuration#hardware_timers > > I'm far from an expert on timers so I could be wrong, I still find this > > a bit of esoteric stuff. > > > Same here: I wrote that because its in my setup, and AFAIK its the kernel > timer... Just found this article: > https://www.kernel.org/doc/Documentation/rtc.txt It suggests that > /dev/rtc is the "old" RTC, and /dev/rtc0 /dev/rtc1 .. rtcX is the new > system. I have got /dev/rtc0 present on my system. I'll investigate into > whether JACK / the ALSA backend use /dev/rtc, or if it really isn't used > anymore. > > > > * Using setpci in this context is deprecated unless you're still using > > am machine with a PCI bus which I highly doubt: > > http://wiki.linuxaudio.org/wiki/system_configuration#pci_bus_latency > > > I was advised on IRC that the PCI bandwidth reset would help, and it didn't > negatively influence my machine: hence I added it. > > lspci -v -t shows that my firewire, cardbus ("CardBus bridge: Texas > Instruments PCIxx12 Cardbus Controller") SD card and multimedia card > readers are on the same branch, connected to the root node. I suppose > rm-modding those kernel modules would ensure better performance (not using > them probably suffices). > > The graphics chip is on its own node, also connected directly to the root. > I don't think it'll get in the way.. Setting the latencies won't hurt, I'll > leave it there for now but add a note that on modern hardware its not > necessary. > > Thanks for the feedback! -Harry Anyone got an idiot's guide for compiling a suitable kernel for debian squeeze? I don't really have any problems but feel sure I could get more out of this beast (a quad core AMD), and might get slightly better timing. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Wed Jul 3 13:27:54 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Wed, 3 Jul 2013 14:27:54 +0100 Subject: [LAU] Slightly O/T was: linux-rt & presonus 1818VSL In-Reply-To: <20130703142644.2330f647@debian> References: <51D3D618.2@autostatic.com> <20130703142644.2330f647@debian> Message-ID: <20130703142754.04a6d90d@debian> On Wed, 3 Jul 2013 14:26:44 +0100 Will Godfrey wrote: > On Wed, 3 Jul 2013 13:27:38 +0100 > Harry van Haaren wrote: > > > On Wed, Jul 3, 2013 at 8:43 AM, Jeremy Jongepier wrote: > > > > > On 07/03/2013 12:19 AM, Harry van Haaren wrote: > > > Nice article Harry! Some remarks though: > > > * Is /dev/rtc the kernel timer? I thought it was the device node for > > > access to the real-time clock. I don't even think it's necessary to > > > prioritize it anymore because most applications use the kernel timer: > > > http://wiki.linuxaudio.org/wiki/system_configuration#hardware_timers > > > I'm far from an expert on timers so I could be wrong, I still find this > > > a bit of esoteric stuff. > > > > > Same here: I wrote that because its in my setup, and AFAIK its the kernel > > timer... Just found this article: > > https://www.kernel.org/doc/Documentation/rtc.txt It suggests that > > /dev/rtc is the "old" RTC, and /dev/rtc0 /dev/rtc1 .. rtcX is the new > > system. I have got /dev/rtc0 present on my system. I'll investigate into > > whether JACK / the ALSA backend use /dev/rtc, or if it really isn't used > > anymore. > > > > > > > * Using setpci in this context is deprecated unless you're still using > > > am machine with a PCI bus which I highly doubt: > > > http://wiki.linuxaudio.org/wiki/system_configuration#pci_bus_latency > > > > > I was advised on IRC that the PCI bandwidth reset would help, and it didn't > > negatively influence my machine: hence I added it. > > > > lspci -v -t shows that my firewire, cardbus ("CardBus bridge: Texas > > Instruments PCIxx12 Cardbus Controller") SD card and multimedia card > > readers are on the same branch, connected to the root node. I suppose > > rm-modding those kernel modules would ensure better performance (not using > > them probably suffices). > > > > The graphics chip is on its own node, also connected directly to the root. > > I don't think it'll get in the way.. Setting the latencies won't hurt, I'll > > leave it there for now but add a note that on modern hardware its not > > necessary. > > > > Thanks for the feedback! -Harry > > > Anyone got an idiot's guide for compiling a suitable kernel for debian squeeze? > I don't really have any problems but feel sure I could get more out of this > beast (a quad core AMD), and might get slightly better timing. Oops! Not Squeeze, but Wheezy :o -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From jeremy at autostatic.com Wed Jul 3 13:35:23 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 03 Jul 2013 15:35:23 +0200 Subject: [LAU] Slightly O/T was: linux-rt & presonus 1818VSL In-Reply-To: <20130703142754.04a6d90d@debian> References: <51D3D618.2@autostatic.com> <20130703142644.2330f647@debian> <20130703142754.04a6d90d@debian> Message-ID: <51D4289B.80406@autostatic.com> On 07/03/2013 03:27 PM, Will Godfrey wrote: >> > >> > Anyone got an idiot's guide for compiling a suitable kernel for debian squeeze? >> > I don't really have any problems but feel sure I could get more out of this >> > beast (a quad core AMD), and might get slightly better timing. > > Oops! > Not Squeeze, but Wheezy :o Hi Will, http://wiki.linuxaudio.org/wiki/system_configuration#build_your_own_real-time_kernel_on_debian_wheezy_or_later Regards, Jeremy From gabbe.nord at gmail.com Wed Jul 3 15:05:14 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Wed, 3 Jul 2013 17:05:14 +0200 Subject: [LAU] My second album is done! Message-ID: Hello everyone! I'm very pleased to announce that my second album is finished and online! :D It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a bit more electronic than last one. I prepared a page for this at my new website, http://zthmusic.se/Ordinary_Day_Montage , but my host is a little shaky, so it might not always work. So, if that link doesn't work, you can find the album at: Bandcamp http://zthmusic.bandcamp.com/album/ordinary-day-montage Soundcloud https://soundcloud.com/zthmusic/sets/ordinary-day-montage FLAC/OGG/MP3-formats for download at Piratebay: http://thepiratebay.sx/user/zthmusic/ I'm very excited to be finished and to have completed this. Everything was, as always, 100% made with Linux and Linux software. It's also licensed CC-BY-SA. I wrote a bit about the album and what I've used technically too at http://zthmusic.se/Ordinary_Day_Montage , if anyone wants to check that out! Anyway, thank you for taking your time to listen! I greatly appreciate it! -------------- next part -------------- An HTML attachment was scrubbed... URL: From egor.sanin at gmail.com Wed Jul 3 16:12:07 2013 From: egor.sanin at gmail.com (Egor Sanin) Date: Wed, 3 Jul 2013 12:12:07 -0400 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Hi Gabbe! On 7/3/13, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and online! > :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a > bit more electronic than last one. This is absolutely great! I'm tapping my foot as I type this email, wonderful, easy summer beats. Thanks for sharing your great work. From rustys.lists at gmail.com Wed Jul 3 17:04:51 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Wed, 3 Jul 2013 10:04:51 -0700 Subject: [LAU] command line cricket removal tool? In-Reply-To: <20130702211643.GA1832@rhk.homenet.telecomitalia.it> References: <20130702211643.GA1832@rhk.homenet.telecomitalia.it> Message-ID: Wow, I guess I missed that sidebar in SoundonSound about cricket breeding practices for a quieter studio. Who knew that when they talked about "sex, drugs, and rockNroll" they were talking about crickets too? :-) Rusty On 7/2/13, Tito Latini wrote: > On Tue, Jul 02, 2013 at 09:40:50AM -0700, Rusty Perez wrote: >> Hi folks, >> Does any one know of a simple tool for removing crickets from my home >> studio? :-) > > There are four types of cricket song: The calling song attracts > females and repels other males, and is fairly loud. The courting > song is used when a female cricket is near, and is a very quiet > song. An aggressive song is triggered by chemoreceptors on the > antennae that detect the near presence of another male cricket > and a copulatory song is produced for a brief period after a > successful mating. > -- Wikipedia > > Algorithm to remove the cricket noise (pseudo-code): > > - put a cricket-female near the door of the studio > > - cricketto starts the calling song > > - when you hear the courting song, push the crickets out and > close the door of the studio > > - cricket noise is removed if the copulatory song is out of the studio > > HTH > > tito > From zotz at 100jamz.com Wed Jul 3 17:11:23 2013 From: zotz at 100jamz.com (drew Roberts) Date: Wed, 3 Jul 2013 13:11:23 -0400 Subject: [LAU] command line cricket removal tool? In-Reply-To: References: <20130702211643.GA1832@rhk.homenet.telecomitalia.it> Message-ID: <201307031311.23265.zotz@100jamz.com> On Wednesday 03 July 2013 13:04:51 Rusty Perez wrote: > Wow, I guess I missed that sidebar in SoundonSound about cricket > breeding practices for a quieter studio. > > Who knew that when they talked about "sex, drugs, and rockNroll" they > were talking about crickets too? :-) It is a shrouded homage to Buddy Holly and his band... > > Rusty drew > > On 7/2/13, Tito Latini wrote: > > On Tue, Jul 02, 2013 at 09:40:50AM -0700, Rusty Perez wrote: > >> Hi folks, > >> Does any one know of a simple tool for removing crickets from my home > >> studio? :-) > > > > There are four types of cricket song: The calling song attracts > > females and repels other males, and is fairly loud. The courting > > song is used when a female cricket is near, and is a very quiet > > song. An aggressive song is triggered by chemoreceptors on the > > antennae that detect the near presence of another male cricket > > and a copulatory song is produced for a brief period after a > > successful mating. > > -- Wikipedia > > > > Algorithm to remove the cricket noise (pseudo-code): > > > > - put a cricket-female near the door of the studio > > > > - cricketto starts the calling song > > > > - when you hear the courting song, push the crickets out and > > close the door of the studio > > > > - cricket noise is removed if the copulatory song is out of the studio > > > > HTH > > > > tito > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From paul at linuxaudiosystems.com Wed Jul 3 17:51:38 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Wed, 3 Jul 2013 13:51:38 -0400 Subject: [LAU] Linux Audio podcast, episode001 In-Reply-To: References: <51D33A2F.6090109@woh.rr.com> Message-ID: Louigi - I didn't take it any other way, really. I just thought that it was funny and interesting that the entirely legitimate point about how one (open source) DAW doesn't support one (open source) plugin API is mirrored by at least one almost identical example in the proprietary non-linux world. On Wed, Jul 3, 2013 at 3:12 AM, Louigi Verona wrote: > Sure, Paul, my comment on Ardour was not meant to say that you should > support DSSI. It simply pointed out the facts. > > I feel great respect towards developers of Linux Audio and I totally > understand that each developer has his reasons for doing what he does. So I > was not even asking "why Ardour doesn't support DSSI", I just said that it > doesn't. Obviously, you have good reasons for what you are doing. > > So, I hope my tone did not turn out as demanding as it was not meant to be > at all. > > > You made an interesting point about Windows that people just work with > what they have when on Linux devs spend lots of time fixing things. I > wonder how big is this effect. Do many people engage in this kind of > activity? > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gabbe.nord at gmail.com Wed Jul 3 20:19:47 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Wed, 3 Jul 2013 22:19:47 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Thank you very much for the comment, Egor! I'm very glad you enjoyed it! It seems like the torrents for some reason is out of function. But it's still possible to download through both Soundcloud and Bandcamp. Cheers, Gabriel/zth On Wed, Jul 3, 2013 at 6:12 PM, Egor Sanin wrote: > Hi Gabbe! > > On 7/3/13, Gabbe Nord wrote: > > Hello everyone! > > > > I'm very pleased to announce that my second album is finished and online! > > :D > > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a > > bit more electronic than last one. > > This is absolutely great! I'm tapping my foot as I type this email, > wonderful, easy summer beats. Thanks for sharing your great work. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Wed Jul 3 21:54:21 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Thu, 4 Jul 2013 01:54:21 +0400 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Very nice, "Come and Play" is fantastic! Like the sound quality. Superb work and great achievement for you personally and for Linux Audio community in general! Thanks for this music! -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Thu Jul 4 06:24:20 2013 From: gnome at hawaii.rr.com (david) Date: Wed, 03 Jul 2013 20:24:20 -1000 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound Message-ID: <51D51514.1030007@hawaii.rr.com> For a couple of weeks now, it tries to find audio device(s), times out after about 30 seconds and kills the search process, and the system comes up with no audio whatever. aplay -l reports no soundcards found. I can modprobe snd-usb-audio and the USB card works. USB card works on other machines without any gyrations at all. I can modprobe snd-intel8x0 and ALSA then sees the onboard Intel sound. lsusb shows the sound card even when ALSA doesn't. lspci shows the onboard sound device. Kernel is 32-bit non-RT 3.7.5. alsa-base was updated to 1.0.25. There were no hardware changes. This is running on my temporary replacement very-old (12+ years old) Toshiba laptop. (The 9-year-old Toshiba laptop is dead. With grace and a declining credit card balance, a new laptop could arrive the end of this month!) Debian Sid has been running very successfully on this machine for a long while. I've set up a basic script I can run to load the two drivers, but they still refuse to load on bootup. Clues? -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From gnome at hawaii.rr.com Thu Jul 4 06:58:54 2013 From: gnome at hawaii.rr.com (david) Date: Wed, 03 Jul 2013 20:58:54 -1000 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: <51D51D2E.6060609@hawaii.rr.com> On 07/03/2013 05:05 AM, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and online! :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is > a bit more electronic than last one. > > I prepared a page for this at my new website, > http://zthmusic.se/Ordinary_Day_Montage , but my host is a little shaky, > so it might not always work. So, if that link doesn't work, you can find > the album at: > > Bandcamp > http://zthmusic.bandcamp.com/album/ordinary-day-montage > Soundcloud > https://soundcloud.com/zthmusic/sets/ordinary-day-montage > FLAC/OGG/MP3-formats for download at Piratebay: > http://thepiratebay.sx/user/zthmusic/ Hmm, many of the Pirate Bay downloads were Windows EXEs????? Trying the OGG torrent right now, but there appear to be no peers sharing it. There's one peer sharing the first FLAC file, but at 3KB/s download it's going to be a long time before I get to enjoy your album! Cannot put it up on archive.org? Downloaded from SoundCloud. > I'm very excited to be finished and to have completed this. Everything > was, as always, 100% made with Linux and Linux software. It's also > licensed CC-BY-SA. I wrote a bit about the album and what I've used > technically too at http://zthmusic.se/Ordinary_Day_Montage , if anyone > wants to check that out! > > Anyway, thank you for taking your time to listen! I greatly appreciate it! Don't have time to really listen right now, but I like what you're doing in Intro. My wife's got one of the televised music talent contests running very loudly in the same room, headphones aren't doing a very good job of isolating me from that. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From clemens at ladisch.de Thu Jul 4 08:27:13 2013 From: clemens at ladisch.de (Clemens Ladisch) Date: Thu, 04 Jul 2013 10:27:13 +0200 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D51514.1030007@hawaii.rr.com> References: <51D51514.1030007@hawaii.rr.com> Message-ID: <51D531E1.7000501@ladisch.de> david wrote: > For a couple of weeks now, it tries to find audio device(s), times out > after about 30 seconds and kills the search process, and the system > comes up with no audio whatever. > [...] > I've set up a basic script I can run to load the two drivers, but they > still refuse to load on bootup. What do you mean with "refuse"? Any rrror messages? In the system log? Regards, Clemens From gabbe.nord at gmail.com Thu Jul 4 08:57:14 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Thu, 4 Jul 2013 10:57:14 +0200 Subject: [LAU] My second album is done! In-Reply-To: <51D51D2E.6060609@hawaii.rr.com> References: <51D51D2E.6060609@hawaii.rr.com> Message-ID: Hey, sorry about the mess with the torrents! Check this page out where you can download them all: http://www.zthmusic.se/available_downloads Louigi: Thanks a lot! :D I'm very glad that you liked Come and Play, it was one of the most fun to make, as it's a pretty different song both in terms of sound and structure, from what I usually do. Thanks again! On Thu, Jul 4, 2013 at 8:58 AM, david wrote: > On 07/03/2013 05:05 AM, Gabbe Nord wrote: > >> Hello everyone! >> >> I'm very pleased to announce that my second album is finished and online! >> :D >> It's called "Ordinary Day Montage", and it consists of 8 tracks, and is >> a bit more electronic than last one. >> >> I prepared a page for this at my new website, >> http://zthmusic.se/Ordinary_**Day_Montage, but my host is a little shaky, >> so it might not always work. So, if that link doesn't work, you can find >> the album at: >> >> Bandcamp >> http://zthmusic.bandcamp.com/**album/ordinary-day-montage >> Soundcloud >> https://soundcloud.com/**zthmusic/sets/ordinary-day-**montage >> FLAC/OGG/MP3-formats for download at Piratebay: >> http://thepiratebay.sx/user/**zthmusic/ >> > > Hmm, many of the Pirate Bay downloads were Windows EXEs????? > > Trying the OGG torrent right now, but there appear to be no peers sharing > it. There's one peer sharing the first FLAC file, but at 3KB/s download > it's going to be a long time before I get to enjoy your album! > > Cannot put it up on archive.org? > > Downloaded from SoundCloud. > > > I'm very excited to be finished and to have completed this. Everything >> was, as always, 100% made with Linux and Linux software. It's also >> licensed CC-BY-SA. I wrote a bit about the album and what I've used >> technically too at http://zthmusic.se/Ordinary_**Day_Montage, if anyone >> wants to check that out! >> >> Anyway, thank you for taking your time to listen! I greatly appreciate it! >> > > Don't have time to really listen right now, but I like what you're doing > in Intro. My wife's got one of the televised music talent contests running > very loudly in the same room, headphones aren't doing a very good job of > isolating me from that. > > -- > David > gnome at hawaii.rr.com > authenticity, honesty, community > http://dancingtreefrog.com > http://clanjones.org/david/ > http://dancing-treefrog.**deviantart.com/ > ______________________________**_________________ > Linux-audio-user mailing list > Linux-audio-user at lists.**linuxaudio.org > http://lists.linuxaudio.org/**listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Thu Jul 4 09:47:59 2013 From: gnome at hawaii.rr.com (david) Date: Wed, 03 Jul 2013 23:47:59 -1000 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D531E1.7000501@ladisch.de> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> Message-ID: <51D544CF.80509@hawaii.rr.com> On 07/03/2013 10:27 PM, Clemens Ladisch wrote: > david wrote: >> For a couple of weeks now, it tries to find audio device(s), times out >> after about 30 seconds and kills the search process, and the system >> comes up with no audio whatever. >> [...] >> I've set up a basic script I can run to load the two drivers, but they >> still refuse to load on bootup. > > What do you mean with "refuse"? Any rrror messages? In the system log? According to syslog, it's finds and sets up both the USB device and the Intel snd_intel8x0 just fine. No errors there. OK, so I guess they're loading. But ALSA still insists there are no soundcards. Does ALSA have a log somewhere? If I watch the screen during bootup, it's udev timing out when trying to do something with the USB controller the USB card is connected to. The udev error doesn't appear in dmesg or any logs. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From brendan.jones.it at gmail.com Thu Jul 4 10:10:48 2013 From: brendan.jones.it at gmail.com (Brendan Jones) Date: Thu, 04 Jul 2013 12:10:48 +0200 Subject: [LAU] Fedora Jam Message-ID: <51D54A28.3010402@gmail.com> Hi all, just a quick announcement of the official Fedora Jam audio Spin. Some of you may have already been using the nightly composes, but now you can find the published Fedora 19 version here: http://spins.fedoraproject.org/jam-kde/ For linuxsampler, PureData, SuperCollider and others, the Planet CCRMA repository is now available for Fedora 19 (thanks Fernando!). http://ccrma.stanford.edu/planetccrma/software/ cheers Brendan From julien at mail.upb.de Thu Jul 4 10:20:18 2013 From: julien at mail.upb.de (Julien Claassen) Date: Thu, 4 Jul 2013 12:20:18 +0200 (CEST) Subject: [LAU] My second album is done! In-Reply-To: References: <51D51D2E.6060609@hawaii.rr.com> Message-ID: Hej Gabbe! It's very nice of you to upload the album to your website, but your server is really shoddy. Thus I can't complete a download. I think I might wait a little. Either it will get better later on or maybe you can get a torrent running after all. Warm regards and thanks for your efforts Julien ---------------------------------------- http://juliencoder.de/nama/music.html From dlphillips at woh.rr.com Thu Jul 4 10:21:49 2013 From: dlphillips at woh.rr.com (Dave Phillips) Date: Thu, 04 Jul 2013 06:21:49 -0400 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: <51D54CBD.6050703@woh.rr.com> On 07/03/2013 11:05 AM, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and > online! :D Nicely done ! Beautiful clear sound through studio phones, I'll check it out on the main system after the wife wakes up today. :) Best, dp From brendan.jones.it at gmail.com Thu Jul 4 11:14:34 2013 From: brendan.jones.it at gmail.com (Brendan Jones) Date: Thu, 04 Jul 2013 13:14:34 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: <51D5591A.4080407@gmail.com> On 07/03/2013 05:05 PM, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and online! :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is > a bit more electronic than last one. > > I prepared a page for this at my new website, > http://zthmusic.se/Ordinary_Day_Montage , but my host is a little shaky, > so it might not always work. So, if that link doesn't work, you can find > the album at: > > Bandcamp > http://zthmusic.bandcamp.com/album/ordinary-day-montage > Soundcloud > https://soundcloud.com/zthmusic/sets/ordinary-day-montage > FLAC/OGG/MP3-formats for download at Piratebay: > http://thepiratebay.sx/user/zthmusic/ > > I'm very excited to be finished and to have completed this. Everything > was, as always, 100% made with Linux and Linux software. It's also > licensed CC-BY-SA. I wrote a bit about the album and what I've used > technically too at http://zthmusic.se/Ordinary_Day_Montage , if anyone > wants to check that out! > > Anyway, thank you for taking your time to listen! I greatly appreciate it! Awesome work yet again. Congrats Gabbe! From clemens at ladisch.de Thu Jul 4 11:26:06 2013 From: clemens at ladisch.de (Clemens Ladisch) Date: Thu, 04 Jul 2013 13:26:06 +0200 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D544CF.80509@hawaii.rr.com> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> Message-ID: <51D55BCE.9040904@ladisch.de> david wrote: > On 07/03/2013 10:27 PM, Clemens Ladisch wrote: >> Any rrror messages? In the system log? > > According to syslog, it's finds and sets up both the USB device and > the Intel snd_intel8x0 just fine. No errors there. OK, so I guess > they're loading. But ALSA still insists there are no soundcards. Does > ALSA have a log somewhere? That is the ALSA log. Do you have the device nodes in /dev/snd/? Are they accessible for your user? > If I watch the screen during bootup, it's udev timing out when trying > to do something with the USB controller the USB card is connected to. Is that the message? "doing something ... timed out"? Regards, Clemens From jeremy at autostatic.com Thu Jul 4 11:40:54 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 04 Jul 2013 13:40:54 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <62544.188.26.169.14.1372691772.squirrel@boosthardware.com> References: <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> <51D195BD.5000209@autostatic.com> <62544.188.26.169.14.1372691772.squirrel@boosthardware.com> Message-ID: <51D55F46.3040308@autostatic.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/01/2013 05:16 PM, Patrick Shirkey wrote: > > On Tue, July 2, 2013 12:44 am, Jeremy Jongepier wrote: >> On 07/01/2013 04:12 PM, Patrick Shirkey wrote: >>> IIUC the potential issues with period sizes with non powers of >>> two are at the application level. JACK is agnostic about it. >>> >> >> Yes, guitarix for instance doesn't like period sizes that are not >> a power of two. >> >>> Anyway it's a little spooky that you get better performance >>> with -p64 than -p128 even in playback only mode. >> >> Well I'm testing and haven't done any real stress-tests yet. >> Maybe it was a one-off. At least I get undistorted sound out of >> this device. Now if I could only figure out why JACK bails out >> every once so often. > > Does this happen with or without wifi enabled? > >> And why full-duplex doesn't work, well, it works but the audio >> coming out is distorted. >> > > System load causing distortion on the Audio device in duplex mode. > It suggests something is wrong with the bus and or interrupts but I > don't see why it would be ok in playback only mode. Not sure if it > is related but I have noticed that my old usb quattro often takes a > while to warm up. Sometimes I have to run it for a few hours and > start/stop jack several times to get undistorted output. > > Another issue is the Rockchip devs probably haven't tested the > device at low latency with duplex mode so they haven't picked up > that issue. We can try flagging it with them. > > I forgot what the audio chipset is. Can you post the output of cat > /proc/asound/cards? > jeremy at rk3066:~$ cat /proc/asound/cards 0 [RK29RK1000 ]: RK29_RK1000 - RK29_RK1000 RK29_RK1000 1 [HDMI ]: ROCKCHIP_HDMI - ROCKCHIP HDMI ROCKCHIP HDMI 2 [Device ]: USB-Audio - Generic USB Audio Device Generic USB Audio Device at usb-usb20_host-1.1, full speed 'Device' is a $2 USB audio interface with a C-Media chipset. But I got one step further: jeremy at rk3066:~$ uname -a Linux rk3066 3.0.36-rt58 #1 SMP PREEMPT RT Thu Jul 4 13:18:23 CEST 2013 armv7l GNU/Linux So I've got a RT kernel running on it. Moving on :) Jeremy > > > > -- 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 > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with undefined - http://www.enigmail.net/ iQIcBAEBAgAGBQJR1V9GAAoJEF63J34m6LiUyVIP/AsanFJ8SIpGZroYhLppKSL6 pcaxTQBI/FrC1W6vXCMywVEXuLtmarZ6XGMljj7F1y97Mi/k9YgBgV7s9Xem1WyD uPGO5/tbAt1qMxVxGSi4qodf1nJYadC8/qJzWGq+vqGJMAGZiLrzSd7pjKt8L3/2 1oL6WkOBkpySef94zBgJgRHhca7UEQMv4aH0/92gutE3Wex6ctbEm5GvwCc3mLyS rAZhsHVmG2Pp1odj4r+4a5KIc3ouKv3ZtwrgVCy2Yz1CQoeD7JoyjQgyvj8gS3Vi K2+cqBwwpNs5uplU6XLMmDNPpdpGF/f/1hqbD1wlhHIy46jWQebJTJCcQL7pH86O L7ESyd9FiY8g6DtqOCN6DW/MGRBjWEEGTLtIQAn94AgeQ4VhAbrM4as0QFpr+1GI XYuFZhOww6mBZZKWCbzOAk+sRYJvnatIfrRIJhSyly/PKPT9OFjqGKd3x+V91370 Lv8wv4sew1DfaB/GSLh8CmyMyGygBR042o56jkJK5WypAmoke+evYl5HitEcqP9w LTLraRgE09Q3I8YVsok4Sp9iGt612yj+3kkdMDpejjgTY/ELluYBX/OOJq2rVt9x fnKEU+1oGSOC30mGtECOILKdd4cbFIqF5I47wRBAqJuABCO455puzo6tETiaO3+Q v1Cer+mb9mr/T3rOP5u7 =iTTB -----END PGP SIGNATURE----- From treees at gmail.com Thu Jul 4 15:37:05 2013 From: treees at gmail.com (=?ISO-8859-1?Q?Juan_Fabi=E1n?=) Date: Thu, 4 Jul 2013 17:37:05 +0200 Subject: [LAU] My second album is done! In-Reply-To: <51D5591A.4080407@gmail.com> References: <51D5591A.4080407@gmail.com> Message-ID: Wow, great job, amazing sound and great tunes. I'd love to know how did you setup the drumset, are they the defaults in Hydrogen? Maybe i'll touch a bit the sound of hhat, cymbals and ride to make it more "real". For my ears that would be just perfect. Anyway, i'm absolutely amazed by your sounds! 2013/7/4 Brendan Jones > On 07/03/2013 05:05 PM, Gabbe Nord wrote: > >> Hello everyone! >> >> I'm very pleased to announce that my second album is finished and online! >> :D >> It's called "Ordinary Day Montage", and it consists of 8 tracks, and is >> a bit more electronic than last one. >> >> I prepared a page for this at my new website, >> http://zthmusic.se/Ordinary_**Day_Montage, but my host is a little shaky, >> so it might not always work. So, if that link doesn't work, you can find >> the album at: >> >> Bandcamp >> http://zthmusic.bandcamp.com/**album/ordinary-day-montage >> Soundcloud >> https://soundcloud.com/**zthmusic/sets/ordinary-day-**montage >> FLAC/OGG/MP3-formats for download at Piratebay: >> http://thepiratebay.sx/user/**zthmusic/ >> >> I'm very excited to be finished and to have completed this. Everything >> was, as always, 100% made with Linux and Linux software. It's also >> licensed CC-BY-SA. I wrote a bit about the album and what I've used >> technically too at http://zthmusic.se/Ordinary_**Day_Montage, if anyone >> wants to check that out! >> >> Anyway, thank you for taking your time to listen! I greatly appreciate it! >> > Awesome work yet again. Congrats Gabbe! > > ______________________________**_________________ > 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 rustys.lists at gmail.com Thu Jul 4 17:10:36 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Thu, 4 Jul 2013 10:10:36 -0700 Subject: [LAU] My second album is done! In-Reply-To: References: <51D5591A.4080407@gmail.com> Message-ID: Nice work Gabbe, I have to admit that the genre is not exactly my thing, but I was quite impressed by your manipulation of effects and sculpting of the instrument sounds. I kindof enjoyed the reverse sounds, and sortof envelope manipulations. I also enjoyed your rhythmic play on a couple of the tunes. It was quite clear and clean through my studio phones. Nice work! Rusty On 7/4/13, Juan Fabi?n wrote: > Wow, great job, amazing sound and great tunes. I'd love to know how did you > setup the drumset, are they the defaults in Hydrogen? Maybe i'll touch a > bit the sound of hhat, cymbals and ride to make it more "real". For my ears > that would be just perfect. > > Anyway, i'm absolutely amazed by your sounds! > > > 2013/7/4 Brendan Jones > >> On 07/03/2013 05:05 PM, Gabbe Nord wrote: >> >>> Hello everyone! >>> >>> I'm very pleased to announce that my second album is finished and >>> online! >>> :D >>> It's called "Ordinary Day Montage", and it consists of 8 tracks, and is >>> a bit more electronic than last one. >>> >>> I prepared a page for this at my new website, >>> http://zthmusic.se/Ordinary_**Day_Montage, >>> but my host is a little shaky, >>> so it might not always work. So, if that link doesn't work, you can find >>> the album at: >>> >>> Bandcamp >>> http://zthmusic.bandcamp.com/**album/ordinary-day-montage >>> Soundcloud >>> https://soundcloud.com/**zthmusic/sets/ordinary-day-**montage >>> FLAC/OGG/MP3-formats for download at Piratebay: >>> http://thepiratebay.sx/user/**zthmusic/ >>> >>> I'm very excited to be finished and to have completed this. Everything >>> was, as always, 100% made with Linux and Linux software. It's also >>> licensed CC-BY-SA. I wrote a bit about the album and what I've used >>> technically too at >>> http://zthmusic.se/Ordinary_**Day_Montage, >>> if anyone >>> wants to check that out! >>> >>> Anyway, thank you for taking your time to listen! I greatly appreciate >>> it! >>> >> Awesome work yet again. Congrats Gabbe! >> >> ______________________________**_________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.**linuxaudio.org >> http://lists.linuxaudio.org/**listinfo/linux-audio-user >> > From rustys.lists at gmail.com Thu Jul 4 17:36:50 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Thu, 4 Jul 2013 10:36:50 -0700 Subject: [LAU] eyes verses ears Message-ID: Hi folks, Let me start out by saying that I don't intend for this to be a "which is better" discussion. This is more of a "why?" discussion, and, "what can you do with it which can't be done otherwise?" discussion. In reviewing the list of applications Gabbe used on the new album, there is mention of an application used to visualize the mix. I got to wondering. A mix is purely an auditory medium, right? So, (and I'll play dumb here) why do we need to visualize a mix? Now, part of this is purely selfish, since I happen to be blind, and so these visualization tools are not available to me. But, though I understand the advantages of an onscreen mixer, or a mouse driven interface, since, theoretically everything conveyed in an audible mix is received by the ears, shouldn't it be enough to experience it, perceive it, analyze it audibly? Now, of course, I understand that deficiencies in speakers or headphones, audio volume limitations, maybe even hearing limitations are likely some reasons for using a visual method for analyzing an audio medium, but are there others? I also understand the idea that, if a tool is available, then it should, be used, but is there something which can be achieved, in the audio spectrum, which can only be achieved via the visual medium? Just some thoughts. Rusty From bob at mellowood.ca Thu Jul 4 18:10:04 2013 From: bob at mellowood.ca (Bob van der Poel) Date: Thu, 4 Jul 2013 11:10:04 -0700 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: Interesting points ... especially today, since I just got "elected" to conduct a new concert band we're setting up here (for beginners ... should be interesting). But, having done a bit of study about the role of a conductor it really does bring the "audio only" question to the forefront. And I'd not talking about the highly visible music we see at rock concerts, etc. As an audience member I find myself playing lots of attention to the conductor as well as the physical movements of players: the tension in the face of a cello player, the stance of a trumpeter, etc. So, for live music ... yeah, the visual becomes quite important. But, when we translate this to a recording, the visual is all but lost. Even in a video recording, I get very little sense of the visual. Maybe I need a much bigger screen? I have decent vision, so I have no idea how this translates to a blind person. I can only assume that other senses assume the role of sight? On Thu, Jul 4, 2013 at 10:36 AM, Rusty Perez wrote: > Hi folks, > Let me start out by saying that I don't intend for this to be a > "which is better" discussion. This is more of a "why?" discussion, > and, "what can you do with it which can't be done otherwise?" > discussion. > > In reviewing the list of applications Gabbe used on the new album, > there is mention of an application used to visualize the mix. I got > to wondering. > A mix is purely an auditory medium, right? So, (and I'll play dumb > here) why do we need to visualize a mix? > Now, part of this is purely selfish, since I happen to be blind, and > so these visualization tools are not available to me. But, though I > understand the advantages of an onscreen mixer, or a mouse driven > interface, since, theoretically everything conveyed in an audible mix > is received by the ears, shouldn't it be enough to experience it, > perceive it, analyze it audibly? > > Now, of course, I understand that deficiencies in speakers or > headphones, audio volume limitations, maybe even hearing limitations > are likely some reasons for using a visual method for analyzing an > audio medium, but are there others? > > I also understand the idea that, if a tool is available, then it > should, be used, but is there something which can be achieved, in the > audio spectrum, which can only be achieved via the visual medium? > > Just some thoughts. > Rusty > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob at mellowood.ca WWW: http://www.mellowood.ca From surfacepatterns at gmail.com Thu Jul 4 18:51:00 2013 From: surfacepatterns at gmail.com (Devin Anderson) Date: Thu, 4 Jul 2013 11:51:00 -0700 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Hi Gabbe, On Wed, Jul 3, 2013 at 8:05 AM, Gabbe Nord wrote: > I'm very pleased to announce that my second album is finished and online! :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a > bit more electronic than last one. I'm listening to your album while writing code. I'm really enjoying your music. The melodies are beautiful. The transitions between songs are beautifully executed, and yet each song is obviously distinct from the others before it. Each part sits in the mix on its own, and doesn't crowd the other parts. If I had to pick one target for improvement, I'd say that the mix feels a bit too compressed in some parts of the first couple songs; however, I think now that might be an artistic choice, as the fourth and fifth songs have a lot of interesting, fantastic dynamics. I love the use of resonance in the synth sound at the end of "Come and Play". Great job. -- Devin Anderson surfacepatterns (at) gmail (dot) com blog - http://surfacepatterns.blogspot.com/ midisnoop - http://midisnoop.googlecode.com/ psinsights - http://psinsights.googlecode.com/ synthclone - http://synthclone.googlecode.com/ From pshirkey at boosthardware.com Thu Jul 4 19:00:50 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Fri, 5 Jul 2013 05:00:50 +1000 (EST) Subject: [LAU] Google TV device w/ AV out In-Reply-To: <51D55F46.3040308@autostatic.com> References: <51A35F6F.3050807@autostatic.com> <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> <51D195BD.5000209@autostatic.com> <62544.188.26.169.14.1372691772.squirrel@boosthardware.com> <51D55F46.3040308@autostatic.com> Message-ID: <54257.188.26.10.227.1372964450.squirrel@boosthardware.com> On Thu, July 4, 2013 9:40 pm, Jeremy Jongepier wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/01/2013 05:16 PM, Patrick Shirkey wrote: >> >> On Tue, July 2, 2013 12:44 am, Jeremy Jongepier wrote: >>> On 07/01/2013 04:12 PM, Patrick Shirkey wrote: >>>> IIUC the potential issues with period sizes with non powers of >>>> two are at the application level. JACK is agnostic about it. >>>> >>> >>> Yes, guitarix for instance doesn't like period sizes that are not >>> a power of two. >>> >>>> Anyway it's a little spooky that you get better performance >>>> with -p64 than -p128 even in playback only mode. >>> >>> Well I'm testing and haven't done any real stress-tests yet. >>> Maybe it was a one-off. At least I get undistorted sound out of >>> this device. Now if I could only figure out why JACK bails out >>> every once so often. >> >> Does this happen with or without wifi enabled? >> >>> And why full-duplex doesn't work, well, it works but the audio >>> coming out is distorted. >>> >> >> System load causing distortion on the Audio device in duplex mode. >> It suggests something is wrong with the bus and or interrupts but I >> don't see why it would be ok in playback only mode. Not sure if it >> is related but I have noticed that my old usb quattro often takes a >> while to warm up. Sometimes I have to run it for a few hours and >> start/stop jack several times to get undistorted output. >> >> Another issue is the Rockchip devs probably haven't tested the >> device at low latency with duplex mode so they haven't picked up >> that issue. We can try flagging it with them. >> >> I forgot what the audio chipset is. Can you post the output of cat >> /proc/asound/cards? >> > > jeremy at rk3066:~$ cat /proc/asound/cards > 0 [RK29RK1000 ]: RK29_RK1000 - RK29_RK1000 > RK29_RK1000 > 1 [HDMI ]: ROCKCHIP_HDMI - ROCKCHIP HDMI > ROCKCHIP HDMI > 2 [Device ]: USB-Audio - Generic USB Audio Device > Generic USB Audio Device at usb-usb20_host-1.1, > full speed > > 'Device' is a $2 USB audio interface with a C-Media chipset. Have you been using this for testing at low latency? > But I got > one step further: > > jeremy at rk3066:~$ uname -a > Linux rk3066 3.0.36-rt58 #1 SMP PREEMPT RT Thu Jul 4 13:18:23 CEST > 2013 armv7l GNU/Linux > > So I've got a RT kernel running on it. Moving on :) > Nicely done! What was the blocker in the end? -- Patrick Shirkey Boost Hardware Ltd From jeremy at autostatic.com Thu Jul 4 19:13:27 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 04 Jul 2013 21:13:27 +0200 Subject: [LAU] Google TV device w/ AV out In-Reply-To: <54257.188.26.10.227.1372964450.squirrel@boosthardware.com> References: <51B9B2AC.7090807@autostatic.com> <51BEF851.80306@autostatic.com> <51C02CD7.6050000@autostatic.com> <51C1B49D.8060500@autostatic.com> <51CD9504.8080201@autostatic.com> <65215.188.26.169.14.1372467499.squirrel@boosthardware.com> <51CED1CE.6090406@autostatic.com> <57008.188.26.169.14.1372543051.squirrel@boosthardware.com> <51D16858.8090908@autostatic.com> <52909.89.47.0.197.1372679426.squirrel@boosthardware.com> <51D16F50.2080206@autostatic.com> <53825.89.47.0.197.1372687963.squirrel@boosthardware.com> <51D195BD.5000209@autostatic.com> <62544.188.26.169.14.1372691772.squirrel@boosthardware.com> <51D55F46.3040308@autostatic.com> <54257.188.26.10.227.1372964450.squirrel@boosthardware.com> Message-ID: <51D5C957.8050603@autostatic.com> On 07/04/2013 09:00 PM, Patrick Shirkey wrote: >> jeremy at rk3066:~$ cat /proc/asound/cards >> > 0 [RK29RK1000 ]: RK29_RK1000 - RK29_RK1000 >> > RK29_RK1000 >> > 1 [HDMI ]: ROCKCHIP_HDMI - ROCKCHIP HDMI >> > ROCKCHIP HDMI >> > 2 [Device ]: USB-Audio - Generic USB Audio Device >> > Generic USB Audio Device at usb-usb20_host-1.1, >> > full speed >> > >> > 'Device' is a $2 USB audio interface with a C-Media chipset. > Have you been using this for testing at low latency? > Yes. It's actually the same as the device used here: http://wiki.linuxaudio.org/wiki/jack_latency_tests > >> > But I got >> > one step further: >> > >> > jeremy at rk3066:~$ uname -a >> > Linux rk3066 3.0.36-rt58 #1 SMP PREEMPT RT Thu Jul 4 13:18:23 CEST >> > 2013 armv7l GNU/Linux >> > >> > So I've got a RT kernel running on it. Moving on :) >> > > Nicely done! > > What was the blocker in the end? drivers/mmc/host/rk29_sdmmc.c I've attached a patch. It's probably a quick and dirty hack but it seems to work ok, ran it for a couple of hours today without any issues. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: rk29_sdmmc.c.diff Type: text/x-patch Size: 389 bytes Desc: not available URL: From gabbe.nord at gmail.com Thu Jul 4 20:11:53 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Thu, 4 Jul 2013 22:11:53 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Thank you so much for you comments everyone!! Really wonderful to hear, thanks! Brendan: We spoke briefly on IRC, but thank you very much again! Juan: Thank you very much! It's actually not Hydrogen at all, but only the Salamander Drumkit through LinuxSampler, sequenced in Ardour. And I agree with the realism on the drums, I wish I had more of an ear for what sounds real and not with drums. However, usually I make a conscious choice to trade realism for an additional sound I like or so. Like in the 3rd song Still, where I'd like to meet the drummer who could play the ride, hihat and snare the same way that I've made it :D Rusty: Thanks! I'm glad the production seems to hold in your headphones too! Devin: Oh, thank you so much! It makes me very glad that you commented the transitions as well, I was aware it was a bit of a gamble to put them in, but hopefully they weren't too much! On Thu, Jul 4, 2013 at 8:51 PM, Devin Anderson wrote: > Hi Gabbe, > > On Wed, Jul 3, 2013 at 8:05 AM, Gabbe Nord wrote: > > > I'm very pleased to announce that my second album is finished and > online! :D > > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a > > bit more electronic than last one. > > I'm listening to your album while writing code. I'm really enjoying > your music. The melodies are beautiful. The transitions between > songs are beautifully executed, and yet each song is obviously > distinct from the others before it. Each part sits in the mix on its > own, and doesn't crowd the other parts. > > If I had to pick one target for improvement, I'd say that the mix > feels a bit too compressed in some parts of the first couple songs; > however, I think now that might be an artistic choice, as the fourth > and fifth songs have a lot of interesting, fantastic dynamics. > > I love the use of resonance in the synth sound at the end of "Come and > Play". > > Great job. > > -- > Devin Anderson > surfacepatterns (at) gmail (dot) com > > blog - http://surfacepatterns.blogspot.com/ > midisnoop - http://midisnoop.googlecode.com/ > psinsights - http://psinsights.googlecode.com/ > synthclone - http://synthclone.googlecode.com/ > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rustys.lists at gmail.com Thu Jul 4 20:45:23 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Thu, 4 Jul 2013 13:45:23 -0700 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: Hi Bob, That's not quite the direction I was going, but very interesting nonetheless. I know people enjoy watching me perform--I play guitar and sing, and am a looper/percussionist as well. I am familiar with the concept of the "solo" face for guitarists. :-) I would imagine that watching a person perform live must be just as facinating as the music. People always comment on the visual aspects of a musical performance, and sometimes I think it's meaningless, and other times I don't, because it quite obviously is part of the experience for most every one else. Rusty On 7/4/13, Bob van der Poel wrote: > Interesting points ... especially today, since I just got "elected" to > conduct a new concert band we're setting up here (for beginners ... > should be interesting). But, having done a bit of study about the role > of a conductor it really does bring the "audio only" question to the > forefront. And I'd not talking about the highly visible music we see > at rock concerts, etc. > > As an audience member I find myself playing lots of attention to the > conductor as well as the physical movements of players: the tension in > the face of a cello player, the stance of a trumpeter, etc. > > So, for live music ... yeah, the visual becomes quite important. But, > when we translate this to a recording, the visual is all but lost. > Even in a video recording, I get very little sense of the visual. > Maybe I need a much bigger screen? > > I have decent vision, so I have no idea how this translates to a blind > person. I can only assume that other senses assume the role of sight? > > On Thu, Jul 4, 2013 at 10:36 AM, Rusty Perez > wrote: >> Hi folks, >> Let me start out by saying that I don't intend for this to be a >> "which is better" discussion. This is more of a "why?" discussion, >> and, "what can you do with it which can't be done otherwise?" >> discussion. >> >> In reviewing the list of applications Gabbe used on the new album, >> there is mention of an application used to visualize the mix. I got >> to wondering. >> A mix is purely an auditory medium, right? So, (and I'll play dumb >> here) why do we need to visualize a mix? >> Now, part of this is purely selfish, since I happen to be blind, and >> so these visualization tools are not available to me. But, though I >> understand the advantages of an onscreen mixer, or a mouse driven >> interface, since, theoretically everything conveyed in an audible mix >> is received by the ears, shouldn't it be enough to experience it, >> perceive it, analyze it audibly? >> >> Now, of course, I understand that deficiencies in speakers or >> headphones, audio volume limitations, maybe even hearing limitations >> are likely some reasons for using a visual method for analyzing an >> audio medium, but are there others? >> >> I also understand the idea that, if a tool is available, then it >> should, be used, but is there something which can be achieved, in the >> audio spectrum, which can only be achieved via the visual medium? >> >> Just some thoughts. >> Rusty >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > -- > **** Listen to my CD at http://www.mellowood.ca/music/cedars **** > Bob van der Poel ** Wynndel, British Columbia, CANADA ** > EMAIL: bob at mellowood.ca > WWW: http://www.mellowood.ca > From julien at mail.upb.de Thu Jul 4 21:11:29 2013 From: julien at mail.upb.de (Julien Claassen) Date: Thu, 4 Jul 2013 23:11:29 +0200 (CEST) Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: Hello Rusty! I didn't read the other replies, but here are a couple of thoughts. 1. Visual feedback can speed up work. Seeing is two-dimensional and instantaneous. So you can see something quickly, whichlistening takes a while to find. 2. Modern mixing techniques. Take a metal recording, you want the bassdrum at exactly the same instant, that the bass is playing, so you can use compressors and some other processing to great effect and maximum volume. The bass is played live and so are the drums. For drums, there have been times, where people used triggering of samples and then later corrected only the important parts via MIDI quantisation. In all other cases it's a question of cutting things out and moving them with precision. 3. Finding peaks and other annomalities in the frequency spectrum. You can hear it, but again, it can be quicker to start from the visual image. I'm sure there are more uses for all kinds of visualisation utilities. Some more useful than others. Some of these things can be managed by ear easily, others with difficulty and yet others might be managed by some clever automated analysing software. But then a computer is sometimes not as good as a human. It is fast, but it doesn't know much about difficult decisions. These would have to be programmed. They could take up CPU and they might yield bad results. Thus it's a question of which is more work: looking at it and doing it that way from scratch or using an automated system and correcting the errors. That's my take on it. One last probable addition could be, that entering things on a keyboard, together with the time limitation on sound, that I mentioned before, it takes time, is prone to errors and typos. A mouse click in such instances is very fast. Warm regards and good luck in your quest for knowledge Julien ---------------------------------------- http://juliencoder.de/nama/music.html From julien at mail.upb.de Thu Jul 4 22:11:14 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 5 Jul 2013 00:11:14 +0200 (CEST) Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Hello Gabbe! Before I begin to say any word about the music you made, I will say: thank you for all the troubles you got up to to give me a chance to listen to it. This is very kind of you and - be assured - very much appreciated! As a whole, I very much agree with Dave's vote of the clear sound and good mix. The mixes have power, sound well balanced to me. What's even better, is that they fall on willing ears. In recent weeks I've turned more and more towards electronic music. the intro is a good introduction to the sonic world of your album. Slowly taking me in, preparing for the Skandinavian sound and feel. The width and openness. Only to surprise me later on. there is that Skandinavian feel poking through, but not as obvious as before. I like the bassdrum of the intro. It appears later on again. It has bite and pumps a little, not in a compression artefact way. It's somewhere between an edited acoustic bassdrum and something purely electronic. It might even fill the four-to-the-floor niche. It is alive is the nice four-to-the-floor piece. And it already shows something, which characterises the whole album. The tracks flow seemlessly from one into the other. Not only by ambient sounds, but by concept. If it had been me, I'd have mixed the snare louder. But that way it keeps the attention, while still deliviering. I can't completely let go with that, because, there is that small bit, which seems so uncharacteristic. I also love the somehow oneiric harmonies. Tending towards minor, yet rarely pure minor chords. While having a touch of 90s nostalgia about it, the mix sounds very much like the current decade. Do you use "overcompression" to get this slightly pumping effect? I noticed that throughout the album and had been wondering. If so: how do you compress? The whole song or only certain groups of instruments? - Again the bassdrum is nice, crisp and very well defined. Ha! The blend to Still is lovely. I recognise the sound and I must admit, that I never dreamt it could be used in such a place. Well sed. Still: I think here the bassdrum is a little too loud in the mix, but not so loud as to bannish the thought of enjoyment. :-) And so we have turned to a more hiphop styled track. And it still keeps in the spirit of the earlier tracks. As I said: it's a sonic world, this album. Listening to it again, I feel, that so far, it appears more like a change in arrangement and style than in harmony. It's bordering on my threshhold of boredom. If it wasn't for the very nice mixes and well executed styles and the morphing of one into the other... but there is that and it holds me. You know, that my threshhold of boredom is quite low. Since I'm still more of a prog and baroque listener than electronica. Come and play; why not. I'm game. :-) This sets a much lighter tone. I love the slight touch of breakbeat in this piece. Also the main instrument is interesting. the way the piece changes from almost elevator music into almost dubstep is great. The way the movement picks up and the arrangement fills up from almost meditative spheric guitar to the full beat. Teh effects you used, the well-placed delay and how you managed to use that to instantly isolate the main "lead" and the rhythm in places. Well done. good trick! Full Moon baffled me at first. Perhaps because I approached it with preformed notions, of what it should have been. That way it's the piece, that I still perceive as the weakest piece on this album. It sounds to common place for me. To worldly. The introduction and the "bridge" are the best bits. They talk to me of a full moon and the quiet night. The rest is more of the beach party next door with its drinking, barbecuing crowd. At least it's a Swedish party and not one of those Mallorca crimes, that they probably call "a beach happening" or "afterwork holiday". :-) Get out the "let oel" and let's say "skol" (sorry can't spell it and didn't find it quickly :-) ). I wonder, what you thought, when you called the next composition "Touchdown Germany"? Was it the ambient sound from the lecture hall at my uni? Or is it the plump beat? That beat, which makes me think of Flaubert, who said something about music of the heavens, that we hear, but only produce something, which sounds like it's being played on rusting tin drums, written for crippled bears. :-) It has a very raw edge this piece, while keeping with the flow of the dreamy melodies. the Circus reminds me even more of dubstep. Not really, but the stuttering rhythm and this change from third notes in a quarter beat. Only your version of it is like Debusy's sweet juicy apple eaten with knife and fork in comparison to Gershwin. Neither is wrong or right, but one is more tamed than the other. Rhythmic acrobatics in pensive music. A circus indeed. :-) Have you ever been to Korea? If my quick look at google didn't lead me astray completely, it should be somewhere in South Korea. Whereever Kojan is, we have returned to the good, old idiom of hiphop with the drumkit from the first track. If I were to analyse this piece, I'd say the instrument reminiscent of an electric piano in the middle has a slight touch of typica Asian harmony and the sawtoothy synth almost gives an impression of chiptunes synthesis. With the impressions we get, I might suggest, that this is to conjure up an idea of all those game consoles, that the youngsters over there love so much or at least use so much. One of my old classmates would at least agree with the interest in game consoles, if not wioth my implications on this piece. :-) In a way it's a good song to lead me out. It seems so much more worldly. The applause, the crowd, the rest of the world, that stands so far removed from the images of moonlit nights - parties or no parties - and the still, windy atmosphere of the album. You worked a great deal with white noise, which gives one the feeling of wind in the dunes. It's a sound I heard a lot in radioplays and this white noise patch - or patches - are close to that sound. Somewhere between wind and waves or the mixture of the two? All in all, it's a good album and I will be sure to listen to it again. Perhaps not in one go, as I did today, but now I've taken in teh atmosphere, I can quickly travel back there. It's a good piece of work Always slightly dreamy, dark, depressive and still with a consoling lightness behind the pregnant beats and occasional distorted synths. A good balance between the single mindedness of centre beats and basses and parnoramic harmonies and effects. I like it. And I hope, that you will get a whole load of feedback and a couple of new listeners. What will you give us next time? Perhaps once more something with the occasional lyrics? Some Swedish hiphop or some English song of longing? Or something different? Warm regards and thanks for sharing Julien ---------------------------------------- http://juliencoder.de/nama/music.html From gnome at hawaii.rr.com Thu Jul 4 23:02:37 2013 From: gnome at hawaii.rr.com (david) Date: Thu, 04 Jul 2013 13:02:37 -1000 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D55BCE.9040904@ladisch.de> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> <51D55BCE.9040904@ladisch.de> Message-ID: <51D5FF0D.9030507@hawaii.rr.com> On 07/04/2013 01:26 AM, Clemens Ladisch wrote: > david wrote: >> On 07/03/2013 10:27 PM, Clemens Ladisch wrote: >>> Any rrror messages? In the system log? >> >> According to syslog, it's finds and sets up both the USB device and >> the Intel snd_intel8x0 just fine. No errors there. OK, so I guess >> they're loading. But ALSA still insists there are no soundcards. Does >> ALSA have a log somewhere? > > That is the ALSA log. > > Do you have the device nodes in /dev/snd/? Are they accessible for > your user? The only two showing are seq and timer. Both are accessible to the audio group, which I'm part of. >> If I watch the screen during bootup, it's udev timing out when trying >> to do something with the USB controller the USB card is connected to. > > Is that the message? "doing something ... timed out"? Let me see if I can scroll lock the display long enough to note down exactly what it reports. OK, here goes! udevd[418]: failed to execute '/lib/udev/[mtp-probe' 'mtp-probe /sys/devices/pci0000:00/0000:00:1d.7/usb1/1-2 1 3': no such file or directory udevd[420]: failed to execute '/lib/udev/[mtp-probe' 'mtp-probe /sys/devices/pci0000:00/0000:00:1d.1/usb3/3-1 3 2': no such file or directory udevd[446]: failed to execute '/lib/udev/[mtp-probe' 'mtp-probe /sys/devices/pci0000:00/0000:00:1d.7/usb1/1-1/1/1.4/1-1.4.4 1 7': no such file or directory udevd[391]: 'sbin/modprobe -b pci:v00008086d000024C5sv00001179sd00000241bc04sc01i00' [396] udevd[391]: 'sbin/modprobe -b pci:v00008086d000024C5sv00001179sd00000241bc04sc01i00' [396] terminated by signal 9 [Killed] udevd[378]: 'sbin/modprobe -b usb:v088Bp2902d0100dc00sc00do00ic01sc01ip00in00' [501] udevd[378]: 'sbin/modprobe -b usb:v088Bp2902d0100dc00sc00do00ic01sc01ip00in00' [501] terminated by signal 9 [Killed] Later there's an entry saying alsactl failed to restore, no sound cards found. PCI devices 1d.1 and 1d.7 appear on lspci: 00:1d.1 USB controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI Controller #2 (rev 03) 00:1d.7 USB controller: Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller (rev 03) USB controller 1 is this device on lspci: 00:1d.0 USB controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI Controller #1 (rev 03) The built-in Intel audio is pci device 1f.d. The USB audio is this device from lsusb: Bus 003 Device 002: ID 08bb:2902 Texas Instruments PCM2902 Audio Codec Don't know how to map from USB controller to bus. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From dawsonwu at rahul.net Fri Jul 5 00:12:58 2013 From: dawsonwu at rahul.net (Ken Dawson) Date: Thu, 04 Jul 2013 17:12:58 -0700 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D5FF0D.9030507@hawaii.rr.com> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> <51D55BCE.9040904@ladisch.de> <51D5FF0D.9030507@hawaii.rr.com> Message-ID: <51D60F8A.7070304@rahul.net> On 07/04/2013 04:02 PM, david wrote: > Let me see if I can scroll lock the display long enough to note down > exactly what it reports. OK, here goes! > Tangentially, I've found that a camcorder (or similar) is quite handy in this situation. /ken From xiphmont at gmail.com Fri Jul 5 02:14:48 2013 From: xiphmont at gmail.com (Monty Montgomery) Date: Thu, 4 Jul 2013 22:14:48 -0400 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: > A mix is purely an auditory medium, right? So, (and I'll play dumb > here) why do we need to visualize a mix? Eyes are better at some things than ears. A regular example from some audiophile equipment forums... I won't name names, but on one particular forum 'Just listen!' is used as a panacea to dismiss any kind of scientific or objective discussion of equipment. The obvious response is, "Are you suggesting [Company name] take the oscilloscopes, spectrum analyzers, multimeters and computers away from their design engineers?" Visualization can be quite useful, even in audio. Cheers, Monty From malnourite at gmail.com Fri Jul 5 04:36:01 2013 From: malnourite at gmail.com (J. Liles) Date: Thu, 4 Jul 2013 21:36:01 -0700 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: On Thu, Jul 4, 2013 at 7:14 PM, Monty Montgomery wrote: > > A mix is purely an auditory medium, right? So, (and I'll play dumb > > here) why do we need to visualize a mix? > > Eyes are better at some things than ears. > > A regular example from some audiophile equipment forums... I won't > name names, but on one particular forum 'Just listen!' is used as a > panacea to dismiss any kind of scientific or objective discussion of > equipment. > > The obvious response is, "Are you suggesting [Company name] take the > oscilloscopes, spectrum analyzers, multimeters and computers away from > their design engineers?" > > Visualization can be quite useful, even in audio. > > Cheers, > Monty > Let's not forget the distinction between visualization and measurement. The number of people (and we're talking about musicians/producers here) looking at visualizations of audio and the number actually using the visualization to measure something meaningful are, I imagine, very different. Measurement is obviously invaluable when designing DSP algorithms etc. It is not necessary *at all* in order to make music. Peak/clipping indicators are pretty handy, but after that the returns diminish rapidly. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 5 06:20:54 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 05 Jul 2013 08:20:54 +0200 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: <1373005254.1022.12.camel@archlinux> Visualisation is helpful for troubleshooting, if e.g. something does sound distorted, then taking a look at a meterbridge could be timesaving. Phase correlation does ensure that the phases are ok for airplay. However, visualisation isn't needed, but could be helpful. For analog recordings measurements at least are needed for bias calibration. Regards, Ralf From clemens at ladisch.de Fri Jul 5 06:32:40 2013 From: clemens at ladisch.de (Clemens Ladisch) Date: Fri, 05 Jul 2013 08:32:40 +0200 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D5FF0D.9030507@hawaii.rr.com> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> <51D55BCE.9040904@ladisch.de> <51D5FF0D.9030507@hawaii.rr.com> Message-ID: <51D66888.4090206@ladisch.de> david wrote: > On 07/04/2013 01:26 AM, Clemens Ladisch wrote: >> david wrote: >>> According to syslog, it's finds and sets up both the USB device and >>> the Intel snd_intel8x0 just fine. No errors there. >> >> Do you have the device nodes in /dev/snd/? > > The only two showing are seq and timer. Please show all the syslog messages. Regards, Clemens From paul at linuxaudiosystems.com Fri Jul 5 07:33:31 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 5 Jul 2013 03:33:31 -0400 Subject: [LAU] eyes verses ears In-Reply-To: References: Message-ID: I tend to view "visualization" mostly as "on-screen memory" - it is a way to avoid the user having to *actually* remember the state of (potentially) thousands of different controls, by presenting them to him/her in a relatively fast-lookup sort of way. did i solo that? what was the gain level of that ? what processing am i doing with that signal? etc. etc. there is nothing inherent to the task of recording and editing music that requires a visual presentation, but doing without one when working with anything but the simplest possible setup requires the user to constantly remember a large amount of stuff - not always easy to do. mixing consoles, besides actually doing signal processing, also serve this purpose even though they do not (generally) provide any visualization of waveform data or present a timeline. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rustys.lists at gmail.com Fri Jul 5 07:44:03 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Fri, 5 Jul 2013 00:44:03 -0700 Subject: [LAU] eyes verses ears In-Reply-To: <1373005254.1022.12.camel@archlinux> References: <1373005254.1022.12.camel@archlinux> Message-ID: Yes Ralf and J. your distinction between measuring and just visualizing makes lots of sense to me. And Julien, I think you are basically saying the same thing. A graphical tool may make it quicker and easier to isolate problems and take measurements to improve audio quality. Monty, I'm going to play a little devil's advocate here. I wouldn't say that anyone needs to throw away their measurement tools when engineering equipment for best performance, but I have heard of blind listening comparisons where audiofiles couldn't reliably tell the difference between two given studio amplifiers, or between given sets of speaker cables. I recognize that "just listen" may be a trite way of dismissing valid points, but isn't listening the primary goal here? But I do agree with you that if using a visual representation helps more quickly and accurately improve the auditory experience, then that's a good thing. Interesting stuff Rusty On 7/4/13, Ralf Mardorf wrote: > Visualisation is helpful for troubleshooting, if e.g. something does > sound distorted, then taking a look at a meterbridge could be > timesaving. > Phase correlation does ensure that the phases are ok for airplay. > However, visualisation isn't needed, but could be helpful. > > For analog recordings measurements at least are needed for bias > calibration. > > Regards, > Ralf > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > From gnome at hawaii.rr.com Fri Jul 5 09:00:41 2013 From: gnome at hawaii.rr.com (david) Date: Thu, 04 Jul 2013 23:00:41 -1000 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D60F8A.7070304@rahul.net> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> <51D55BCE.9040904@ladisch.de> <51D5FF0D.9030507@hawaii.rr.com> <51D60F8A.7070304@rahul.net> Message-ID: <51D68B39.8040902@hawaii.rr.com> On 07/04/2013 02:12 PM, Ken Dawson wrote: > On 07/04/2013 04:02 PM, david wrote: > >> Let me see if I can scroll lock the display long enough to note down >> exactly what it reports. OK, here goes! > > Tangentially, I've found that a camcorder (or similar) is quite handy in > this situation. Fortunately for me, scroll lock stopped things so I could write it down. I don't have anything that can shoot video. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From gabbe.nord at gmail.com Fri Jul 5 09:38:06 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Fri, 5 Jul 2013 11:38:06 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Julien: :D Very well worth all the trouble. I'm speechless, thank you for the great feedback! I will try and respond to it: * I actually have sidechain compression on more or less all tracks of the album, in various forms. I wouldn't call it overcompression per say, as it's just acting as a tremolo, but I do compress a fair bit too. For the pumping effect, I generally compress a little of almost everything (even cymbals and crashes), which I think helps a little with the rhythm in the song. As for generally compressing as in "master compressing" (or what to call it), the electronical songs especially I compress a lot, as that's how I like them personally (thick and in your face kind of). I do however realize that that's not what everyone likes. I have however "grown out of" a lot of overcompression, at least if I can say so myself, as I no longer push my mixes further just for the loudness. Who knows, maybe I'll dare mixing even more dynamical next time around! ;) * Ha! I was hoping you'd recognize brushing your teeth in the middle of my album! ;) Thank you for the great samples, I put a link to them together with a thank you on my website (http://www.zthmusic.se/Ordinary_Day_Montageif you haven't seen) * Full Moon: I tend to agree with you. It ended up being less interesting/more "chilled out"-than I intended. Next time it'll be more interesting, I promise! ;) And hahaha, sk?l! L?tt?l ?r g?tt! * Right on the money there with the german in the lecture hall. I liked the ambient sample so much due to the clarity and stereo qualities of it, that I just had to incorporate it to the name somehow. * Interesting that Kojan actually exists! To make a long story short, koja means something like "tree house" or "pillow fort" (not entirely accurate, but still) in Swedish, and it's something I've come to call my "studio" (read: corner of my apartment where I record music) just because I've put up so many blankets for the acoustics (dunno if it helps, but it's fun) that it looks like a koja. So that's why that got named like it did! * With whitenoise, I'm just a sucker for it. That's one of the elements I love the most about modern electronic music. I also love ambient sounds (wind and waves in particular), and as it somehow sounds reminiscent of wind and waves at the same time, it's something I really like and almost always end up using. Again, thank you very much for this Julien! It's always awesome to read your feedback and thoughts, and I learn a lot from your observations that I hopefully can improve on 'til next time. Speaking of next time, my current plan is to make sure that the next kind of thing like this I do is 100% with vocals on every track, in one form or another. Feels like it's time! ;) Thanks again! On Fri, Jul 5, 2013 at 12:11 AM, Julien Claassen wrote: > Hello Gabbe! > Before I begin to say any word about the music you made, I will say: > thank you for all the troubles you got up to to give me a chance to listen > to it. This is very kind of you and - be assured - very much appreciated! > As a whole, I very much agree with Dave's vote of the clear sound and > good mix. The mixes have power, sound well balanced to me. What's even > better, is that they fall on willing ears. In recent weeks I've turned more > and more towards electronic music. > the intro is a good introduction to the sonic world of your album. > Slowly taking me in, preparing for the Skandinavian sound and feel. The > width and openness. Only to surprise me later on. there is that > Skandinavian feel poking through, but not as obvious as before. I like the > bassdrum of the intro. It appears later on again. It has bite and pumps a > little, not in a compression artefact way. It's somewhere between an edited > acoustic bassdrum and something purely electronic. It might even fill the > four-to-the-floor niche. > It is alive is the nice four-to-the-floor piece. And it already shows > something, which characterises the whole album. The tracks flow seemlessly > from one into the other. Not only by ambient sounds, but by concept. If it > had been me, I'd have mixed the snare louder. But that way it keeps the > attention, while still deliviering. I can't completely let go with that, > because, there is that small bit, which seems so uncharacteristic. I also > love the somehow oneiric harmonies. Tending towards minor, yet rarely pure > minor chords. While having a touch of 90s nostalgia about it, the mix > sounds very much like the current decade. Do you use "overcompression" to > get this slightly pumping effect? I noticed that throughout the album and > had been wondering. If so: how do you compress? The whole song or only > certain groups of instruments? - Again the bassdrum is nice, crisp and very > well defined. Ha! The blend to Still is lovely. I recognise the sound and I > must admit, that I never dreamt it could be used in such a place. Well sed. > Still: I think here the bassdrum is a little too loud in the mix, but > not so loud as to bannish the thought of enjoyment. :-) And so we have > turned to a more hiphop styled track. And it still keeps in the spirit of > the earlier tracks. As I said: it's a sonic world, this album. Listening to > it again, I feel, that so far, it appears more like a change in arrangement > and style than in harmony. It's bordering on my threshhold of boredom. If > it wasn't for the very nice mixes and well executed styles and the morphing > of one into the other... but there is that and it holds me. You know, that > my threshhold of boredom is quite low. Since I'm still more of a prog and > baroque listener than electronica. > Come and play; why not. I'm game. :-) This sets a much lighter tone. I > love the slight touch of breakbeat in this piece. Also the main instrument > is interesting. the way the piece changes from almost elevator music into > almost dubstep is great. The way the movement picks up and the arrangement > fills up from almost meditative spheric guitar to the full beat. Teh > effects you used, the well-placed delay and how you managed to use that to > instantly isolate the main "lead" and the rhythm in places. Well done. good > trick! > Full Moon baffled me at first. Perhaps because I approached it with > preformed notions, of what it should have been. That way it's the piece, > that I still perceive as the weakest piece on this album. It sounds to > common place for me. To worldly. The introduction and the "bridge" are the > best bits. They talk to me of a full moon and the quiet night. The rest is > more of the beach party next door with its drinking, barbecuing crowd. At > least it's a Swedish party and not one of those Mallorca crimes, that they > probably call "a beach happening" or "afterwork holiday". :-) Get out the > "let oel" and let's say "skol" (sorry can't spell it and didn't find it > quickly :-) ). > I wonder, what you thought, when you called the next composition > "Touchdown Germany"? Was it the ambient sound from the lecture hall at my > uni? Or is it the plump beat? That beat, which makes me think of Flaubert, > who said something about music of the heavens, that we hear, but only > produce something, which sounds like it's being played on rusting tin > drums, written for crippled bears. :-) It has a very raw edge this piece, > while keeping with the flow of the dreamy melodies. > the Circus reminds me even more of dubstep. Not really, but the > stuttering rhythm and this change from third notes in a quarter beat. Only > your version of it is like Debusy's sweet juicy apple eaten with knife and > fork in comparison to Gershwin. Neither is wrong or right, but one is more > tamed than the other. Rhythmic acrobatics in pensive music. A circus > indeed. :-) > Have you ever been to Korea? If my quick look at google didn't lead me > astray completely, it should be somewhere in South Korea. Whereever Kojan > is, we have returned to the good, old idiom of hiphop with the drumkit from > the first track. If I were to analyse this piece, I'd say the instrument > reminiscent of an electric piano in the middle has a slight touch of typica > Asian harmony and the sawtoothy synth almost gives an impression of > chiptunes synthesis. With the impressions we get, I might suggest, that > this is to conjure up an idea of all those game consoles, that the > youngsters over there love so much or at least use so much. One of my old > classmates would at least agree with the interest in game consoles, if not > wioth my implications on this piece. :-) In a way it's a good song to lead > me out. It seems so much more worldly. The applause, the crowd, the rest of > the world, that stands so far removed from the images of moonlit nights - > parties or no parties - and the still, windy atmosphere of the album. You > worked a great deal with white noise, which gives one the feeling of wind > in the dunes. It's a sound I heard a lot in radioplays and this white noise > patch - or patches - are close to that sound. Somewhere between wind and > waves or the mixture of the two? > All in all, it's a good album and I will be sure to listen to it again. > Perhaps not in one go, as I did today, but now I've taken in teh > atmosphere, I can quickly travel back there. It's a good piece of work > Always slightly dreamy, dark, depressive and still with a consoling > lightness behind the pregnant beats and occasional distorted synths. A good > balance between the single mindedness of centre beats and basses and > parnoramic harmonies and effects. I like it. And I hope, that you will get > a whole load of feedback and a couple of new listeners. > What will you give us next time? Perhaps once more something with the > occasional lyrics? Some Swedish hiphop or some English song of longing? Or > something different? > Warm regards and thanks for sharing > Julien > > ------------------------------**---------- > http://juliencoder.de/nama/**music.html > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gabbe.nord at gmail.com Fri Jul 5 09:42:13 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Fri, 5 Jul 2013 11:42:13 +0200 Subject: [LAU] eyes verses ears In-Reply-To: References: <1373005254.1022.12.camel@archlinux> Message-ID: Hello Rusty! As you referred to me in the original post, I just figured I'd briefly explain what I mainly use the analyzer for. It's as simple as me not having a subwoofer or a reliable set of headphones for the bass. Neither my monitors nor my studio heaphones have any noteworthy bass-response under 70hz or so, which is where most of my bass is usually. I have a set of headphones that are of the "gamer-kind", which I've learned to use for bass. So, what I do when mixing usually, is that I use the analyzer to check the bass. I've learned through my time mixing my things just about how I want it to sound, or at least the general idea. It's in no way any substitute for the final mix and master (where I find enough systems to listen to, and reliably mix the bass), but it's an easy way for me to keep it in control while I'm doing the heavy work. That's my use-case at least. :) On Fri, Jul 5, 2013 at 9:44 AM, Rusty Perez wrote: > Yes Ralf and J. your distinction between measuring and just > visualizing makes lots of sense to me. And Julien, I think you are > basically saying the same thing. A graphical tool may make it quicker > and easier to isolate problems and take measurements to improve audio > quality. > > Monty, I'm going to play a little devil's advocate here. > I wouldn't say that anyone needs to throw away their measurement tools > when engineering equipment for best performance, but I have heard of > blind listening comparisons where audiofiles couldn't reliably tell > the difference between two given studio amplifiers, or between given > sets of speaker cables. > I recognize that "just listen" may be a trite way of dismissing valid > points, but isn't listening the primary goal here? > But I do agree with you that if using a visual representation helps > more quickly and accurately improve the auditory experience, then > that's a good thing. > > Interesting stuff > Rusty > > > > On 7/4/13, Ralf Mardorf wrote: > > Visualisation is helpful for troubleshooting, if e.g. something does > > sound distorted, then taking a look at a meterbridge could be > > timesaving. > > Phase correlation does ensure that the phases are ok for airplay. > > However, visualisation isn't needed, but could be helpful. > > > > For analog recordings measurements at least are needed for bias > > calibration. > > > > Regards, > > Ralf > > > > _______________________________________________ > > 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 julien at mail.upb.de Fri Jul 5 09:57:17 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 5 Jul 2013 11:57:17 +0200 (CEST) Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Hello Gabbe! You're welcome and I'm very sure you put much more work into this than I did. :-) thank you for the explanations. It's a pitty, that sidechain compression doesn't really work with Ecasound. It appears to be a valuable tool, when you're doing electronic music. Kepp 'em coming! Julien ---------------------------------------- http://juliencoder.de/nama/music.html From gnome at hawaii.rr.com Fri Jul 5 10:05:42 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 00:05:42 -1000 Subject: [LAU] Updated system, now refuses to load snd-usb-audio and module for onboard sound In-Reply-To: <51D66888.4090206@ladisch.de> References: <51D51514.1030007@hawaii.rr.com> <51D531E1.7000501@ladisch.de> <51D544CF.80509@hawaii.rr.com> <51D55BCE.9040904@ladisch.de> <51D5FF0D.9030507@hawaii.rr.com> <51D66888.4090206@ladisch.de> Message-ID: <51D69A76.8030106@hawaii.rr.com> On 07/04/2013 08:32 PM, Clemens Ladisch wrote: > david wrote: >> On 07/04/2013 01:26 AM, Clemens Ladisch wrote: >>> david wrote: >>>> According to syslog, it's finds and sets up both the USB device and >>>> the Intel snd_intel8x0 just fine. No errors there. >>> >>> Do you have the device nodes in /dev/snd/? >> >> The only two showing are seq and timer. > > Please show all the syslog messages. See other message with that. For reference, I just shutdown and powered back on, and now aplay -l lists the onboard sound but not the USB card. The log shows USB card being found and setup. At the same time, my external USB drive is showing a solid red (reading/writing) light but the system log reports it not responding. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From gabbe.nord at gmail.com Fri Jul 5 10:20:17 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Fri, 5 Jul 2013 12:20:17 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: Thinking about it, I think you can achieve more or less the same result by using a bus with a tremolo plugin on. Maybe worth checking out if sidechaining per say does not work! Cheers, Gabriel/zth On Fri, Jul 5, 2013 at 11:57 AM, Julien Claassen wrote: > Hello Gabbe! > You're welcome and I'm very sure you put much more work into this than I > did. :-) > thank you for the explanations. It's a pitty, that sidechain compression > doesn't really work with Ecasound. It appears to be a valuable tool, when > you're doing electronic music. > Kepp 'em coming! > Julien > > ------------------------------**---------- > http://juliencoder.de/nama/**music.html > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Fri Jul 5 10:33:27 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Fri, 5 Jul 2013 12:33:27 +0200 Subject: [LAU] carla & OSC Message-ID: Hallo, Carla is a beautifil piece of software - thank you for that. I am not able to control params with OSC, only with MIDI msg I use pd and my patch looks like: [connect 127.0.0.1 15609( | | [/Carla/1/set_volume 0( | / | [packOSC] |/ [tcpsend] I suppose, it should set volume of first plugin to zero, but it does not. what I am doing wrong ? thank you. fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From arve.barsnes at gmail.com Fri Jul 5 12:29:09 2013 From: arve.barsnes at gmail.com (Arve Barsnes) Date: Fri, 5 Jul 2013 14:29:09 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: On 3 July 2013 17:05, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and online! > :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a > bit more electronic than last one. > > Sounds absolutely wonderful, a great followup to your previous album! Arve -------------- next part -------------- An HTML attachment was scrubbed... URL: From diego.simak at gmail.com Fri Jul 5 13:45:02 2013 From: diego.simak at gmail.com (Diego Simak) Date: Fri, 5 Jul 2013 10:45:02 -0300 Subject: [LAU] carla & OSC In-Reply-To: References: Message-ID: 2013/7/5 Fero Kiraly : > Hallo, > > Carla is a beautifil piece of software - thank you for that. > I am not able to control params with OSC, only with MIDI msg > > I use pd and my patch looks like: > > [connect 127.0.0.1 15609( > | > | [/Carla/1/set_volume 0( > | / > | [packOSC] > |/ > [tcpsend] > > I suppose, it should set volume of first plugin to zero, but it does not. > what I am doing wrong ? > > thank you. > > fero > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > I'm not a PD expert but seems that you need to add 'send' in the OSC message, like this: [send /Carla/1/set_volume 0( is 15609 a TCP or UDP port? I can see that Carla now does support TCP but please double check which type of connection are you using. If it is a UDP connection type change tcpsend with udpsend. just for the reference: http://en.flossmanuals.net/pure-data/ch065_osc/ From falktx at gmail.com Fri Jul 5 15:32:44 2013 From: falktx at gmail.com (Filipe Coelho) Date: Fri, 05 Jul 2013 16:32:44 +0100 Subject: [LAU] carla & OSC In-Reply-To: References: Message-ID: <51D6E71C.6050605@gmail.com> On 07/05/2013 11:33 AM, Fero Kiraly wrote: > Hallo, > > Carla is a beautifil piece of software - thank you for that. > I am not able to control params with OSC, only with MIDI msg > > I use pd and my patch looks like: > > [connect 127.0.0.1 15609( > | > | [/Carla/1/set_volume 0( > | / > | [packOSC] > |/ > [tcpsend] > > I suppose, it should set volume of first plugin to zero, but it does not. > what I am doing wrong ? > > thank you. > The first plugin starts at 0, not 1. You should make sure the parameter types match. In this case, set_volume, must be float. You can check the logs tab to see if a message has been ignored. And be sure to use the TCP or UDP address correctly. I know that the "oscsend" tool only works with UDP. For reference, this command works fine here: $ oscsend localhost 19708 /Carla/0/set_volume f 0.8 From guido-scholz at gmx.net Fri Jul 5 15:41:13 2013 From: guido-scholz at gmx.net (Guido Scholz) Date: Fri, 5 Jul 2013 17:41:13 +0200 Subject: [LAU] [LAD] Updates In-Reply-To: <20130702214925.GB6346@linuxaudio.org> References: <20130702214925.GB6346@linuxaudio.org> Message-ID: <20130705154113.GA4920@traun.gscholz.bayernline.de> Am Tue, 02. Jul 2013 um 21:49:25 +0000 schrieb Fons Adriaensen: > Hello all, Hello fons, > Note to AMS devs: zita-alsa-pcmi is a near drop-in replacement > for clalsadrv-2.0.0: > * Change the library name in the build files > * s/#include/#include/ > * s/Alsa_driver/Alsa_pcmi/ > * s/->stat()/->state()/ I would not recommend this to ams users. Use the cvs version instead. Guido -- http://www.wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From fons at linuxaudio.org Fri Jul 5 15:50:12 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 5 Jul 2013 15:50:12 +0000 Subject: [LAU] [LAD] Updates In-Reply-To: <20130705154113.GA4920@traun.gscholz.bayernline.de> References: <20130702214925.GB6346@linuxaudio.org> <20130705154113.GA4920@traun.gscholz.bayernline.de> Message-ID: <20130705155012.GA24644@linuxaudio.org> On Fri, Jul 05, 2013 at 05:41:13PM +0200, Guido Scholz wrote: > > Note to AMS devs: zita-alsa-pcmi is a near drop-in replacement > > for clalsadrv-2.0.0: > > > * Change the library name in the build files > > * s/#include/#include/ > > * s/Alsa_driver/Alsa_pcmi/ > > * s/->stat()/->state()/ > > I would not recommend this to ams users. Use the cvs version instead. It was meant for the devs ! 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 leoave at gmail.com Fri Jul 5 16:29:58 2013 From: leoave at gmail.com (Leonardo Palomares) Date: Fri, 5 Jul 2013 09:29:58 -0700 Subject: [LAU] Off topic - Bluegrass for boys! Message-ID: Link: http://on.ted.com/BanjoBoys Not all kids are full time into computer games and facebook. This video will give you new hope on the next generation. I know David Phillips will enjoy this.... Leo -------------- next part -------------- An HTML attachment was scrubbed... URL: From guido-scholz at gmx.net Fri Jul 5 16:35:19 2013 From: guido-scholz at gmx.net (Guido Scholz) Date: Fri, 5 Jul 2013 18:35:19 +0200 Subject: [LAU] [LAD] Updates In-Reply-To: <20130705155012.GA24644@linuxaudio.org> References: <20130702214925.GB6346@linuxaudio.org> <20130705154113.GA4920@traun.gscholz.bayernline.de> <20130705155012.GA24644@linuxaudio.org> Message-ID: <20130705163519.GA13011@traun.gscholz.bayernline.de> Am Fri, 05. Jul 2013 um 15:50:12 +0000 schrieb Fons Adriaensen: Hi Fons, > > I would not recommend this to ams users. Use the cvs version instead. > It was meant for the devs ! OK, sorry. I would also not recommend this to developers. Please use the ams cvs version instead. Guido -- http://www.wie-im-flug.net/ http://www.lug-burghausen.org/ -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From bob at mellowood.ca Fri Jul 5 17:23:51 2013 From: bob at mellowood.ca (Bob van der Poel) Date: Fri, 5 Jul 2013 10:23:51 -0700 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: References: Message-ID: Sort of off topic ... but is anyone getting TED videos to play on Linux with Firefox? On Fri, Jul 5, 2013 at 9:29 AM, Leonardo Palomares wrote: > Link: > http://on.ted.com/BanjoBoys > > Not all kids are full time into computer games and facebook. > > This video will give you new hope on the next generation. > > I know David Phillips will enjoy this.... > > Leo > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob at mellowood.ca WWW: http://www.mellowood.ca From david.santamauro at gmail.com Fri Jul 5 17:33:09 2013 From: david.santamauro at gmail.com (David Santamauro) Date: Fri, 05 Jul 2013 13:33:09 -0400 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: References: Message-ID: <51D70355.4060801@gmail.com> Great talent, and yes, watching now on Firefox ... On 07/05/2013 01:23 PM, Bob van der Poel wrote: > Sort of off topic ... but is anyone getting TED videos to play on > Linux with Firefox? > > On Fri, Jul 5, 2013 at 9:29 AM, Leonardo Palomares wrote: >> Link: >> http://on.ted.com/BanjoBoys >> >> Not all kids are full time into computer games and facebook. >> >> This video will give you new hope on the next generation. >> >> I know David Phillips will enjoy this.... >> >> Leo >> >> >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> > > > From gnome at hawaii.rr.com Fri Jul 5 18:38:29 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 08:38:29 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D70355.4060801@gmail.com> References: <51D70355.4060801@gmail.com> Message-ID: <51D712A5.9090109@hawaii.rr.com> Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: " It appears that we are unable to play this video in your browser for one of the following reasons: you have JavaScript disabled your device does not support HTML5 video you have an old version of the Adobe Flash Player. Download the latest Flash player to view this video." IIRC, HMTL5 video support is very dependent on video drivers: one great disadvantage compared to good old proprietary Flash. Well, downloaded the MP4 and tried to play it, and it won't play. Rather, it plays, but there's no sound and no video. Deleted it and tried again, and now mplayer plays it. They're pretty damn good. On 07/05/2013 07:33 AM, David Santamauro wrote: > Great talent, and yes, watching now on Firefox ... > > On 07/05/2013 01:23 PM, Bob van der Poel wrote: >> Sort of off topic ... but is anyone getting TED videos to play on >> Linux with Firefox? >> >> On Fri, Jul 5, 2013 at 9:29 AM, Leonardo Palomares >> wrote: >>> Link: >>> http://on.ted.com/BanjoBoys >>> >>> Not all kids are full time into computer games and facebook. >>> >>> This video will give you new hope on the next generation. >>> >>> I know David Phillips will enjoy this.... >>> >>> Leo -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From althompson58 at gmail.com Fri Jul 5 19:05:46 2013 From: althompson58 at gmail.com (Al Thompson) Date: Fri, 05 Jul 2013 15:05:46 -0400 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D712A5.9090109@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> Message-ID: <51D7190A.4090800@gmail.com> On 07/05/2013 02:38 PM, david wrote: > Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: > " It appears that we are unable to play this video in your browser for > one of the following reasons: > > you have JavaScript disabled > your device does not support HTML5 video > you have an old version of the Adobe Flash Player. Download the > latest Flash player to view this video." > > IIRC, HMTL5 video support is very dependent on video drivers: one > great disadvantage compared to good old proprietary Flash. > > Well, downloaded the MP4 and tried to play it, and it won't play. > Rather, it plays, but there's no sound and no video. Deleted it and > tried again, and now mplayer plays it. > Plays fine here on Fedora 14 (64 bit) and Firefox 22.0. -- --- My bands, CD projects, music, news, and pictures: http://www.lateralforce.com My blog, with commentary on a variety of things, including audio, mixing, equipment, etc, is at: http://audioandmore.wordpress.com Staat hei?t das k?lteste aller kalten Ungeheuer. Kalt l?gt es auch; und diese L?ge kriecht aus seinem Munde: 'Ich, der Staat, bin das Volk.' - [Friedrich Nietzsche] From bob at mellowood.ca Fri Jul 5 19:12:06 2013 From: bob at mellowood.ca (Bob van der Poel) Date: Fri, 5 Jul 2013 12:12:06 -0700 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D712A5.9090109@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> Message-ID: After thinking about this and seeing a few messages about it working on FF ... I did a few tests. 1. On my wife's computer running xfce it works just fine. So, it's not FF 2. Logged in as "Guest" on my box. This is Ubuntu 12.04. As guest it works just fine. So, it's not FF and it's not the flash plug in either since I'd be using the same version. 3. Back to my home and I tried some things with FF. Ran in "safe mode" and then tried using a new profile. Neither works. So, darned if I have any more ideas ... and here I thought it was something silly TED was doing :) On Fri, Jul 5, 2013 at 11:38 AM, david wrote: > Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: > " It appears that we are unable to play this video in your browser for one > of the following reasons: > > you have JavaScript disabled > your device does not support HTML5 video > you have an old version of the Adobe Flash Player. Download the latest > Flash player to view this video." > > IIRC, HMTL5 video support is very dependent on video drivers: one great > disadvantage compared to good old proprietary Flash. > > Well, downloaded the MP4 and tried to play it, and it won't play. Rather, it > plays, but there's no sound and no video. Deleted it and tried again, and > now mplayer plays it. > > They're pretty damn good. > > On 07/05/2013 07:33 AM, David Santamauro wrote: > >> Great talent, and yes, watching now on Firefox ... >> >> On 07/05/2013 01:23 PM, Bob van der Poel wrote: >>> >>> Sort of off topic ... but is anyone getting TED videos to play on >>> Linux with Firefox? >>> >>> On Fri, Jul 5, 2013 at 9:29 AM, Leonardo Palomares >>> wrote: >>>> >>>> Link: >>>> http://on.ted.com/BanjoBoys >>>> >>>> Not all kids are full time into computer games and facebook. >>>> >>>> This video will give you new hope on the next generation. >>>> >>>> I know David Phillips will enjoy this.... >>>> >>>> Leo > > > > -- > David > gnome at hawaii.rr.com > authenticity, honesty, community > http://dancingtreefrog.com > http://clanjones.org/david/ > http://dancing-treefrog.deviantart.com/ > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob at mellowood.ca WWW: http://www.mellowood.ca From gnome at hawaii.rr.com Fri Jul 5 19:35:33 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 09:35:33 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> Message-ID: <51D72005.7030207@hawaii.rr.com> Running XFCE here, it doesn't work. My wife's puter has much older version of FF (thanks, Linux Mint!) with no HTML5 support as near as I can tell. When she watches vids, they're all in Flash format. I don't think it's FF per se, it's the video driver being used by the particular machine. Some support HTML5, some don't. Even true (I understand) on Windows, but to a much lesser extent. On 07/05/2013 09:12 AM, Bob van der Poel wrote: > After thinking about this and seeing a few messages about it working > on FF ... I did a few tests. > > 1. On my wife's computer running xfce it works just fine. So, it's not FF > > 2. Logged in as "Guest" on my box. This is Ubuntu 12.04. As guest it > works just fine. So, it's not FF and it's not the flash plug in either > since I'd be using the same version. > > 3. Back to my home and I tried some things with FF. Ran in "safe mode" > and then tried using a new profile. Neither works. > > So, darned if I have any more ideas ... and here I thought it was > something silly TED was doing :) > > On Fri, Jul 5, 2013 at 11:38 AM, david wrote: >> Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: >> " It appears that we are unable to play this video in your browser for one >> of the following reasons: >> >> you have JavaScript disabled >> your device does not support HTML5 video >> you have an old version of the Adobe Flash Player. Download the latest >> Flash player to view this video." >> >> IIRC, HMTL5 video support is very dependent on video drivers: one great >> disadvantage compared to good old proprietary Flash. >> >> Well, downloaded the MP4 and tried to play it, and it won't play. Rather, it >> plays, but there's no sound and no video. Deleted it and tried again, and >> now mplayer plays it. >> >> They're pretty damn good. >> >> On 07/05/2013 07:33 AM, David Santamauro wrote: >> >>> Great talent, and yes, watching now on Firefox ... >>> >>> On 07/05/2013 01:23 PM, Bob van der Poel wrote: >>>> >>>> Sort of off topic ... but is anyone getting TED videos to play on >>>> Linux with Firefox? >>>> >>>> On Fri, Jul 5, 2013 at 9:29 AM, Leonardo Palomares >>>> wrote: >>>>> >>>>> Link: >>>>> http://on.ted.com/BanjoBoys >>>>> >>>>> Not all kids are full time into computer games and facebook. >>>>> >>>>> This video will give you new hope on the next generation. >>>>> >>>>> I know David Phillips will enjoy this.... >>>>> >>>>> Leo -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From gnome at hawaii.rr.com Fri Jul 5 19:38:09 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 09:38:09 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D7190A.4090800@gmail.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D7190A.4090800@gmail.com> Message-ID: <51D720A1.3090602@hawaii.rr.com> On 07/05/2013 09:05 AM, Al Thompson wrote: > On 07/05/2013 02:38 PM, david wrote: >> Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: >> " It appears that we are unable to play this video in your browser for >> one of the following reasons: >> >> you have JavaScript disabled >> your device does not support HTML5 video >> you have an old version of the Adobe Flash Player. Download the >> latest Flash player to view this video." >> >> IIRC, HMTL5 video support is very dependent on video drivers: one >> great disadvantage compared to good old proprietary Flash. >> >> Well, downloaded the MP4 and tried to play it, and it won't play. >> Rather, it plays, but there's no sound and no video. Deleted it and >> tried again, and now mplayer plays it. > > Plays fine here on Fedora 14 (64 bit) and Firefox 22.0. What video driver are you using? My laptop has Intel 855GM video. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From dlphillips at woh.rr.com Fri Jul 5 20:09:32 2013 From: dlphillips at woh.rr.com (Dave Phillips) Date: Fri, 05 Jul 2013 16:09:32 -0400 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: References: Message-ID: <51D727FC.1090901@woh.rr.com> On 07/05/2013 12:29 PM, Leonardo Palomares wrote: > Link: > http://on.ted.com/BanjoBoys > > Not all kids are full time into computer games and facebook. > > This video will give you new hope on the next generation. > > I know David Phillips will enjoy this.... > > Leo > Thanks for the link, Leo, it is enjoyable. Those are very talented kids. For another look at talented kids performing there's this famous vid : http://www.youtube.com/watch?v=gSedE5sU3uc Weirdness plus. Best, dp > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Fri Jul 5 20:27:28 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 10:27:28 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D727FC.1090901@woh.rr.com> References: <51D727FC.1090901@woh.rr.com> Message-ID: <51D72C30.6000301@hawaii.rr.com> On 07/05/2013 10:09 AM, Dave Phillips wrote: > On 07/05/2013 12:29 PM, Leonardo Palomares wrote: >> Link: >> http://on.ted.com/BanjoBoys >> >> Not all kids are full time into computer games and facebook. >> >> This video will give you new hope on the next generation. >> >> I know David Phillips will enjoy this.... >> >> Leo >> > > Thanks for the link, Leo, it is enjoyable. Those are very talented kids. > > For another look at talented kids performing there's this famous vid : > > http://www.youtube.com/watch?v=gSedE5sU3uc > > Weirdness plus. > > Best, > > dp Hmm, so North Korea is now cloning children? Or it pushes kids into showbiz early like some idiot parents in America do? Definitely creepy. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From bob at mellowood.ca Fri Jul 5 20:30:20 2013 From: bob at mellowood.ca (Bob van der Poel) Date: Fri, 5 Jul 2013 13:30:20 -0700 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D72005.7030207@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D72005.7030207@hawaii.rr.com> Message-ID: > I don't think it's FF per se, it's the video driver being used by the > particular machine. Some support HTML5, some don't. Even true (I understand) > on Windows, but to a much lesser extent. Don't think it is the video driver. On my computer I'm running the nvidia binary. Logging in as Guest I can watch; using my account I can't. Same computer. Same video. Odd thing about this is that there is no directory for the Guest account ... not sure if that would make a difference. Oh, just created a new account on my computer. It has a small firefox directory, etc. And TED works. But, try as I might it will not work from my home directory. I do have a .adobe directory in my home. Just deleted that and restarted FF. Nope, no ted video. Mind you, just about any other video stuff I watch works fine. All very strange! So, it's not xfce. It's not the video driver. Not Firefox. And, not a profile issue (I created a new profile in my ff and got the same error). -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob at mellowood.ca WWW: http://www.mellowood.ca From gnome at hawaii.rr.com Fri Jul 5 21:32:09 2013 From: gnome at hawaii.rr.com (david) Date: Fri, 05 Jul 2013 11:32:09 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D72005.7030207@hawaii.rr.com> Message-ID: <51D73B59.5060003@hawaii.rr.com> On 07/05/2013 10:30 AM, Bob van der Poel wrote: >> I don't think it's FF per se, it's the video driver being used by the >> particular machine. Some support HTML5, some don't. Even true (I understand) >> on Windows, but to a much lesser extent. > > Don't think it is the video driver. On my computer I'm running the > nvidia binary. Logging in as Guest I can watch; using my account I > can't. Same computer. Same video. Odd thing about this is that there > is no directory for the Guest account ... not sure if that would make > a difference. Oh, just created a new account on my computer. It has a > small firefox directory, etc. And TED works. But, try as I might it > will not work from my home directory. > > I do have a .adobe directory in my home. Just deleted that and > restarted FF. Nope, no ted video. Mind you, just about any other video > stuff I watch works fine. > > All very strange! > > So, it's not xfce. It's not the video driver. Not Firefox. And, not a > profile issue (I created a new profile in my ff and got the same > error). Well, YouTube works here, but not TED. I have an .adobe folder, and it contains subfolders for Acrobat and Flash. I suspect that if you had it there with a Flash folder, your machine has Flash installed on it, and maybe FF is using Flash instead of HTML5? Strangely enough, audio is now working for me in Firefox, after years of it not working. Possibly has something to do with the fact that the system is currently refusing to load the Intel audio driver. So when the USB sound card driver is loaded (manually), FF has no choice but to play through the USB card. :-) -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From lsd at wootangent.net Fri Jul 5 23:17:42 2013 From: lsd at wootangent.net (Leigh Dyer) Date: Sat, 06 Jul 2013 09:17:42 +1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D712A5.9090109@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> Message-ID: <51D75416.2020109@wootangent.net> On 6/07/13 4:38 AM, david wrote: > Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: > " It appears that we are unable to play this video in your browser for > one of the following reasons: > > you have JavaScript disabled > your device does not support HTML5 video > you have an old version of the Adobe Flash Player. Download the > latest Flash player to view this video." > > IIRC, HMTL5 video support is very dependent on video drivers: one great > disadvantage compared to good old proprietary Flash. > It shouldn't be dependent on drivers, but not all browsers that support HTML5 video (or audio, for that matter) will play all HTML5 videos. Different browsers support different codecs, and while Chrome, Safari, and IE support proprietary codecs like H.264 and AAC, Firefox supports only free codecs, like Theora, VP8 (aka WebM) and Vorbis. So, while Firefox mostly works with HTML5 video on Youtube (since Google supports WebM there for many videos), most other sites only offer H.264 video via HTML5, which won't work with Firefox. Most of those sites will fall back to using Flash to play such videos on Firefox if it's available, though. Thanks Leigh From althompson58 at gmail.com Fri Jul 5 23:37:50 2013 From: althompson58 at gmail.com (Al Thompson) Date: Fri, 05 Jul 2013 19:37:50 -0400 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D720A1.3090602@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D7190A.4090800@gmail.com> <51D720A1.3090602@hawaii.rr.com> Message-ID: <51D758CE.1060008@gmail.com> On 07/05/2013 03:38 PM, david wrote: > On 07/05/2013 09:05 AM, Al Thompson wrote: >> On 07/05/2013 02:38 PM, david wrote: >>> Doesn't play here on Debian Sid Linux with Firefox 22. Displays >>> message: >>> " It appears that we are unable to play this video in your browser for >>> one of the following reasons: >>> >>> you have JavaScript disabled >>> your device does not support HTML5 video >>> you have an old version of the Adobe Flash Player. Download the >>> latest Flash player to view this video." >>> >>> IIRC, HMTL5 video support is very dependent on video drivers: one >>> great disadvantage compared to good old proprietary Flash. >>> >>> Well, downloaded the MP4 and tried to play it, and it won't play. >>> Rather, it plays, but there's no sound and no video. Deleted it and >>> tried again, and now mplayer plays it. >> >> Plays fine here on Fedora 14 (64 bit) and Firefox 22.0. > > What video driver are you using? My laptop has Intel 855GM video. > i915, but I don't remember why. It's on an HP laptop, and it works fine, so that's all that really matters to me. -- --- My bands, CD projects, music, news, and pictures: http://www.lateralforce.com My blog, with commentary on a variety of things, including audio, mixing, equipment, etc, is at: http://audioandmore.wordpress.com Staat hei?t das k?lteste aller kalten Ungeheuer. Kalt l?gt es auch; und diese L?ge kriecht aus seinem Munde: 'Ich, der Staat, bin das Volk.' - [Friedrich Nietzsche] From pshirkey at boosthardware.com Sat Jul 6 05:46:06 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Sat, 6 Jul 2013 15:46:06 +1000 (EST) Subject: [LAU] Yocto : embedded Linux platform Message-ID: <49876.188.26.254.75.1373089566.squirrel@boosthardware.com> Hi, For those of you who are interested in embedded Linux and Linux Audio/Multimedia you may also be interested in the ongoing development around the Yocto project which seeks to provide a compiler framework and platform for creating bespoke embedded distributions. Heavily sponsored by Intel and a number of other large multinational companies. Also has nothing to do with Android. Just pure Linux without all the other crap. https://www.yoctoproject.org/ Cheers -- Patrick Shirkey Boost Hardware Ltd From pshirkey at boosthardware.com Sat Jul 6 05:57:20 2013 From: pshirkey at boosthardware.com (Patrick Shirkey) Date: Sat, 6 Jul 2013 15:57:20 +1000 (EST) Subject: [LAU] leap motion sdk open Message-ID: <50036.188.26.254.75.1373090240.squirrel@boosthardware.com> Hi, FYI, the Leap Motion SDK and Developer portal has been opened up today for anyone to join. https://developer.leapmotion.com -- Patrick Shirkey Boost Hardware Ltd From dawsonwu at rahul.net Sat Jul 6 06:07:21 2013 From: dawsonwu at rahul.net (Ken Dawson) Date: Fri, 05 Jul 2013 23:07:21 -0700 Subject: [LAU] Yocto : embedded Linux platform In-Reply-To: <49876.188.26.254.75.1373089566.squirrel@boosthardware.com> References: <49876.188.26.254.75.1373089566.squirrel@boosthardware.com> Message-ID: <51D7B419.4000805@rahul.net> On 07/05/2013 10:46 PM, Patrick Shirkey wrote: > Hi, > > For those of you who are interested in embedded Linux and Linux > Audio/Multimedia you may also be interested in the ongoing development > around the Yocto project which seeks to provide a compiler framework and > platform for creating bespoke embedded distributions. Heavily sponsored > by Intel and a number of other large multinational companies. Also has > nothing to do with Android. Just pure Linux without all the other crap. > > https://www.yoctoproject.org/ > ... with another high-quality community supporting it. /ken From julien at mail.upb.de Sat Jul 6 10:51:42 2013 From: julien at mail.upb.de (Julien Claassen) Date: Sat, 6 Jul 2013 12:51:42 +0200 (CEST) Subject: [LAU] Problems with zita-a2j/j2a Message-ID: Hello everyone! I recently started getting trouble from zita-a2j/j2a. It worked like a treat before. I'm trying to integrate my E-MU 1212m into my JACK setup, which mainly uses my Delta 1010 LT. Now I only get: Starting synchronisation. Alsa_pcmi: poll timed out. Repeating over and over again. I tried changing the period size of zita-a2j as well as the period size of JACK. No luck. the MIDI of the E-MU is working, I've tried it. I can't really check the audio by itself without a great hassle, but I assume, that this too should be OK. Any idea, why this might have happened? I didn't start anything CPU-intensive or change anything ese in the system, tht I am aware of. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From fons at linuxaudio.org Sat Jul 6 11:58:34 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 6 Jul 2013 11:58:34 +0000 Subject: [LAU] Problems with zita-a2j/j2a In-Reply-To: References: Message-ID: <20130706115834.GA31535@linuxaudio.org> On Sat, Jul 06, 2013 at 12:51:42PM +0200, Julien Claassen wrote: > I recently started getting trouble from zita-a2j/j2a. It worked > like a treat before. I'm trying to integrate my E-MU 1212m into my > JACK setup, which mainly uses my Delta 1010 LT. Now I only get: > Starting synchronisation. > Alsa_pcmi: poll timed out. > Repeating over and over again. I tried changing the period size of > zita-a2j as well as the period size of JACK. No luck. the MIDI of > the E-MU is working, I've tried it. I can't really check the audio > by itself without a great hassle, but I assume, that this too should > be OK. > Any idea, why this might have happened? I didn't start anything > CPU-intensive or change anything ese in the system, tht I am aware Does the E-MU on its own work with jack ? Also when used in capture-only or playback-only mode ? 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 julien at mail.upb.de Sat Jul 6 14:06:54 2013 From: julien at mail.upb.de (Julien Claassen) Date: Sat, 6 Jul 2013 16:06:54 +0200 (CEST) Subject: [LAU] Problems with zita-a2j/j2a In-Reply-To: <20130706115834.GA31535@linuxaudio.org> References: <20130706115834.GA31535@linuxaudio.org> Message-ID: Hello Fons! I didn't find the bug, but since I had to reboot for other purposes, I recheck3ed and it works again. Perhaps something wrong with the firmware or ALSA lost it or whatever. No blame to anyone specific, but certainly blame away from zita, since I did try something before and that too didn't function as planned. Anyway, all is well, which is always a good ending for a story. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From allcoms at gmail.com Sat Jul 6 14:41:04 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Sat, 6 Jul 2013 15:41:04 +0100 Subject: [LAU] Yocto : embedded Linux platform In-Reply-To: <51D7B419.4000805@rahul.net> References: <49876.188.26.254.75.1373089566.squirrel@boosthardware.com> <51D7B419.4000805@rahul.net> Message-ID: Why do I get the feeling you're being less than sincere in your praise of yocto Ken? It used to be called open embedded (and I think OpenZaurus before that) iirc and I've never had much luck with it. On Sat, Jul 6, 2013 at 7:07 AM, Ken Dawson wrote: > On 07/05/2013 10:46 PM, Patrick Shirkey wrote: > > Hi, > > > > For those of you who are interested in embedded Linux and Linux > > Audio/Multimedia you may also be interested in the ongoing development > > around the Yocto project which seeks to provide a compiler framework and > > platform for creating bespoke embedded distributions. Heavily sponsored > > by Intel and a number of other large multinational companies. Also has > > nothing to do with Android. Just pure Linux without all the other crap. > > > > https://www.yoctoproject.org/ > > > > ... with another high-quality community supporting it. > > /ken > _______________________________________________ > 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 Sat Jul 6 15:54:36 2013 From: len at ovenwerks.net (Len Ovens) Date: Sat, 6 Jul 2013 08:54:36 -0700 Subject: [LAU] leap motion sdk open Message-ID: <36500c4d17f14e686ac63747ae1c17e5.squirrel@ssl.ovenwerks.net> On Fri, July 5, 2013 10:57 pm, Patrick Shirkey wrote: > Hi, > > FYI, the Leap Motion SDK and Developer portal has been opened up today for > anyone to join. > > https://developer.leapmotion.com Interesting device. One hopes this does not send all our desktops even more into the made for touchscreen kinds of things. Maybe it will instead open the small touchscreen devices back up towards full desktop kinds of use. (full screen apps annoy me) A tablet could have such a device built into it, a small laser device could project a control interface onto any surface, keyboard (text or music with multi manuals) a mixer, wave editor... of course these things could also be done on the main screen.. but I am thinking the wrist angle for some common tasks would be better on a horizontal surface. A 3D/holographic display is what is needed to complete this gadget. The small tablet/smartphone devices suffer from low screen real estate and so buttons must be big to get touch info accurately, This device could allow for a much bigger space to be used even on a small screen and smaller buttons. How open is the driver? "SDK for Windows 7 and 8, Mac OSX 10.7 and higher, and Ubuntu Linux 12.04 LTS and Ubuntu 13.04 Raring Ringtail" sounds like not open to me. -- Len Ovens www.OvenWerks.net From abonnements at revolwear.com Sat Jul 6 16:07:56 2013 From: abonnements at revolwear.com (Max) Date: Sat, 6 Jul 2013 18:07:56 +0200 Subject: [LAU] leap motion sdk open In-Reply-To: <36500c4d17f14e686ac63747ae1c17e5.squirrel@ssl.ovenwerks.net> References: <36500c4d17f14e686ac63747ae1c17e5.squirrel@ssl.ovenwerks.net> Message-ID: <5834D644-3C45-4EDB-87AB-23098135F63D@revolwear.com> Am 06.07.2013 um 17:54 schrieb Len Ovens : > > On Fri, July 5, 2013 10:57 pm, Patrick Shirkey wrote: >> Hi, >> >> FYI, the Leap Motion SDK and Developer portal has been opened up today for >> anyone to join. >> >> https://developer.leapmotion.com > > Interesting device. One hopes this does not send all our desktops even > more into the made for touchscreen kinds of things. Maybe it will instead > open the small touchscreen devices back up towards full desktop kinds of > use. (full screen apps annoy me) A tablet could have such a device built > into it, a small laser device could project a control interface onto any > surface, keyboard (text or music with multi manuals) a mixer, wave > editor... of course these things could also be done on the main screen.. > but I am thinking the wrist angle for some common tasks would be better on > a horizontal surface. A 3D/holographic display is what is needed to > complete this gadget. > > The small tablet/smartphone devices suffer from low screen real estate and > so buttons must be big to get touch info accurately, This device could > allow for a much bigger space to be used even on a small screen and > smaller buttons. > > How open is the driver? "SDK for Windows 7 and 8, Mac OSX 10.7 and higher, > and Ubuntu Linux 12.04 LTS and Ubuntu 13.04 Raring Ringtail" sounds like > not open to me. maybe you can put you hopes in http://duo3d.com/ which failed its kickstarter but it shows that it's possible to do a leap with of the shelve parts. From len at ovenwerks.net Sat Jul 6 16:24:09 2013 From: len at ovenwerks.net (Len Ovens) Date: Sat, 6 Jul 2013 09:24:09 -0700 Subject: [LAU] leap motion sdk open Message-ID: <286ead29c7f63e550ed09a217fe3b23b.squirrel@ssl.ovenwerks.net> On Sat, July 6, 2013 9:07 am, Max wrote: > Am 06.07.2013 um 17:54 schrieb Len Ovens : >> On Fri, July 5, 2013 10:57 pm, Patrick Shirkey wrote: >>> FYI, the Leap Motion SDK and Developer portal has been opened up today >>> for >>> anyone to join. >>> >>> https://developer.leapmotion.com >> How open is the driver? "SDK for Windows 7 and 8, Mac OSX 10.7 and >> higher, >> and Ubuntu Linux 12.04 LTS and Ubuntu 13.04 Raring Ringtail" sounds like >> not open to me. > > maybe you can put you hopes in http://duo3d.com/ which failed its > kickstarter but it shows that it's possible to do a leap with of the > shelve parts. The main thing it shows is that there are likely to be more than one company making them and that there will likely end up being some sort of IF standard. The duo has pretty pictures and videos, but not much info... in fact less than the leap site. I do wonder how the Leap "requires" internet access. -- Len Ovens www.OvenWerks.net From gnome at hawaii.rr.com Sat Jul 6 19:40:07 2013 From: gnome at hawaii.rr.com (david) Date: Sat, 06 Jul 2013 09:40:07 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D758CE.1060008@gmail.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D7190A.4090800@gmail.com> <51D720A1.3090602@hawaii.rr.com> <51D758CE.1060008@gmail.com> Message-ID: <51D87297.7040801@hawaii.rr.com> On 07/05/2013 01:37 PM, Al Thompson wrote: > On 07/05/2013 03:38 PM, david wrote: >> On 07/05/2013 09:05 AM, Al Thompson wrote: >>> On 07/05/2013 02:38 PM, david wrote: >>>> Doesn't play here on Debian Sid Linux with Firefox 22. Displays >>>> message: >>>> " It appears that we are unable to play this video in your browser for >>>> one of the following reasons: >>>> >>>> you have JavaScript disabled >>>> your device does not support HTML5 video >>>> you have an old version of the Adobe Flash Player. Download the >>>> latest Flash player to view this video." >>>> >>>> IIRC, HMTL5 video support is very dependent on video drivers: one >>>> great disadvantage compared to good old proprietary Flash. >>>> >>>> Well, downloaded the MP4 and tried to play it, and it won't play. >>>> Rather, it plays, but there's no sound and no video. Deleted it and >>>> tried again, and now mplayer plays it. >>> >>> Plays fine here on Fedora 14 (64 bit) and Firefox 22.0. >> >> What video driver are you using? My laptop has Intel 855GM video. > > i915, but I don't remember why. It's on an HP laptop, and it works > fine, so that's all that really matters to me. That's much newer than the 8xx series! -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From gnome at hawaii.rr.com Sat Jul 6 19:44:13 2013 From: gnome at hawaii.rr.com (david) Date: Sat, 06 Jul 2013 09:44:13 -1000 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D75416.2020109@wootangent.net> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D75416.2020109@wootangent.net> Message-ID: <51D8738D.80105@hawaii.rr.com> On 07/05/2013 01:17 PM, Leigh Dyer wrote: > On 6/07/13 4:38 AM, david wrote: >> Doesn't play here on Debian Sid Linux with Firefox 22. Displays message: >> " It appears that we are unable to play this video in your browser for >> one of the following reasons: >> >> you have JavaScript disabled >> your device does not support HTML5 video >> you have an old version of the Adobe Flash Player. Download the >> latest Flash player to view this video." >> >> IIRC, HMTL5 video support is very dependent on video drivers: one great >> disadvantage compared to good old proprietary Flash. >> > It shouldn't be dependent on drivers, but not all browsers that support > HTML5 video (or audio, for that matter) will play all HTML5 videos. > Different browsers support different codecs, and while Chrome, Safari, > and IE support proprietary codecs like H.264 and AAC, Firefox supports > only free codecs, like Theora, VP8 (aka WebM) and Vorbis. > > So, while Firefox mostly works with HTML5 video on Youtube (since Google > supports WebM there for many videos), most other sites only offer H.264 > video via HTML5, which won't work with Firefox. Most of those sites will > fall back to using Flash to play such videos on Firefox if it's > available, though. > > Thanks > Leigh Thanks, I think a cause has been identified. I have H.264 libraries installed from deb-multimedia. Too bad someone hasn't done like Adobe and written a FF plugin for H.264. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From kevinc at cosgroves.us Sat Jul 6 22:36:34 2013 From: kevinc at cosgroves.us (Kevin Cosgrove) Date: Sat, 06 Jul 2013 15:36:34 -0700 Subject: [LAU] Ripping Vinyl Message-ID: <20130706223634.886C745842@joseph.cosgroves.us> Hiya, I'm going to use this Audacity recipe http://manual.audacityteam.org/o/man/sample_workflow_for_lp_digitization.html to rip my album collection. I might opt for Gnome Wave Cleaner to get rid of the crackle, if any. Is there any alternate method which I should consider for better results? Thanks.... -- Kevin From bob at mellowood.ca Sun Jul 7 00:51:40 2013 From: bob at mellowood.ca (Bob van der Poel) Date: Sat, 6 Jul 2013 17:51:40 -0700 Subject: [LAU] Off topic - Bluegrass for boys! In-Reply-To: <51D8738D.80105@hawaii.rr.com> References: <51D70355.4060801@gmail.com> <51D712A5.9090109@hawaii.rr.com> <51D75416.2020109@wootangent.net> <51D8738D.80105@hawaii.rr.com> Message-ID: This is really driving me nuts! I've been trying all kinds of things to get the ted.com vids to play using FF. I'm down to the conclusion that there is something in my firefox profile preventing this. If I create a new profile or start in safe mode I can NOT run ted.com video. I get a message in my flash window saying "loading" and then an error 2032. But, with a completely new/empty .mozilla directory things work fine. And, now I just deleted some old profiles from my existing ./mozilla/firefox directory and left myself with just one profile. And IT WORKS. -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob at mellowood.ca WWW: http://www.mellowood.ca From ralf.mardorf at alice-dsl.net Sun Jul 7 12:03:01 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 07 Jul 2013 14:03:01 +0200 Subject: [LAU] Ripping Vinyl Message-ID: <1373198581.1623.0.camel@archlinux> On Sat, 2013-07-06 at 15:36 -0700, Kevin Cosgrove wrote: > http://manual.audacityteam.org/o/man/sample_workflow_for_lp_digitization.html Don't follow this idiotic instruction. Clean a "normal dusty" LP only with a special blanket, don't wash such an LP. Don't edit the recording, e.g. don't add compression, if needed do this for your classical car collection only, not to archive LPs. From axeldenstore at gmail.com Sun Jul 7 14:02:19 2013 From: axeldenstore at gmail.com (alexander) Date: Sun, 07 Jul 2013 17:02:19 +0300 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: <51D974EB.6040602@gmail.com> Great stuff, it's really fun to hear my sampling libraries in action! On 03/07/13 18:05, Gabbe Nord wrote: > Hello everyone! > > I'm very pleased to announce that my second album is finished and > online! :D > It's called "Ordinary Day Montage", and it consists of 8 tracks, and > is a bit more electronic than last one. > > I prepared a page for this at my new website, > http://zthmusic.se/Ordinary_Day_Montage , but my host is a little > shaky, so it might not always work. So, if that link doesn't work, you > can find the album at: > > Bandcamp > http://zthmusic.bandcamp.com/album/ordinary-day-montage > Soundcloud > https://soundcloud.com/zthmusic/sets/ordinary-day-montage > FLAC/OGG/MP3-formats for download at Piratebay: > http://thepiratebay.sx/user/zthmusic/ > > I'm very excited to be finished and to have completed this. Everything > was, as always, 100% made with Linux and Linux software. It's also > licensed CC-BY-SA. I wrote a bit about the album and what I've used > technically too at http://zthmusic.se/Ordinary_Day_Montage , if anyone > wants to check that out! > > Anyway, thank you for taking your time to listen! I greatly appreciate it! > > > _______________________________________________ > 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 malnourite at gmail.com Sun Jul 7 21:50:51 2013 From: malnourite at gmail.com (J. Liles) Date: Sun, 7 Jul 2013 14:50:51 -0700 Subject: [LAU] Non Mixer Spatializer Demo Message-ID: As many of you already know, Non Mixer (http://non.tuxfamily.org) has since its inception provided some cushy features for dealing with Ambisonics mixes. Lately I've been working on extending these features to the next level. This is a quick demonstration of the new Spatializer module and the associated Spatialization Console. What this does is provide synthetic distance cues as well as a cushy interface for placing sounds in virtual space. http://youtu.be/GVm5Jd1WDWw I'm hoping to have these new features ready for testing soon--but my free time is very limited. As always, donations are welcome and very much appreciated. (http://non.tuxfamily.org/wiki/Donations) A note about the video: Every time I try to make one of these screencasts, I run into the same problem: nothing works. Thus, I had to aim my video camera at the screen and record audio from my computer via line-out. This is problematic for a number of reasons. 1) I can only record stereo this way 2) my camera only samples audio at 32KHz and 3) When youtube transcodes the video it adds an annoying click about once per second (I believe this is due to a mismatch between the camera's framerate and youtube's expectations. I would love to provide a theora/vorbis screencapture video instead, but, alas, I cannot find any tools that can capture screen activity and record audio via JACK in sync. Anyway, the poor quality of the video is not due to a lack of effort. From ralf.mardorf at alice-dsl.net Sun Jul 7 22:00:14 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 00:00:14 +0200 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: Message-ID: <1373234414.663.4.camel@archlinux> On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: > http://youtu.be/GVm5Jd1WDWw Can't be played with Firefox 22.0. Google Chrome can play it, but IMO the need for Adobe Flash Player should be dropped by videos done by the Linux community. I won't install Adobe Flash Player anymore and usually I don't have Chrome installed either. "NOTE: Adobe Flash Player 11.2 will be the last version to target Linux as a supported platform. Adobe will continue to provide security backports to Flash Player 11.2 for Linux." From malnourite at gmail.com Sun Jul 7 22:08:55 2013 From: malnourite at gmail.com (J. Liles) Date: Sun, 7 Jul 2013 15:08:55 -0700 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: <1373234414.663.4.camel@archlinux> References: <1373234414.663.4.camel@archlinux> Message-ID: On Sun, Jul 7, 2013 at 3:00 PM, Ralf Mardorf wrote: > On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: >> http://youtu.be/GVm5Jd1WDWw > > Can't be played with Firefox 22.0. Google Chrome can play it, but IMO > the need for Adobe Flash Player should be dropped by videos done by the > Linux community. > > I won't install Adobe Flash Player anymore and usually I don't have > Chrome installed either. > > "NOTE: Adobe Flash Player 11.2 will be the last version to target Linux > as a supported platform. Adobe will continue to provide security > backports to Flash Player 11.2 for Linux." > Ralf, please, by all means help by fixing recordmydesktop or one of the other tools so that I and others can record such demos using only free-software and open codecs. Or point me at some solution that works. I have very limited options in this regard. I have no desire to use youtube. From louigi.verona at gmail.com Sun Jul 7 22:18:32 2013 From: louigi.verona at gmail.com (Louigi Verona) Date: Mon, 8 Jul 2013 02:18:32 +0400 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: <1373234414.663.4.camel@archlinux> References: <1373234414.663.4.camel@archlinux> Message-ID: Slightly OT: I am usually very liberal when it comes to people's worldview and this is sincere behavior, I really do respect other people's beliefs and ideals and I do not need to make any special effort to politely leave my opinions to myself. However, sometimes I think it is helpful to point out certain things, so that people at the very least are aware that an opposing opinion exists. I'll be brief. 1. I think this is frustrating when you post a link to YouTube and someone writes a long letter, explaining how they dislike proprietary Flash and how it does not work on their system or on their browser. Please, this is YouTube, one of the most used services in the world. If you cannot watch it - get a plugin to download videos. There is no need to tell us for the hundredth time that you cannot be bothered to find a working technical solution to a problem that is simpler than anything else going on in your command line. Seriously. 2. Adobe Flash is proprietary and we all know it. Period. Thus, it is really not necessary to use every situation you can to push your free software fundamentalism on other people. If I use Linux, it does not mean that I agree with Richard Stallman on everything. This is not specifically to Ralf, since I have read other people doing this as well. I apologize for going OT and would please ask to not take this personally as this is not meant to be. Sincerely, Louigi Verona. http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Sun Jul 7 22:25:35 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 00:25:35 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> Message-ID: <1373235935.663.18.camel@archlinux> Fixing the desktop recording issues is beyond my scope. However, Firefox is able to play many YouTube videos, so it seems to be an issue regarding to a used codec or similar. There's no need to avoid YouTube, you "only" need to convert your camera recordings in a way, that the videos can be played with HTML 5 capable browsers. I don't know how much work the "only" to do would cause, since I have given up Linux video a long, long time ago. I already have more than enough issues with Linux audio, still couldn't get the RME card working as it should work ;). From malnourite at gmail.com Sun Jul 7 22:36:27 2013 From: malnourite at gmail.com (J. Liles) Date: Sun, 7 Jul 2013 15:36:27 -0700 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373235935.663.18.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> Message-ID: On Sun, Jul 7, 2013 at 3:25 PM, Ralf Mardorf wrote: > Fixing the desktop recording issues is beyond my scope. > > However, Firefox is able to play many YouTube videos, so it seems to be > an issue regarding to a used codec or similar. > > There's no need to avoid YouTube, you "only" need to convert your camera > recordings in a way, that the videos can be played with HTML 5 capable > browsers. > > I don't know how much work the "only" to do would cause, since I have > given up Linux video a long, long time ago. I already have more than > enough issues with Linux audio, still couldn't get the RME card working > as it should work ;). > Youtube transcodes all uploads to suit their architecture. The codec of the uploaded video makes no difference. From ralf.mardorf at alice-dsl.net Sun Jul 7 22:40:38 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 00:40:38 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373235935.663.18.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> Message-ID: <1373236838.663.24.camel@archlinux> Hi Louigi, the reason that now relatively many people don't want Flash Player anymore is that it's not maintained for Linux anymore and even on other platforms it's not available. In the past I used Adobe Flash Player on Linux myself and I still use proprietary software with Linux. So at least for me the reason isn't Linux operating system religion, it's simply that Flash Player is a Windows only thing, not only an issue for Linux. Regards, Ralf From ralf.mardorf at alice-dsl.net Sun Jul 7 22:45:18 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 00:45:18 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373236838.663.24.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> Message-ID: <1373237118.663.26.camel@archlinux> On Mon, 2013-07-08 at 00:40 +0200, Ralf Mardorf wrote: > So at least for me the reason isn't Linux operating system religion, > it's simply that Flash Player is a Windows only thing, not only an issue > for Linux. Sure, it's available for Mac too, but I guess you know what I mean. From ralf.mardorf at alice-dsl.net Sun Jul 7 22:50:13 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 00:50:13 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> Message-ID: <1373237413.663.29.camel@archlinux> On Sun, 2013-07-07 at 15:36 -0700, J. Liles wrote: > Youtube transcodes all uploads to suit their architecture. The codec > of the uploaded video makes no difference. Some days ago, a video from another thread on this list and there was a discussion about Flash Player vs HTML 5 issues too: http://www.youtube.com/watch?v=gSedE5sU3uc This video can be played with Firefox, without Flash Player and it's on YouTube. From malnourite at gmail.com Sun Jul 7 22:50:19 2013 From: malnourite at gmail.com (J. Liles) Date: Sun, 7 Jul 2013 15:50:19 -0700 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373237118.663.26.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> <1373237118.663.26.camel@archlinux> Message-ID: On Sun, Jul 7, 2013 at 3:45 PM, Ralf Mardorf wrote: > On Mon, 2013-07-08 at 00:40 +0200, Ralf Mardorf wrote: >> So at least for me the reason isn't Linux operating system religion, >> it's simply that Flash Player is a Windows only thing, not only an issue >> for Linux. > > Sure, it's available for Mac too, but I guess you know what I mean. > Ralf, respectfully, if you can't offer any solutions, then your comments offer nothing of value. They simply serve to distract the members of this list. From ralf.mardorf at alice-dsl.net Sun Jul 7 23:31:17 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 01:31:17 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> <1373237118.663.26.camel@archlinux> Message-ID: <1373239877.663.35.camel@archlinux> On Sun, 2013-07-07 at 15:50 -0700, J. Liles wrote: > Ralf, respectfully, if you can't offer any solutions, then your > comments offer nothing of value. They simply serve to distract the > members of this list. Perhaps this http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-July/093553.html + this http://en.wikipedia.org/wiki/YouTube#Quality_and_codecs does help?! User often can't support solutions ;), but we experience that things don't work and IMO this should be reported. The Flash Player issue leads to discussions on Linux and FreeBSD lists, often people are unable to get Flash Player working and soon or later it definitive won't work anymore. From countfuzzball at gmail.com Mon Jul 8 03:33:38 2013 From: countfuzzball at gmail.com (Andrew C) Date: Mon, 8 Jul 2013 04:33:38 +0100 Subject: [LAU] Music made with linux: Another classic ruined. Message-ID: More mindless musical wankery and overdubs. Mustn't forget the overdub. Swiftly demolishing The Animals' rendition of House of the rising sun in 1 minute and 40 seconds. https://soundcloud.com/lladbeldi/another-classic-ruined Flame away! :D Andrew. -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.kronlachner at student.tugraz.at Mon Jul 8 09:21:19 2013 From: m.kronlachner at student.tugraz.at (Matthias Kronlachner) Date: Mon, 08 Jul 2013 12:21:19 +0300 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: Message-ID: <51DA848F.30802@student.tugraz.at> looks great! waiting for being able to test those. i realized that you set the maximum channel count per track to 16 in non-mixer. apart from being easily changed in the source, is there a reason to do that? do you intend to support lv2 at some point? thanks for the great work, matthias On 7/8/13 12:50 AM, J. Liles wrote: > As many of you already know, Non Mixer (http://non.tuxfamily.org) has > since its inception provided some cushy features for dealing with > Ambisonics mixes. > > Lately I've been working on extending these features to the next level. > > This is a quick demonstration of the new Spatializer module and the > associated Spatialization Console. What this does is provide synthetic > distance cues as well as a cushy interface for placing sounds in > virtual space. > > http://youtu.be/GVm5Jd1WDWw > > I'm hoping to have these new features ready for testing soon--but my > free time is very limited. > > As always, donations are welcome and very much appreciated. > (http://non.tuxfamily.org/wiki/Donations) > > A note about the video: > > Every time I try to make one of these screencasts, I run into the same > problem: nothing works. Thus, I had to aim my video camera at the > screen and record audio from my computer via line-out. This is > problematic for a number of reasons. 1) I can only record stereo this > way 2) my camera only samples audio at 32KHz and 3) When youtube > transcodes the video it adds an annoying click about once per second > (I believe this is due to a mismatch between the camera's framerate > and youtube's expectations. I would love to provide a theora/vorbis > screencapture video instead, but, alas, I cannot find any tools that > can capture screen activity and record audio via JACK in sync. Anyway, > the poor quality of the video is not due to a lack of effort. > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From julien at mail.upb.de Mon Jul 8 09:22:58 2013 From: julien at mail.upb.de (Julien Claassen) Date: Mon, 8 Jul 2013 11:22:58 +0200 (CEST) Subject: [LAU] Music made with linux: Another classic ruined. In-Reply-To: References: Message-ID: Hello Andrew! In comparison to the original it is ruined. But at least it's thourroughly ruined, so that is alright. :-) Good 80s sound, using the alesis HW16 drumkit. One of the soundfonts, that was floating around? The thing, that realy interests me, is how you got the lead and where you got it from? There's obviously some distortion.At first I thought it was a Nordlead3 program, but then I recognised the differences. But the lead was a fantastic decision. In the beginning it sounded almost like the typical MIDI file player of dubious quality, but that lead lent some live and motion to it. there's not much movement in the sound itself, but the way you played it and its own sonic character contrasted very well with the rest. In keeping with the road of your piece Best Julien :-) ---------------------------------------- http://juliencoder.de/nama/music.html From karl at aspodata.se Mon Jul 8 09:28:15 2013 From: karl at aspodata.se (Karl Hammar) Date: Mon, 8 Jul 2013 11:28:15 +0200 (CEST) Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373236838.663.24.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> Message-ID: <20130708092815.C1ACC8043961@turkos.aspodata.se> Ralf: > the reason that now relatively many people don't want Flash Player > anymore is that it's not maintained for Linux anymore and even on other > platforms it's not available. In the past I used Adobe Flash Player on > Linux myself and I still use proprietary software with Linux. ... Interesting, here in Sweden, the former state television company is "broadcasting" tv over Internet. They are using flash and their sister company (the one collectiong the tv-licence fees) wants every- body with a computer and an internet connection to pay. If one can show that flash is non-linux, perhaps linux users could go free. Do you have any proof that would hold at a court ? Regards, /Karl Hammar ----------------------------------------------------------------------- Asp? Data Lilla Asp? 148 S-742 94 ?sthammar Sweden +46 173 140 57 From neil at neilcsmith.net Mon Jul 8 09:46:10 2013 From: neil at neilcsmith.net (Neil C Smith) Date: Mon, 8 Jul 2013 10:46:10 +0100 Subject: [LAU] Music made with linux: Another classic ruined. In-Reply-To: References: Message-ID: On 8 July 2013 10:22, Julien Claassen wrote: > In comparison to the original it is ruined. But at least it's thourroughly > ruined, so that is alright. :-) Yes, a most excellent ruination which I like almost as much (take this as a big complement!) as this - http://youtu.be/Ou5_ypX4_iw [OT] I once made a 6hr road trip with only the 386DX album to listen to - an experience I, and most especially my passengers, won't forget! :-) Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source, graphical environment for rapid development of intermedia performance tools, projections and interactive spaces - http://code.google.com/p/praxis OpenEye - specialist web solutions for the cultural, education, charitable and local government sectors - http://openeye.info From countfuzzball at gmail.com Mon Jul 8 10:26:55 2013 From: countfuzzball at gmail.com (Andrew C) Date: Mon, 8 Jul 2013 11:26:55 +0100 Subject: [LAU] Music made with linux: Another classic ruined. In-Reply-To: References: Message-ID: Hey Julian and Neil, Thanks for the kind words, glad you both liked it. Yeah, the backing track and vibraphone/oohs are from my Yamaha PSR-E413 (a home keyboard, but does the trick most times). The lead sound is from the Bristol Polysix run through Guitarix for the distortion. The patch is fairly simple, six oscillators set to a pulse wave with a width of about 64-67% and each one slightly detuned, sent through a lowpass filter with a cutoff at about 10 o' clock, resonance at about 9 o' clock and filter tracking set to fully open. No velocity tracking (in keeping with the spirit of the original Polysix). The majority of the sound comes from the patch itself and how I set Bristol to run on startup. The distortion is fairly easy to do: Cranking it up is better. If you want, I can send you the patches off-list, not too sure if guitarix is at all easy to control from the console though. Also, while 386DX's version is rather admirable and I enjoyed it immensely, I believe that this should get the crowning achievement for best worst cover: http://www.youtube.com/watch?v=RlNYIYK7Itk Thanks again, Andrew. On Mon, Jul 8, 2013 at 10:46 AM, Neil C Smith wrote: > On 8 July 2013 10:22, Julien Claassen wrote: > > In comparison to the original it is ruined. But at least it's > thourroughly > > ruined, so that is alright. :-) > > Yes, a most excellent ruination which I like almost as much (take this > as a big complement!) as this - http://youtu.be/Ou5_ypX4_iw > > [OT] I once made a 6hr road trip with only the 386DX album to listen > to - an experience I, and most especially my passengers, won't forget! > :-) > > Best wishes, > > Neil > > -- > Neil C Smith > Artist : Technologist : Adviser > http://neilcsmith.net > > Praxis LIVE - open-source, graphical environment for rapid development > of intermedia performance tools, projections and interactive spaces - > http://code.google.com/p/praxis > > OpenEye - specialist web solutions for the cultural, education, > charitable and local government sectors - http://openeye.info > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lsd at wootangent.net Mon Jul 8 10:28:20 2013 From: lsd at wootangent.net (Leigh Dyer) Date: Mon, 08 Jul 2013 20:28:20 +1000 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: <1373234414.663.4.camel@archlinux> References: <1373234414.663.4.camel@archlinux> Message-ID: <51DA9444.5060207@wootangent.net> On 8/07/13 8:00 AM, Ralf Mardorf wrote: > On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: >> http://youtu.be/GVm5Jd1WDWw > > Can't be played with Firefox 22.0. Google Chrome can play it, but IMO > the need for Adobe Flash Player should be dropped by videos done by the > Linux community. Youtube usually does transcode videos to WebM format for playback on Firefox, etc. using only free codecs. I can't say why Youtube hasn't done that yet for this video; it could just be that it takes a while for the WebM transcodes to happen, and not enough time has elapsed yet. Perhaps within a few days the WebM version will be available and you'll be able to watch the video in Firefox. As far as major video sites go, Youtube is way ahead of the curve in its support for free codecs -- there's no alternative video sharing that I know of that offers such (relatively) comprehensive WebM support. The alternative is for people posting videos to host them themselves, but that's non-trivial, since you also have to also provide a h.264/AAC version to have the video work on IE, Safari, and most mobile devices. It'd be a shame if users of those platforms were unable to see some of the cool things happening in the free software world. So... it's not an easy problem to solve, and currently, there's not really a better solution than Youtube. Thanks Leigh From lsd at wootangent.net Mon Jul 8 10:30:14 2013 From: lsd at wootangent.net (Leigh Dyer) Date: Mon, 08 Jul 2013 20:30:14 +1000 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <1373237413.663.29.camel@archlinux> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373237413.663.29.camel@archlinux> Message-ID: <51DA94B6.9040403@wootangent.net> On 8/07/13 8:50 AM, Ralf Mardorf wrote: > On Sun, 2013-07-07 at 15:36 -0700, J. Liles wrote: >> Youtube transcodes all uploads to suit their architecture. The codec >> of the uploaded video makes no difference. > > Some days ago, a video from another thread on this list and there was a > discussion about Flash Player vs HTML 5 issues too: > > http://www.youtube.com/watch?v=gSedE5sU3uc > > This video can be played with Firefox, without Flash Player and it's on > YouTube. That's just due to the fact that Youtube has transcoded that particular video to WebM; it has nothing to do with the format it was uploaded in. Thanks Leigh From julien at mail.upb.de Mon Jul 8 10:37:18 2013 From: julien at mail.upb.de (Julien Claassen) Date: Mon, 8 Jul 2013 12:37:18 +0200 (CEST) Subject: [LAU] Music made with linux: Another classic ruined. In-Reply-To: References: Message-ID: Hi Andrew! I'd like the Bristol patch, if that's nobother. I can use the LADSPA guitarix suite, but I ca't load patches. Still, that can be configured I'm sure. The basic interest is in the synth patch. although from your description, I think I'll be able to recreate it with ease. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From lsd at wootangent.net Mon Jul 8 10:42:11 2013 From: lsd at wootangent.net (Leigh Dyer) Date: Mon, 08 Jul 2013 20:42:11 +1000 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <20130708092815.C1ACC8043961@turkos.aspodata.se> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> <20130708092815.C1ACC8043961@turkos.aspodata.se> Message-ID: <51DA9783.70106@wootangent.net> On 8/07/13 7:28 PM, Karl Hammar wrote: > Ralf: >> the reason that now relatively many people don't want Flash Player >> anymore is that it's not maintained for Linux anymore and even on other >> platforms it's not available. In the past I used Adobe Flash Player on >> Linux myself and I still use proprietary software with Linux. > ... > > Interesting, here in Sweden, the former state television company > is "broadcasting" tv over Internet. They are using flash and their > sister company (the one collectiong the tv-licence fees) wants every- > body with a computer and an internet connection to pay. > > If one can show that flash is non-linux, perhaps linux users could > go free. > > Do you have any proof that would hold at a court ? Here's the official announcement from Adobe -- they haven't dropped Linux support as such, but they've certainly cut back on it: http://blogs.adobe.com/flashplayer/2012/02/adobe-and-google-partnering-for-flash-player-on-linux.html In short, Adobe has stopped releasing new versions of Flash as a plugin for Firefox, etc. -- it will receive security fixes for the next five years, but no new features. Adobe will continue to make new versions of Flash available within Google Chrome, though. Thans Leigh From ralf.mardorf at alice-dsl.net Mon Jul 8 10:55:08 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 08 Jul 2013 12:55:08 +0200 Subject: [LAU] Flash Player - Was: Non Mixer Spatializer Demo In-Reply-To: <20130708092815.C1ACC8043961@turkos.aspodata.se> References: <1373234414.663.4.camel@archlinux> <1373235935.663.18.camel@archlinux> <1373236838.663.24.camel@archlinux> <20130708092815.C1ACC8043961@turkos.aspodata.se> Message-ID: <1373280908.663.30.camel@archlinux> On Mon, 2013-07-08 at 11:28 +0200, Karl Hammar wrote: > Ralf: > > the reason that now relatively many people don't want Flash Player > > anymore is that it's not maintained for Linux anymore and even on other > > platforms it's not available. In the past I used Adobe Flash Player on > > Linux myself and I still use proprietary software with Linux. > ... > > Interesting, here in Sweden, the former state television company > is "broadcasting" tv over Internet. They are using flash and their > sister company (the one collectiong the tv-licence fees) wants every- > body with a computer and an internet connection to pay. > > If one can show that flash is non-linux, perhaps linux users could > go free. > > Do you have any proof that would hold at a court That's another issue we had in Germany too. The issue now is solved in Germany, everybody who has got a residence is forced to pay for television, even if s/he won't and/or can't watch. Exceptions are rare, e.g. some handicapped persons don't need to pay. Before the law was changed they tried to get money from computer users too, but many people, including me, successfully fight against this, so they didn't get money from us. I really don't watch scripted realty and all that high-class crap. One assumption of the former administration was, that everybody does own a mobile pone with media players. They didn't believe me, that I even don't own a mobile phone without a media player. The evidence you want to know, doesn't help you. "NOTE: Adobe Flash Player 11.2 will be the last version to target Linux as a supported platform. Adobe will continue to provide security backports to Flash Player 11.2 for Linux." - http://get.adobe.com/flashplayer/?promoid=JZEFT "latest versions are 11.7.700.224 (Win), 11.7.700.225 (Mac) and 11.2.202.291 (Linux)" - http://www.adobe.com/support/flashplayer/downloads.html The day when Flash Player for Linux will be too outdated is near. Regards, Ralf From willgodfrey at musically.me.uk Mon Jul 8 19:14:06 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 8 Jul 2013 20:14:06 +0100 Subject: [LAU] Music made with linux: Another classic ruined. In-Reply-To: References: Message-ID: <20130708201406.517272f5@debian> On Mon, 8 Jul 2013 04:33:38 +0100 Andrew C wrote: > More mindless musical wankery and overdubs. Mustn't forget the overdub. > Swiftly demolishing The Animals' rendition of House of the rising sun in 1 > minute and 40 seconds. > > https://soundcloud.com/lladbeldi/another-classic-ruined > > Flame away! :D > > Andrew. Almost genteel :o Very enjoyable. -- 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 bjb-linux-audio-user at deus.net Mon Jul 8 19:50:51 2013 From: bjb-linux-audio-user at deus.net (Ben Bell) Date: Mon, 8 Jul 2013 20:50:51 +0100 Subject: [LAU] Debut prog-rock album, all produced with Open Source! Message-ID: <20130708195051.GL32655@deus.net> Hi folks, Having seen all the wonderful Linux-produced efforts being announced here I couldn't resist posting this one. Last Monday my band, Fusion Orchestra 2, released their new album "Casting Shadows". All the official press releases are concentrating on the normal sorts of stuff and I'm not supposed to get into software politics in them, but I'm quietly jumping up and down about the fact that the whole thing, from start to finish, was produced in Linux using Open Source tools. Music, artwork, the lot. The bulk of the work was courtesy of Ardour (2.8.x) with mastering handled by JAMin. At some point I'll probably write a blog about the process of producing the album but for now, I'm just going to share a sampler of it: https://soundcloud.com/benjamesbell/casting-shadows-sampler/s-ad5hC Oh, and on the off-chance anyone is interested enough to buy a copy, make sure you add something to the delivery notes saying you read about it here and for each copy sold that way we'll donate $1 to Ardour. I doubt it'll make much of a splash but if everyone using Ardour does something similar I'm sure it'll help. And now that's all done I can finally make the switch to Ardour 3 for the next project ;) bjb From willgodfrey at musically.me.uk Mon Jul 8 19:58:06 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Mon, 8 Jul 2013 20:58:06 +0100 Subject: [LAU] Debut prog-rock album, all produced with Open Source! In-Reply-To: <20130708195051.GL32655@deus.net> References: <20130708195051.GL32655@deus.net> Message-ID: <20130708205806.4a0ae9fd@debian> On Mon, 8 Jul 2013 20:50:51 +0100 Ben Bell wrote: > Hi folks, > > Having seen all the wonderful Linux-produced efforts being announced here > I couldn't resist posting this one. > > Last Monday my band, Fusion Orchestra 2, released their new album "Casting > Shadows". All the official press releases are concentrating on the normal > sorts of stuff and I'm not supposed to get into software politics in them, > but I'm quietly jumping up and down about the fact that the whole thing, > from start to finish, was produced in Linux using Open Source tools. Music, > artwork, the lot. > > The bulk of the work was courtesy of Ardour (2.8.x) with mastering handled > by JAMin. At some point I'll probably write a blog about the process of > producing the album but for now, I'm just going to share a sampler of > it: > > https://soundcloud.com/benjamesbell/casting-shadows-sampler/s-ad5hC > > Oh, and on the off-chance anyone is interested enough to buy a copy, make > sure you add something to the delivery notes saying you read about it here > and for each copy sold that way we'll donate $1 to Ardour. I doubt it'll > make much of a splash but if everyone using Ardour does something similar I'm > sure it'll help. > > And now that's all done I can finally make the switch to Ardour 3 for the > next project ;) > > bjb Very impressed with this sampler. I hope the album does well for you. -- 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 fero.kiraly at gmail.com Mon Jul 8 20:11:55 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Mon, 8 Jul 2013 22:11:55 +0200 Subject: [LAU] linux-rt & presonus 1818VSL In-Reply-To: References: Message-ID: i tried to diggin in IRQ setup for low latency. situation is much better; 48000/64/3 but I would like to have maximum :) before my last ideas gonna to die, I would like to ask if my asound.rc looks good: pcm.!default { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } } ctl.mixer0 { type hw card 0 } my setup: archlinux, presonusVSL 1818, Fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From althompson58 at gmail.com Mon Jul 8 20:28:50 2013 From: althompson58 at gmail.com (Al Thompson) Date: Mon, 08 Jul 2013 16:28:50 -0400 Subject: [LAU] Debut prog-rock album, all produced with Open Source! In-Reply-To: <20130708195051.GL32655@deus.net> References: <20130708195051.GL32655@deus.net> Message-ID: <51DB2102.3000008@gmail.com> On 07/08/2013 03:50 PM, Ben Bell wrote: > Hi folks, > > Having seen all the wonderful Linux-produced efforts being announced here > I couldn't resist posting this one. > > Last Monday my band, Fusion Orchestra 2, released their new album "Casting > Shadows". All the official press releases are concentrating on the normal > sorts of stuff and I'm not supposed to get into software politics in them, > but I'm quietly jumping up and down about the fact that the whole thing, > from start to finish, was produced in Linux using Open Source tools. Music, > artwork, the lot. > > The bulk of the work was courtesy of Ardour (2.8.x) with mastering handled > by JAMin. At some point I'll probably write a blog about the process of > producing the album but for now, I'm just going to share a sampler of > it: > > https://soundcloud.com/benjamesbell/casting-shadows-sampler/s-ad5hC > Very nice!!!! My favorite genre of music. Are they keyboards, specifically the Hammond, plugins, or real instruments? -- --- My bands, CD projects, music, news, and pictures: http://www.lateralforce.com My blog, with commentary on a variety of things, including audio, mixing, equipment, etc, is at: http://audioandmore.wordpress.com Staat hei?t das k?lteste aller kalten Ungeheuer. Kalt l?gt es auch; und diese L?ge kriecht aus seinem Munde: 'Ich, der Staat, bin das Volk.' - [Friedrich Nietzsche] From bjb-linux-audio-user at deus.net Mon Jul 8 20:44:09 2013 From: bjb-linux-audio-user at deus.net (Ben Bell) Date: Mon, 8 Jul 2013 21:44:09 +0100 Subject: [LAU] Debut prog-rock album, all produced with Open Source! In-Reply-To: <51DB2102.3000008@gmail.com> References: <20130708195051.GL32655@deus.net> <51DB2102.3000008@gmail.com> Message-ID: <20130708204409.GO32655@deus.net> On Mon, Jul 08, 2013 at 04:28:50PM -0400, Al Thompson wrote: > Very nice!!!! My favorite genre of music. Thank you! > Are they keyboards, specifically the Hammond, plugins, or real instruments? They're all real(ish) keyboards. The Hammond's an XK3c, the piano's a digital Yamaha P80, the clav's from a Nord Stage. The synths are primarily Prophet 08 and Nord Stage again, and the mellotron impressions are from an Emu Classic Keys. The latter will probably be replaced with Tajiguy samples in future recordings. The guitars are through Marshall, Aguilar, and Yamaha pre-amps, though in a few places I recorded them quite clean and livened them up with the C* Audio Plugins. CAPS plate reverb and Freeverb did their share of lifting too :) bjb From allcoms at gmail.com Mon Jul 8 20:54:59 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Mon, 8 Jul 2013 21:54:59 +0100 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: <51DA9444.5060207@wootangent.net> References: <1373234414.663.4.camel@archlinux> <51DA9444.5060207@wootangent.net> Message-ID: There are more free software-friendly alternatives to YT. I uploaded my Ardour 3 MIDI tutorial to archive.org, for example: http://archive.org/details/movies On Mon, Jul 8, 2013 at 11:28 AM, Leigh Dyer wrote: > On 8/07/13 8:00 AM, Ralf Mardorf wrote: > >> On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: >> >>> http://youtu.be/GVm5Jd1WDWw >>> >> >> Can't be played with Firefox 22.0. Google Chrome can play it, but IMO >> the need for Adobe Flash Player should be dropped by videos done by the >> Linux community. >> > > Youtube usually does transcode videos to WebM format for playback on > Firefox, etc. using only free codecs. I can't say why Youtube hasn't done > that yet for this video; it could just be that it takes a while for the > WebM transcodes to happen, and not enough time has elapsed yet. Perhaps > within a few days the WebM version will be available and you'll be able to > watch the video in Firefox. > > As far as major video sites go, Youtube is way ahead of the curve in its > support for free codecs -- there's no alternative video sharing that I know > of that offers such (relatively) comprehensive WebM support. > > The alternative is for people posting videos to host them themselves, but > that's non-trivial, since you also have to also provide a h.264/AAC version > to have the video work on IE, Safari, and most mobile devices. It'd be a > shame if users of those platforms were unable to see some of the cool > things happening in the free software world. > > So... it's not an easy problem to solve, and currently, there's not really > a better solution than Youtube. > > Thanks > Leigh > > > ______________________________**_________________ > 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 malnourite at gmail.com Mon Jul 8 21:30:24 2013 From: malnourite at gmail.com (J. Liles) Date: Mon, 8 Jul 2013 14:30:24 -0700 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> <51DA9444.5060207@wootangent.net> Message-ID: On Mon, Jul 8, 2013 at 1:54 PM, Dan MacDonald wrote: > There are more free software-friendly alternatives to YT. I uploaded my > Ardour 3 MIDI tutorial to archive.org, for example: > > http://archive.org/details/movies > I'm aware of archive.org and would have preferred to use it (had I been able to generate an ogg/theora movie) However, since I imagine archive.orgis using the same tools that are available to me for transcoding, then I fail to see how they could transcode the output of this particular video camera without the audio going out of sync--as I tried literally every piece of free-software in myriad combinations on it. I suppose it's worth a shot, but considering that it took me some 2 and a half hours to upload the enormous mjpeg encoded video to youtube--I wasn't anxious to repeat the process. -------------- next part -------------- An HTML attachment was scrubbed... URL: From allcoms at gmail.com Mon Jul 8 22:59:51 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Mon, 8 Jul 2013 23:59:51 +0100 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> <51DA9444.5060207@wootangent.net> Message-ID: I have used RecordItNow (under KXStudio) to create screencasts painlessly recently. I then used KDEnlive to normalize the audio - done! It works fine in RMD mode but I had issues (no sound nor stop) when using its ffmpeg recording backend instead - natch. On Mon, Jul 8, 2013 at 10:30 PM, J. Liles wrote: > > On Mon, Jul 8, 2013 at 1:54 PM, Dan MacDonald wrote: > >> There are more free software-friendly alternatives to YT. I uploaded my >> Ardour 3 MIDI tutorial to archive.org, for example: >> >> http://archive.org/details/movies >> > > I'm aware of archive.org and would have preferred to use it (had I been > able to generate an ogg/theora movie) However, since I imagine archive.orgis using the same tools that are available to me for transcoding, then I > fail to see how they could transcode the output of this particular video > camera without the audio going out of sync--as I tried literally every > piece of free-software in myriad combinations on it. I suppose it's worth a > shot, but considering that it took me some 2 and a half hours to upload the > enormous mjpeg encoded video to youtube--I wasn't anxious to repeat the > process. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lsd at wootangent.net Mon Jul 8 23:18:18 2013 From: lsd at wootangent.net (Leigh Dyer) Date: Tue, 09 Jul 2013 09:18:18 +1000 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: <1373234414.663.4.camel@archlinux> <51DA9444.5060207@wootangent.net> Message-ID: <51DB48BA.80708@wootangent.net> On 09/07/13 06:54, Dan MacDonald wrote: > There are more free software-friendly alternatives to YT. I uploaded my > Ardour 3 MIDI tutorial to archive.org , for example: > > http://archive.org/details/movies Does archive.org automatically transcode videos to WebM and H.264/AAC for playback across all platforms/devices? I've been trying to find information about that on the website, but I haven't find it yet. Thanks Leigh > > > On Mon, Jul 8, 2013 at 11:28 AM, Leigh Dyer > wrote: > > On 8/07/13 8:00 AM, Ralf Mardorf wrote: > > On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: > > http://youtu.be/GVm5Jd1WDWw > > > Can't be played with Firefox 22.0. Google Chrome can play it, > but IMO > the need for Adobe Flash Player should be dropped by videos done > by the > Linux community. > > > Youtube usually does transcode videos to WebM format for playback on > Firefox, etc. using only free codecs. I can't say why Youtube hasn't > done that yet for this video; it could just be that it takes a while > for the WebM transcodes to happen, and not enough time has elapsed > yet. Perhaps within a few days the WebM version will be available > and you'll be able to watch the video in Firefox. > > As far as major video sites go, Youtube is way ahead of the curve in > its support for free codecs -- there's no alternative video sharing > that I know of that offers such (relatively) comprehensive WebM support. > > The alternative is for people posting videos to host them > themselves, but that's non-trivial, since you also have to also > provide a h.264/AAC version to have the video work on IE, Safari, > and most mobile devices. It'd be a shame if users of those platforms > were unable to see some of the cool things happening in the free > software world. > > So... it's not an easy problem to solve, and currently, there's not > really a better solution than Youtube. > > Thanks > Leigh > > > _________________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.__linuxaudio.org > > http://lists.linuxaudio.org/__listinfo/linux-audio-user > > > From kevinc at cosgroves.us Tue Jul 9 02:52:22 2013 From: kevinc at cosgroves.us (Kevin Cosgrove) Date: Mon, 08 Jul 2013 19:52:22 -0700 Subject: [LAU] Ripping Vinyl In-Reply-To: <1373198581.1623.0.camel@archlinux> Message-ID: <20130709025222.BD292BE05B@joseph.cosgroves.us> On 7 July 2013 at 14:03, Ralf Mardorf wrote: > On Sat, 2013-07-06 at 15:36 -0700, Kevin Cosgrove wrote: > > http://manual.audacityteam.org/o/man/sample_workflow_for_lp_digitization.html > > Don't follow this idiotic instruction. > > Clean a "normal dusty" LP only with a special blanket, don't > wash such an LP. I have really good LP cleaning equipment. Getting the old audio system running very well is not an issue here. I've kept that gear running since I first put it into service. > Don't edit the recording, e.g. don't add compression, if needed > do this for your classical car collection only, not to archive > LPs. That would almost work for me. I tried many click, crackle and noise correction methods. I couldn't get any of that in Gnome Wave Cleaner (GWC) to produce a result where I could tolerate the fidelity reduction to the music. But, the unwanted audio did decrease substantially, even at low modification settings. After a night long of fiddling, I opted to train the noise reduction in Audacity at the LP lead-out and use that to reduce crackle in the inter-song track regions. I used Audacity's declick feature, but only on actual clicks, not on the whole audio file. I'm quite please with the results of my first LP ripping. Thanks for the advice and links folks! -- Kevin From ralf.mardorf at alice-dsl.net Tue Jul 9 06:15:03 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Tue, 09 Jul 2013 08:15:03 +0200 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: <51DA9444.5060207@wootangent.net> References: <1373234414.663.4.camel@archlinux> <51DA9444.5060207@wootangent.net> Message-ID: <1373350503.681.61.camel@archlinux> On Mon, 2013-07-08 at 20:28 +1000, Leigh Dyer wrote: > On 8/07/13 8:00 AM, Ralf Mardorf wrote: > > On Sun, 2013-07-07 at 14:50 -0700, J. Liles wrote: > >> http://youtu.be/GVm5Jd1WDWw > > > > Can't be played with Firefox 22.0. > > Youtube usually does transcode videos to WebM format for playback on > Firefox, etc. using only free codecs. I can't say why Youtube hasn't > done that yet for this video; it could just be that it takes a while for > the WebM transcodes to happen And you're right. Now Firefox is able to play it too. From julien at mail.upb.de Tue Jul 9 09:12:40 2013 From: julien at mail.upb.de (Julien Claassen) Date: Tue, 9 Jul 2013 11:12:40 +0200 (CEST) Subject: [LAU] Debut prog-rock album, all produced with Open Source! In-Reply-To: <20130708195051.GL32655@deus.net> References: <20130708195051.GL32655@deus.net> Message-ID: Hello Ben! first of all, I'd like to say thanks for sharing these samples with us. I like them. I must admit, that the snipet around 55 seconds reminds me of "Cruel Summer". When you said, that this album takes us through quite a bit, I'd suspected something with even more different sections, but as you said, this is only a sampler and I can see, that this might really be worth to listen to as a whole. I'll see, if I can arrange to get it. I'm sure, this will be possible. It is all well played. I like the instruments and playing itself. Your singer is really good! I like her. The guitarist too appears to have his/her style. It's alwyas good, especially with a debut, to find musicians, who have found their own character. the only thing, that is still really young about this album is the mix. It's good and well balanced. I can hear everything and nothing disturbs me. Only the drum mix sounds rather nondescript. From personal experience I know, that it's a hassle to get a really good and characteristic sound out of a drumkit and still be able to integrate it with a full arrangement, so chapo to that as well. I'm sure such things will devfelop over time. Once you know, that you can do it, you get more and more creative and have more and more concrete ideas on how to evolve the sound into something very personal. the china already has that bit of "personality". It sounded rather strange to me the first time around, but now, that I've acquainted myself with it a little more, I hear it as it is and I hear it as something particular to your sound. Musically I already said, that a couple of sections sound well-connected harmonically. Since I like your chord progressions, that is no problem for me. :-) Over all the walk on the edge between slightly symphonic and band-oriented is well done. If I had to go into comparisons, I'd say on the line between Flower Kings an Spock's Beard. I had a very intense Flower Kings moment starting around 4:32min. As a last remark I want to add, that I also enjoyed the balance between harmonies and melodies. Both have their very distincitve moments and leave their traces in my ears, thus making for an enjoyable and capturing ride. I aprove. :-) Good job to all, who were involved! Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From bjb-linux-audio-user at deus.net Tue Jul 9 09:46:27 2013 From: bjb-linux-audio-user at deus.net (Ben Bell) Date: Tue, 9 Jul 2013 10:46:27 +0100 Subject: [LAU] Debut prog-rock album, all produced with Open Source! In-Reply-To: References: <20130708195051.GL32655@deus.net> Message-ID: <20130709094627.GA17544@deus.net> > first of all, I'd like to say thanks for sharing these samples with us. > I like them. I must admit, that the snipet around 55 seconds reminds me > of "Cruel Summer". Hah! I had to google that, but it amused me. Maybe I should add Bananarama to our list of influences? ;) > Only the drum mix sounds rather nondescript. From personal experience Yes, drums are the hardest thing and I'm very definitely still learning there. The drummer also used a borrowed (my) kit rather than his own for the recordings. I've promised him that next time we'll track using his own kit :) > into something very personal. the china already has that bit of > "personality". It sounded rather strange to me the first time around, but Hah, yes. That china is quite distinctive, but sometimes I think it needs to be place in the next room ;) > I'd say on the line between Flower Kings an Spock's Beard. That's interesting. I think the rest of the band would be surprised by that but the two tracks I used as reference recordings when mixing and mastering were "In The Eyes Of The World" and "On A Perfect Day". I kid ye not! Thank you very much for the detailed feedback :) Ben From seablaede at gmail.com Tue Jul 9 12:53:05 2013 From: seablaede at gmail.com (Thomas Vecchione) Date: Tue, 9 Jul 2013 08:53:05 -0400 Subject: [LAU] Non Mixer Spatializer Demo In-Reply-To: References: Message-ID: Ignoring the flash non-flash conversation.... This is an impressive demo, I do like what I am seeing. Thanks! Seablade On Sun, Jul 7, 2013 at 5:50 PM, J. Liles wrote: > As many of you already know, Non Mixer (http://non.tuxfamily.org) has > since its inception provided some cushy features for dealing with > Ambisonics mixes. > > Lately I've been working on extending these features to the next level. > > This is a quick demonstration of the new Spatializer module and the > associated Spatialization Console. What this does is provide synthetic > distance cues as well as a cushy interface for placing sounds in > virtual space. > > http://youtu.be/GVm5Jd1WDWw > > I'm hoping to have these new features ready for testing soon--but my > free time is very limited. > > As always, donations are welcome and very much appreciated. > (http://non.tuxfamily.org/wiki/Donations) > > A note about the video: > > Every time I try to make one of these screencasts, I run into the same > problem: nothing works. Thus, I had to aim my video camera at the > screen and record audio from my computer via line-out. This is > problematic for a number of reasons. 1) I can only record stereo this > way 2) my camera only samples audio at 32KHz and 3) When youtube > transcodes the video it adds an annoying click about once per second > (I believe this is due to a mismatch between the camera's framerate > and youtube's expectations. I would love to provide a theora/vorbis > screencapture video instead, but, alas, I cannot find any tools that > can capture screen activity and record audio via JACK in sync. Anyway, > the poor quality of the video is not due to a lack of effort. > _______________________________________________ > 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 abhayadevs at gmail.com Wed Jul 10 04:07:29 2013 From: abhayadevs at gmail.com (Abhayadev S) Date: Wed, 10 Jul 2013 09:37:29 +0530 Subject: [LAU] My first mix - for review Message-ID: Hi All, I am newbie interested in recording, mixing & mastering. The links are of my first mix, where i tried to mix and already recorded & over-dubbed (base & lead) electric guitar track with a Hydrogen programmed (by me) drum. I have put guitar tracks (original and in the mix stem) separately for easy review. I used, Ubuntu Studio 13.04 Ardour 3, CLAF (Compressor) and EQ10Q in the guitar/drum tracks and in master bus CALF reverb in the FX bus for both guitar and drum. Scarllet 8i6 tracks in soundcloud, http://soundcloud.com/abhayadevs/venu-electric-guitar-mixdown-r2 - ROCK-ishmix http://soundcloud.com/abhayadevs/venu-electric-guitar-166bpm-original - original guitar stem https://soundcloud.com/abhayadevs/venu-electric-guitar-stem-r1 - guitar stem in the mix http://soundcloud.com/abhayadevs/venu-electric-guitar-mixdown-r1 - drum style 1 looking forward to hear your reviews. Regards, ?Abhay? -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Wed Jul 10 13:28:06 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Wed, 10 Jul 2013 15:28:06 +0200 Subject: [LAU] carla & TAP plugins Message-ID: Hi, Carla is a great piece of software - thank you for that. I have troubles with some TAP plugins (TAP equalizer, pitchshifter), i did not tested all of TAP, only what I need. when loading these plugins carla falls with msg in console: Aborted (core dumped) can I do something with that ? thanks fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Wed Jul 10 13:35:13 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Wed, 10 Jul 2013 15:35:13 +0200 Subject: [LAU] carla & TAP plugins In-Reply-To: References: Message-ID: <51DD6311.4050101@autostatic.com> On 07/10/2013 03:28 PM, Fero Kiraly wrote: > Hi, > Carla is a great piece of software - thank you for that. > I have troubles with some TAP plugins (TAP equalizer, pitchshifter), i did > not tested all of TAP, only what I need. > > when loading these plugins carla falls with msg in console: > > Aborted (core dumped) > > can I do something with that ? > > thanks > fero Hi Fero, You can reach the author of Carla on the LinuxMusicians.com forum: http://forum.linuxmusicians.com/index.php Afaik he doesn't read this mailinglist on a regular basis while on the forum he'll reply promptly. Regards, Jeremy From jjbenham at chicagoguitar.com Wed Jul 10 14:32:10 2013 From: jjbenham at chicagoguitar.com (Jeremiah Benham) Date: Wed, 10 Jul 2013 09:32:10 -0500 Subject: [LAU] Denemo 1.0.4 released! Message-ID: Denemo version 1.0.4 has been released. Denemo is a program for creating music notation. http://www.denemo.org/ The music is displayed as music notation while typing or playing in and is simultaneously typeset via LilyPond typesetter. It can also be played via internal synthesizer. New features in this version: *Playing back repeats in MIDI **Da Capo and Dal Segno also supported *Double-Clicking for Help **Explore the object at the cursor *Learner mode **key presses are shown as you make them **the command and its tooltip is explained *Note Entry by PC-keyboard Improved **The duration keys sound the note entered **They simultaneous set the prevailing duration *Audio Recording **Export to .ogg or .wav files **Live performance also recorded *Translations **Italian Complete **French Complete **Czech Complete **A few strings still not offered to the translators *Bug Fixes **Layout blocks properly supported **Mouse clicking positioning **Accessing the space above the top staff Known Issues for this version: Windows users should reboot after installing Here are the compressed sources (from a mirror) : http://ftpmirror.gnu.org/denemo/denemo-1.0.4.tar.gz If automatic redirection fails, the list of mirrors is at: http://www.gnu.org/order/ftp.html Or if need be you can use the main GNU ftp server: http://ftp.gnu.org/gnu/denemo/denemo-1.0.4.tar.gz Windows Binary: http://denemo.org/downloads/denemo-1.0.4-0.mingw.exe Linux Binary: http://denemo.org/downloads/denemo-1.0.4-0.linux-x86 Mac: http://denemo.org/downloads/denemo-1.0.4-0.darwin-x86.tar.bz2 -------------- next part -------------- An HTML attachment was scrubbed... URL: From falktx at gmail.com Wed Jul 10 16:01:32 2013 From: falktx at gmail.com (Filipe Coelho) Date: Wed, 10 Jul 2013 17:01:32 +0100 Subject: [LAU] carla & TAP plugins In-Reply-To: References: Message-ID: <51DD855C.5050503@gmail.com> On 07/10/2013 02:28 PM, Fero Kiraly wrote: > Hi, > Carla is a great piece of software - thank you for that. > I have troubles with some TAP plugins (TAP equalizer, pitchshifter), i > did not tested all of TAP, only what I need. > > when loading these plugins carla falls with msg in console: > > Aborted (core dumped) > > can I do something with that ? > Hi there, can you give me information on how you build&run Carla please? If you build from source, you can use: make clean make debug gdb --args python3 ./src/carla.py and try to get a backtrace on gdb. In any case, please report any issues to https://github.com/falkTX/Carla/issues From fero.kiraly at gmail.com Wed Jul 10 16:07:12 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Wed, 10 Jul 2013 18:07:12 +0200 Subject: [LAU] carla & TAP plugins Message-ID: I built Carla from AUR. have I try to build it from source ? fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From malnourite at gmail.com Wed Jul 10 16:46:55 2013 From: malnourite at gmail.com (J. Liles) Date: Wed, 10 Jul 2013 09:46:55 -0700 Subject: [LAU] carla & TAP plugins In-Reply-To: References: Message-ID: On Wed, Jul 10, 2013 at 6:28 AM, Fero Kiraly wrote: > Hi, > Carla is a great piece of software - thank you for that. > I have troubles with some TAP plugins (TAP equalizer, pitchshifter), i did > not tested all of TAP, only what I need. > > when loading these plugins carla falls with msg in console: > > Aborted (core dumped) > > can I do something with that ? > > thanks > fero > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > FYI, some of the TAP plugins (TAP EQ in particular) are not RT safe (so an RT host shouldn't even allow you to load them). -------------- next part -------------- An HTML attachment was scrubbed... URL: From falktx at gmail.com Wed Jul 10 21:09:04 2013 From: falktx at gmail.com (Filipe Coelho) Date: Wed, 10 Jul 2013 22:09:04 +0100 Subject: [LAU] carla & TAP plugins In-Reply-To: References: Message-ID: <51DDCD70.9000102@gmail.com> On 07/10/2013 05:07 PM, Fero Kiraly wrote: > I built Carla from AUR. have I try to build it from source ? No need, I was able to reproduce the bug and fix it. I'll make a new bugfix release very soon From fedelogy at gmail.com Wed Jul 10 21:56:29 2013 From: fedelogy at gmail.com (Federico Bruni) Date: Wed, 10 Jul 2013 23:56:29 +0200 Subject: [LAU] ffado-diag: FireWire kernel stack not present Message-ID: Hi there I'm trying to connect my Esi Quatafire to a new laptop which doesn't have any firewire port. I'm using the DisplayPort (known also as Thunderbolt) and an adapter (?) to firewire. If I run ffado-diag (full log attached), it says: === REPORT === FireWire kernel drivers: FireWire kernel stack not present. Please compile the kernel with FireWire support. I'm on debian sid. I have libraw1394-11 version 2.1.0 I wonder if DisplayPort works on Linux... Thanks for your help Federico -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ffado-diag.log Type: application/octet-stream Size: 6490 bytes Desc: not available URL: From julien at mail.upb.de Thu Jul 11 13:00:58 2013 From: julien at mail.upb.de (Julien Claassen) Date: Thu, 11 Jul 2013 15:00:58 +0200 (CEST) Subject: [LAU] New piano Jazz: the Erfurt Sketches Message-ID: Hello Everyone! This has been lying around, completely hidden for over a year now. So I finally release the whole bunch, just as rough and edgey as they are. http://juliencoder.de/nama/erfurt_sketches/ the Erfurt Sketches was mainly written/improvised in Erfurt and mostly about it. Thanks to my kind host and pianist zappa, who taught me my first ever jazz lesson. If you ever happen to come the Erfurt way, see, if he's got a concert. I hope you enjoy these small impressions and pictures. If you feel like feedback, that's appreciated. :-) Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From jannis_achstetter at web.de Thu Jul 11 16:00:54 2013 From: jannis_achstetter at web.de (Jannis Achstetter) Date: Thu, 11 Jul 2013 18:00:54 +0200 Subject: [LAU] ffado-diag: FireWire kernel stack not present In-Reply-To: References: Message-ID: <51DED6B6.4090906@web.de> Hi, Am 10.07.2013 23:56, schrieb Federico Bruni: > Hi there > > I'm trying to connect my Esi Quatafire to a new laptop which doesn't > have any firewire port. I'm using the DisplayPort (known also as > Thunderbolt) and an adapter (?) to firewire. DisplayPort and Thunderbolt are two different things. Displayport works under Linux but IMO only for displays (hence the name?). Thunderbolt doesn't work at all. At least from what I read about it lately. Best regards, Jannis From harryhaaren at gmail.com Thu Jul 11 16:52:46 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Thu, 11 Jul 2013 17:52:46 +0100 Subject: [LAU] OpenAV : Fabla Message-ID: Hey everybody, Its my pleasure to announce that the next OpenAV Productions LV2 plugin is finished! Its called Fabla, and its a performance sampler. Page: http://openavproductions.com/fabla Demo reel: https://vimeo.com/70122957 One year from today Fabla will be released, and each donation motivates the release to be one month earlier. -Harry PS: The release system is the same as the previous OpenAV release for Sorcer, details available here: http://openavproductions.com/support -------------- next part -------------- An HTML attachment was scrubbed... URL: From willgodfrey at musically.me.uk Thu Jul 11 21:53:07 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Thu, 11 Jul 2013 22:53:07 +0100 Subject: [LAU] Slightly O/T was: linux-rt & presonus 1818VSL In-Reply-To: <51D4289B.80406@autostatic.com> References: <51D3D618.2@autostatic.com> <20130703142644.2330f647@debian> <20130703142754.04a6d90d@debian> <51D4289B.80406@autostatic.com> Message-ID: <20130711225307.5265e8f7@debian> On Wed, 03 Jul 2013 15:35:23 +0200 Jeremy Jongepier wrote: > On 07/03/2013 03:27 PM, Will Godfrey wrote: > >> > > >> > Anyone got an idiot's guide for compiling a suitable kernel for debian squeeze? > >> > I don't really have any problems but feel sure I could get more out of this > >> > beast (a quad core AMD), and might get slightly better timing. > > > > Oops! > > Not Squeeze, but Wheezy :o > > Hi Will, > > http://wiki.linuxaudio.org/wiki/system_configuration#build_your_own_real-time_kernel_on_debian_wheezy_or_later > > Regards, > > Jeremy Sorry I'm slow responding. Thanks for this, but when I started to look at it I saw a bit about modifying config. I had a look at the file and found it totally incomprehensible. I think I'll have to stay with things as they are :( -- 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 Thu Jul 11 22:51:48 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Fri, 12 Jul 2013 00:51:48 +0200 Subject: [LAU] zynaddsubfx as dssi Message-ID: <51DF3704.1020305@youmail.dk> Hi I'm trying to get zynaddsubfx going as dssi plugin in renoise on crunchbang (Debian Wheezy). I downloaded the source ZynAddSubFX-2.4.3.tar.bz2, installed the dependencies, ran cmake and ccmake (switching DSSI on) as advised in doc/build.txt. Everything seems to build just fine, running make install places the dssi plugin in /usr/local/lib/dssi: atte at blokhus:~/software/zynaddsubfx/ZynAddSubFX-2.4.3$ ll /usr/local/lib/dssi/libzynaddsubfx_dssi.so -rw-r--r-- 1 root staff 3.6M Jul 12 00:35 /usr/local/lib/dssi/libzynaddsubfx_dssi.so However zynaddsubfx doesn't show up in renoise. I then tried symlinking to /usr/lib/dssi (maybe renoise forgot to look in /usr/local) atte at blokhus:~$ ll /usr/lib/dssi/ total 1.1M drwxr-xr-x 2 root root 4.0K Jun 5 10:21 calf -rw-r--r-- 1 root root 955K May 17 2012 calf.so drwxr-xr-x 2 root root 4.0K Jun 5 09:57 fluidsynth-dssi -rw-r--r-- 1 root root 22K Jul 19 2011 fluidsynth-dssi.so -rw-r--r-- 1 root root 8.1K Dec 3 2011 karplong.so drwxr-xr-x 2 root root 4.0K Jun 5 10:19 less_trivial_synth -rw-r--r-- 1 root root 12K Dec 3 2011 less_trivial_synth.so lrwxrwxrwx 1 root root 42 Jul 12 00:48 libzynaddsubfx_dssi.so -> /usr/local/lib/dssi/libzynaddsubfx_dssi.so drwxr-xr-x 2 root root 4.0K May 29 11:41 nekobee -rw-r--r-- 1 root root 74K Apr 14 2011 nekobee.so drwxr-xr-x 2 root root 4.0K Jun 5 10:19 trivial_sampler -rw-r--r-- 1 root root 11K Dec 3 2011 trivial_sampler.so -rw-r--r-- 1 root root 5.7K Dec 3 2011 trivial_synth.so Still no luck :-( Any ideas? Anybody using the dssi version of zynaddsubfx? -- Atte http://atte.dk http://modlys.dk From fedelogy at gmail.com Fri Jul 12 06:32:40 2013 From: fedelogy at gmail.com (Federico Bruni) Date: Fri, 12 Jul 2013 08:32:40 +0200 Subject: [LAU] ffado-diag: FireWire kernel stack not present In-Reply-To: <51DED6B6.4090906@web.de> References: <51DED6B6.4090906@web.de> Message-ID: 2013/7/11 Jannis Achstetter > Hi, > > Am 10.07.2013 23:56, schrieb Federico Bruni: > > Hi there > > > > I'm trying to connect my Esi Quatafire to a new laptop which doesn't > > have any firewire port. I'm using the DisplayPort (known also as > > Thunderbolt) and an adapter (?) to firewire. > > DisplayPort and Thunderbolt are two different things. Displayport works > under Linux but IMO only for displays (hence the name?). Thunderbolt > doesn't work at all. At least from what I read about it lately. > I see, but the port is the same (actually it's mini DisplayPort). If thunderbolt will be supported in linux one day, I should be able to use it. But I guess that I'll need a USB2 audio interface for this laptop.. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Fri Jul 12 07:46:04 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 12 Jul 2013 09:46:04 +0200 Subject: [LAU] Slightly O/T was: linux-rt & presonus 1818VSL In-Reply-To: <20130711225307.5265e8f7@debian> References: <51D3D618.2@autostatic.com> <20130703142644.2330f647@debian> <20130703142754.04a6d90d@debian> <51D4289B.80406@autostatic.com> <20130711225307.5265e8f7@debian> Message-ID: <51DFB43C.6000101@autostatic.com> On 07/11/2013 11:53 PM, Will Godfrey wrote: > Sorry I'm slow responding. > > Thanks for this, but when I started to look at it I saw a bit about modifying > config. I had a look at the file and found it totally incomprehensible. I think > I'll have to stay with things as they are :( Hello Will, Don't worry about the kernel config file, you barely have to touch it. And when you have to do so there are menu driven tools to do this (make menuconfig/xconfig/gconfig). And then there are very few options you'd like to adjust to begin with like the kernel timer setting and the preemption model setting. I'll add it to the Wiki. Jeremy From p8rpp at aol.com Fri Jul 12 15:24:33 2013 From: p8rpp at aol.com (Peter P.) Date: Fri, 12 Jul 2013 17:24:33 +0200 Subject: [LAU] Call for Applications: IEM Graz Music Residency Program 2014 Message-ID: <20130712152432.GF23694@aol.de> IEM Music Residency Program 2014 - Call for Applications (please distribute) The Institute of Electronic Music and Acoustics (IEM) in Graz, Austria is happy to announce its new Music Residency program and invites applications from artists, scholars and practitioners for the 2014 edition. The Residency is aimed at individuals wishing to pursue projects in performance, composition, installation and sound art, development of tools for art production and related areas. Duration of residency: 5 months Start date: January 1st 2014 (negotiable) Monthly salary: Approx. EUR 1100 (net) APPLICATION DEADLINE: 1st of September 2013 23:59 CEST The Institute of Electronic Music and Acoustics is a department of the University of Music and Performing Arts Graz and was founded in 1965. An internationally well reknown institution in its field, it currently holds a staff of more than 25 members. IEM offers education to students in composition and computer music, sound engineering, contemporary music performance and musicology. It is well connected to the University of Technology as well as to the University of Graz through two joint study programs. The artwork produced at IEM is released through the Institute's own OpenCUBE and Signale concert series, as well as through various collaborations with international artists and institutions. IEM's main activities are centered around the three main research areas Computer Music Artistic Research Signal Processing and Acoustics and contributions to these, or any other field of relevance, are invited for the Music Residency program. What we expect from applicants: -An outstanding project proposal that adds new perspectives to the Institute's activities and resonates well with the interests of IEM members and students. -The succesful applicant will work on-site in Graz for the major part of the Residency. -Willingness to exchange and share ideas, knowledge and results with IEM staff members and students and engage in scholarly discussion. -The ability to work independently within the Institute as well as a strong initiative to integrate into the faculty life. -A dissemination strategy as part of the project proposal that ensures the publication of the work, or documentation thereof, in a suitable high-quality format. This could be achieved for example through the release of media, journal or conference publication, a project website or other means that help to preserve the knowledge gained through the Music Residency and make it available to the public. What we offer to the successful applicant: -Support in artistic production and arts-based research -Exchange with competent and experienced staff members -Work and office space -Contact with peers from similar or other disciplines -Infrastructure (electroacoustic music studios, icosahedral loudspeaker array, motion capture technology) -Concert and presentation facilities (CUBE 24 channel loudspeaker concert space) -Existing networks with local/international partners -We provide a monthly salary of approx. EUR 1100 net per month in addition to health and accident insurance. An application form providing more information is available at http://iem.at/~residency/applicationForm-IEM-MusicResidency2014.odt or http://iem.at/~residency/applicationForm-IEM-MusicResidency2014.docx Feel free to contact residency at iem.at with questions. kind regards Peter Plessas IEM Graz From atte at youmail.dk Fri Jul 12 16:44:22 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Fri, 12 Jul 2013 18:44:22 +0200 Subject: [LAU] New/old album online In-Reply-To: References: Message-ID: <51E03266.9000602@youmail.dk> On 06/26/2013 08:37 PM, Julien Claassen wrote: > Hello everyone! Hi I didn't have time to listen just yet, but wanted to make a few suggestions already: 1) I think a single download of the entire album would make it much more accessible. 2) I would prefer file names like 01_music_for_a_new_world.mp3 and not music.mp3 3) I think you should tag all files (I only downloaded the mp3 files), with the same album and same arist, remember track number to have the album play correctly in various players. Thanks for sharing! -- Atte http://atte.dk http://modlys.dk From julien at mail.upb.de Fri Jul 12 16:56:09 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 12 Jul 2013 18:56:09 +0200 (CEST) Subject: [LAU] New/old album online In-Reply-To: <51E03266.9000602@youmail.dk> References: <51E03266.9000602@youmail.dk> Message-ID: Hello atte! thanks for the suggestions. I've had part of those before. at least my latest EP has tracknumbers in the filenames. I'll do that with the next album. I'm looking forward to hear your feedback about the music. Careful: it's quite varied from song to song. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From rustys.lists at gmail.com Fri Jul 12 17:38:44 2013 From: rustys.lists at gmail.com (Rusty Perez) Date: Fri, 12 Jul 2013 10:38:44 -0700 Subject: [LAU] maximum input level, or normalization and dc offset correction? Message-ID: Hi list, I know this is a contravercial subject, but I'd like to hear some viewpoints. I use a delta 1010lt card. I'm wondering if I should carefully set my input levels for each input source, or, if the digital convenience of normalizing and dc ofset makes up for level defficiencies, without adding noise. I know the concern here is signal to noise ratio. Would my delta add noise if I boosted input levels initially or, in normalizing, am I just boosting the audio with the snr which I would have recordedif I had boosted my input levels in the first place? Does that make sense? Thanks! Rusty From ralf.mardorf at alice-dsl.net Fri Jul 12 17:59:26 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 19:59:26 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: Message-ID: <1373651966.697.142.camel@archlinux> Forget about the theory and listen, after that decide what to do! Before the digital domain becomes important, you already have to think about the optimal operating point of the analog side ;). What do you want to archive? "Creative work" or a "test-signal by the best possible way"? What is more important for you? Do you have a very good creative work to archive? Do you have crap to archive, that only could impress by absolutely optimized sound quality? More: In what context? Best leveling for source A and for source B vs the best leveling for a situation, when you want to play source A and B by a package. IOW the needed dynamic for a motion picture is different to the needed dynamic of a commercial. If there's a package of commercials and a motion picture you want to provide, then each has to fit to the other. Resume: Your question already does include the answer! You don't hear a difference, but you want to know, what is the so called best! Regarding to what? I switched my job from audio engineer to child care, since I can't stand that audio idiocy anymore. If you want to be hip and want to get money for your work, than keep a whole recording at 0 dBFS. If you want to do something that has to do with real art, than take an educated guess and answer yourself. You don't hear a difference and have to ask what's better? Why? We seem to live in an age where recording quality is the best, but the things that are recorded are crap. IMO it's better to have a less good recording quality, but with better stuff that is recorded. From ralf.mardorf at alice-dsl.net Fri Jul 12 18:17:26 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 20:17:26 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: Message-ID: <1373653046.697.150.camel@archlinux> PS: At what sample rate and what bit depth? Who will listen? Bats, humans, god? 48 Khz 32bit float is what you need, perhaps less bit, but do you get audio jitter or any other issues? For your 48 KHz x bit don't care too much about the optimized leveling for the analog and digital side, it's likely more safe to keep the digital side much below -3 dBFS, than to optimize it to 0 dBFS. From wizardofgosz at gmail.com Fri Jul 12 18:17:52 2013 From: wizardofgosz at gmail.com (Ricardus Vincente) Date: Fri, 12 Jul 2013 14:17:52 -0400 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: Message-ID: <51E04850.7090207@gmail.com> On 07/12/2013 01:38 PM, Rusty Perez wrote: > Hi list, > I know this is a contravercial subject, but I'd like to hear some viewpoints. > > I use a delta 1010lt card. I'm wondering if I should carefully set my > input levels for each input source, or, if the digital convenience of > normalizing and dc ofset makes up for level defficiencies, without > adding noise. > > I know the concern here is signal to noise ratio. > Would my delta add noise if I boosted input levels initially or, in > normalizing, am I just boosting the audio with the snr which I would > have recordedif I had boosted my input levels in the first place? > > Does that make sense? > > Thanks! > Rusty This is not controversial. It's simply gain staging. Record fairly hot to tape, but don't clip. If you see red lights turn it down a bit. That said, recording at 24 bit leaves tons of resolution and you don't need to record quite as loudly as we did in the 16 bit ADAT days. I never normalize the tracks in the DAW when mixing. I can't do it with my analog machine, so I don't do it with my DAW. That said, if one or two tracks is recorded really low, I will boost them up to about the same levels as the other tracks. I only normalize after mastering. In other words, when the song is mixed and mastered, the last step is normalizing. You might not want to normalize to zero, though. Some people believe in this new gremlin called inter-sample distortion or some such thing. Rich... From harryhaaren at gmail.com Fri Jul 12 18:36:58 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 12 Jul 2013 19:36:58 +0100 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <51E04850.7090207@gmail.com> References: <51E04850.7090207@gmail.com> Message-ID: On Fri, Jul 12, 2013 at 7:17 PM, Ricardus Vincente wrote: > Some people believe in this new gremlim called inter-sample distortion or some such thing. [Side note on gremlins] Inter sample peaks: neither peak represents the highest value of the wave: the peak exists between the samples. This is a mathematically proven phenomena. The distance to keep from 0dB FS depends on the signal (due to the inter-sample peaks depending on the signal). I tend to stay away 3dB from 0dBFS, I think that suffices... -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Fri Jul 12 18:38:27 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 12 Jul 2013 19:38:27 +0100 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: <51E04850.7090207@gmail.com> Message-ID: On Fri, Jul 12, 2013 at 7:36 PM, Harry van Haaren wrote: > Inter sample peaks: neither peak represents the highest value of the wave: Thinko / typo there.. neither *sample* represents the highest value, the peak is inbetween the *samples*. Sorry! -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 12 18:45:23 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 20:45:23 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? Message-ID: <1373654723.697.167.camel@archlinux> I guess it was off-list by accident?! -------- Forwarded Message -------- From: Ralf Mardorf To: Rusty Perez Subject: Re: [LAU] maximum input level, or normalization and dc offset correction? Date: Fri, 12 Jul 2013 20:40:23 +0200 On Fri, 2013-07-12 at 11:20 -0700, Rusty Perez wrote: > My question was really more of a question to understand what unwanted > noise I may be adding to my recording, which I'm not hearing, either > because of deficient equipment or deficient hearing. :-) A valid question, but don't care about the dynamic, the real issue with mastering music that should sound as good as possible on as much equipment as possible is the whole mix regarding to frequencies, phases vs mono and stereo. It's unlikely that noise will become an issue, it's more likely that the frequencies are biased or that there will be phasing between the channels. What faders do you use? The analog inputs have an optimized working point, perhaps the digital side then is too high or lower than "optimal". I don't know what positive or negative effects are caused by digital remachining. Keep the level below 0 dBFS, resp. add headroom. Too high in all cases is bad, but even a real too low level might be inaudible. "I never normalize the tracks in the DAW when mixing. I can't do it with my analog machine, so I don't do it with my DAW." - Ricardus Vincente and we can use the volume control of our amps to make the music louder and quieter. However, optimal leveling at recording time is better, than postprocessing. Analogy: If a recording does miss frequencies, you can't raise the missing frequencies, you only can rase frequencies that are there, but to silent, but this will come with side effects. From ralf.mardorf at alice-dsl.net Fri Jul 12 18:46:07 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 20:46:07 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: <51E04850.7090207@gmail.com> Message-ID: <1373654767.697.168.camel@archlinux> On Fri, 2013-07-12 at 19:36 +0100, Harry van Haaren wrote: > I tend to stay away 3dB from 0dBFS, I think that suffices... Me too, but to be fair, it's not really needed to level at -3 dBFS. From julien at mail.upb.de Fri Jul 12 18:55:22 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 12 Jul 2013 20:55:22 +0200 (CEST) Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: Message-ID: Hello Rusty! Here are my two pennies. In reasonable ranges this shouldn't make a difference. Especially if you work with 32-BIT float samples. No noise will be added by the normalisatin as such. So no artefacts of the process of amplification. The Delta is a good soundcard. I haven't had problems with it yet. Most of the noise, that I get is from my instruments. My DX7 has a rather low level, but all I can hear, is what I can already hear from the DX7, when plugging in the headphones. The same went for the clavinet. I had to amplify that by about 200-300% and the only real audible effect I got was lots of humming from the power source. It's not well insulated. In general, it is good to set your input levels on the delta rather close to what you need. Only take care, that you don't overdrive the input, because that can't be cleaned up later. With most of my keyboards between 0 and 3DB gain - as seen in alsamixer - is quite fine. Normalisation levels move around 110-120%, which I think is OK. This is quite beyond the point of normalisation in general. There are different views on that. DC offset is a thing quite apart from normalisation. I always apply DC-offset cleanup to every track, just in case. I don't know, which part of the equipment is most likely to course it, but I'm not aware of direct correction or quality facilities on the soundcard. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From grib at billgribble.com Fri Jul 12 18:59:03 2013 From: grib at billgribble.com (Bill Gribble) Date: Fri, 12 Jul 2013 14:59:03 -0400 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <51E04850.7090207@gmail.com> References: <51E04850.7090207@gmail.com> Message-ID: <1373655543.6087.52.camel@cayenne.local> On Fri, 2013-07-12 at 14:17 -0400, Ricardus Vincente wrote: > Record fairly hot to tape, but don't clip. If you see red lights turn > it down a bit. That said, recording at 24 bit leaves tons of resolution > and you don't need to record quite as loudly as we did in the 16 bit > ADAT days. I think you are underselling the benefits of 24-bit recording here. It's not necessary to record "hot" at all. AFAICT "best practice" with modern equipment is to basically work as if you were on an analog console; that is, clipping at no less than +20 dB with respect to your "0 dB" working level. That means you record with signals working at RMS around -20dBFS and then you just don't worry about clipping unless your signal is truly hugely dynamic, and in that case you make allowances as necessary with gain and/or limiting. I tend to believe anything Mike Rivers says, and his article on gain staging (http://mikeriversaudio.files.wordpress.com/2010/10/gainstructure.pdf) indicates that you really do need 20 dB of headroom over RMS to deal with recording real music (see the "Crest Factor" section in the piece mentioned above, which is mandatory reading in any case). 3dB corresponds to roughly 1 bit of amplitude, so working at -20dBFS means you are giving up about 6-7 bits of resolution to headroom, leaving something like 17 bits in your "working" range, plus whatever headroom you use for transients. In your final production step, during mastering, you will likely be exporting to 16-bit PCM for CD, so you aren't really giving up anything except the stress of constantly watching for overs. Practically, working this way in a DAW takes some getting used to, because the "waveform" display of your tracks at -20dBFS looks pretty much like a flat line. You don't get those pretty wave shapes until you zoom in. I have seen arguments, which I can neither refute nor support, that working within the last 1-2 db of your available dynamic range on input may compromise signal quality as well, as you are (for cheaply designed prosumer hardware) pushing against the voltage rails of the analog circuitry in front of the A/D and possibly running into current limitations. This is "plausible" but could also be total bunk, and I wouldn't take either side in a bar bet. Thanks, Bill Gribble From grib at billgribble.com Fri Jul 12 19:01:14 2013 From: grib at billgribble.com (Bill Gribble) Date: Fri, 12 Jul 2013 15:01:14 -0400 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <1373655543.6087.52.camel@cayenne.local> References: <51E04850.7090207@gmail.com> <1373655543.6087.52.camel@cayenne.local> Message-ID: <1373655674.6087.53.camel@cayenne.local> On Fri, 2013-07-12 at 14:59 -0400, Bill Gribble wrote: > 3dB corresponds to roughly 1 bit of amplitude, so working at -20dBFS > means you are giving up about 6-7 bits of resolution to headroom, > leaving something like 17 bits in your "working" range, plus whatever > headroom you use for transients. Oops, that was stupid. 6 dB corresponds to 1 bit, so you lose about 3 bits. Thanks, Bill Gribble From ralf.mardorf at alice-dsl.net Fri Jul 12 19:05:29 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 21:05:29 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: Message-ID: <1373655929.697.175.camel@archlinux> On Fri, 2013-07-12 at 20:55 +0200, Julien Claassen wrote: > reasonable ranges I agree! But it's hard to define "reasonable ranges" > With most of my keyboards between 0 and 3DB Not 0 to 3 dB, but -3 to 0 dBFS ;). For full-scale square -3 dB is headroom, for full-scale sine it's the max, so you should add some headroom, e.g. -6 dBFS or a limiter. From ralf.mardorf at alice-dsl.net Fri Jul 12 19:12:14 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 12 Jul 2013 21:12:14 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <1373655543.6087.52.camel@cayenne.local> References: <51E04850.7090207@gmail.com> <1373655543.6087.52.camel@cayenne.local> Message-ID: <1373656334.697.179.camel@archlinux> On Fri, 2013-07-12 at 14:59 -0400, Bill Gribble wrote: > -20dBFS Headroom depends to the kind of music and the kind of instrument. I agree that this is a good value for many recordings, while for less dynamically music or instruments -3 dBFS should be ok too, not to confuse with the sine vs square situation, regarding to this we need to add -3 dBFS, so at least -6dBFS are usefull, resp. for more daynamic caused by the music or instrument, yes, -20dBFS isn't unlikely, it's a valid headroom. From wizardofgosz at gmail.com Fri Jul 12 20:20:40 2013 From: wizardofgosz at gmail.com (Ricardus Vincente) Date: Fri, 12 Jul 2013 16:20:40 -0400 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: <51E04850.7090207@gmail.com> Message-ID: <51E06518.5000105@gmail.com> On 07/12/2013 02:36 PM, Harry van Haaren wrote: > > wrote: >> Some people believe in this new gremlim called inter-sample distortion > or some such thing. > > [Side note on gremlins] > Inter sample peaks: neither peak represents the highest value of the > wave: the peak exists between the samples. This is a mathematically > proven phenomena. The distance to keep from 0dB FS depends on the signal > (due to the inter-sample peaks depending on the signal). > > I tend to stay away 3dB from 0dBFS, I think that suffices... -Harry I understand that they can exist, but for the waveform to be rendered by the D/A, 3dB over the sample values (in the peake between the samples) seems like an unlikely transient. Further, if the D/A has sufficient headroom (modern good D/A should) I would think it's not really a problem. I still see plenty of mastering engineers normalizing their INCREDIBLY LOUD mixes to 0, and to -.1 So in short, I tend not worry about it. :-) From harryhaaren at gmail.com Fri Jul 12 20:34:27 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 12 Jul 2013 21:34:27 +0100 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <51E06518.5000105@gmail.com> References: <51E04850.7090207@gmail.com> <51E06518.5000105@gmail.com> Message-ID: On Fri, Jul 12, 2013 at 9:20 PM, Ricardus Vincente wrote: > Further, if the D/A has sufficient headroom (modern good D/A should) I would think it's not > really a problem. Found it! I was thinking of a presentation I'd seen somewhere, where the details of inter-sample peaks is properly explained: Slide 22 of Fons' EBU loudness meter presentaton (given at LAC2011 Maynooth, by Jorn Nettingsmeier): http://lac.linuxaudio.org/2011/download/lm-pres.pdf The wave is pretty much predictable: until a (strange) *double 1.0* value sample appears. The D/A conversion goes crazy, a spike to value of approx 3, while the range should be -1 to 1. The worse thing is, the longer this "predictable" part is, the *higher* the spike gets. > So in short, I tend not worry about it. :-) Me neither! Posted this for the sake of completeness. -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Fri Jul 12 20:46:11 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Fri, 12 Jul 2013 22:46:11 +0200 Subject: [LAU] My second album is done! In-Reply-To: References: Message-ID: <51E06B13.6030404@youmail.dk> On 07/05/2013 12:20 PM, Gabbe Nord wrote: > Thinking about it, I think you can achieve more or less the same result > by using a bus with a tremolo plugin on. Maybe worth checking out if > sidechaining per say does not work! Depends if the compression is following a fixed pattern (like a four-on-the-floor bass drumm) or not. Done more subtle, it can be a way of letting things like bass drum cut through the mix. -- Atte http://atte.dk http://modlys.dk From julien at mail.upb.de Fri Jul 12 21:08:23 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 12 Jul 2013 23:08:23 +0200 (CEST) Subject: [LAU] [a bit OT] Vocal microphone Message-ID: Hello! I need to get a vocal microphone. I've had this project several times before, but now I'm decided. There are a couple of alternatives. there is the Shure SM5x and there is the SE-220, if my memory doesn't betray me. Any more suggestions on the topic? I want to pay around 100-200 EUR maximum. This microphone will be meant solely for singing. No instruments to mic. Warm regards and sorry for the OT again Julien ---------------------------------------- http://juliencoder.de/nama/music.html From arnold at arnoldarts.de Fri Jul 12 21:24:24 2013 From: arnold at arnoldarts.de (Arnold Krille) Date: Fri, 12 Jul 2013 23:24:24 +0200 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: <20130712232424.4f969046@xingu.arnoldarts.de> Hi, top-reply because Julien likes that (I heard). There is a reason the SM58 is in every tour-case on every stage on every event in the last 20-30 years. Its a trusted old friend and can take the heat. If you want something a bit better, take the Beta58 which has a bit clearer sound. If you want your mic for the studio the choice is completely different: Get a large-diaphragm microphone, maybe the Studio Projects B1 (I got the little sisters C1 as a stereo-pair and like them), the Rode NT1 or an AKG in that range. You can also use these for micing traditional instruments (relatively close to cello or violin for example), for acoustic guitars and in (not direct) front of guitar-amp. Provided the room is nice too... Have fun, Arnold On Fri, 12 Jul 2013 23:08:23 +0200 (CEST) Julien Claassen wrote: > I need to get a vocal microphone. I've had this project several > times before, but now I'm decided. There are a couple of > alternatives. there is the Shure SM5x and there is the SE-220, if my > memory doesn't betray me. Any more suggestions on the topic? I want > to pay around 100-200 EUR maximum. This microphone will be meant > solely for singing. No instruments to mic. Warm regards and sorry for > the OT again Julien -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: not available URL: From jh at brainiac.com Fri Jul 12 21:31:53 2013 From: jh at brainiac.com (Joe Hartley) Date: Fri, 12 Jul 2013 17:31:53 -0400 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: <20130712173153.4872be06d0ee5974b1dcc320@brainiac.com> On Fri, 12 Jul 2013 23:08:23 +0200 (CEST) Julien Claassen wrote: > Hello! > I need to get a vocal microphone. I recently got a Blue Encore 200 mic for vocals after having a great experience using a friend's in a live setting. I like it quite a bit for vocals. It's a dynamic cardioid mic and needs phantom power. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From jostein at vait.se Fri Jul 12 21:33:49 2013 From: jostein at vait.se (Jostein Chr. Andersen) Date: Fri, 12 Jul 2013 23:33:49 +0200 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: <51E0763D.4080101@vait.se> Hi Julien, On 07/12/2013 11:08 PM, Julien Claassen wrote: > Hello! > I need to get a vocal microphone. I've had this project several times > before, but now I'm decided. There are a couple of alternatives. there > is the Shure SM5x and there is the SE-220, if my memory doesn't betray > me. Any more suggestions on the topic? I want to pay around 100-200 EUR > maximum. This microphone will be meant solely for singing. No > instruments to mic. sE X1 is a really good one and fantastic for the price. Here is links to the specification (PDF file) and the product page: http://www.seelectronics.com/static/downloads/cms/specs/sex1.pdf http://www.seelectronics.com/se-x1-microphone Jostein From leoave at gmail.com Fri Jul 12 22:07:50 2013 From: leoave at gmail.com (Leonardo Palomares) Date: Fri, 12 Jul 2013 15:07:50 -0700 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: Hi Julien: Vocals mic? For that money I would go with: Sennheiser Evolution e935 or the Audix OM7 is a better one. Speaking of Shure SM58, yes, it is the most popular, but not so good in other things. Most popular doesn't mean best (win/Linux) To get bass form them you have to eat it, that's the way it was design, for the rocker guy to sing touching it with the lips. To be honest I got better sound with the Senn e835 than the SM58. And I do like to be different, the rest uses SM58, I use e835. If I need a hammer, I buy a hammer, not a mic. :) Leo -------------- next part -------------- An HTML attachment was scrubbed... URL: From althompson58 at gmail.com Fri Jul 12 22:11:55 2013 From: althompson58 at gmail.com (Al Thompson) Date: Fri, 12 Jul 2013 18:11:55 -0400 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: <51E07F2B.30302@gmail.com> On 07/12/2013 06:07 PM, Leonardo Palomares wrote: > > If I need a hammer, I buy a hammer, not a mic. > > But, if you buy an SM58, you have a hammer AND a mic (and a wheel chock, and a doorstop, and much much more!). -- --- My bands, CD projects, music, news, and pictures: http://www.lateralforce.com My blog, with commentary on a variety of things, including audio, mixing, equipment, etc, is at: http://audioandmore.wordpress.com Staat hei?t das k?lteste aller kalten Ungeheuer. Kalt l?gt es auch; und diese L?ge kriecht aus seinem Munde: 'Ich, der Staat, bin das Volk.' - [Friedrich Nietzsche] From petecrighton at googlemail.com Sat Jul 13 00:54:18 2013 From: petecrighton at googlemail.com (Peter Crighton) Date: Sat, 13 Jul 2013 02:54:18 +0200 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: <20130712232424.4f969046@xingu.arnoldarts.de> References: <20130712232424.4f969046@xingu.arnoldarts.de> Message-ID: Although I am no expert on this and don?t have that much experience with it: I have a R?de NT1-A, and I do love and recommend it! -- Peter Crighton | Musician & Music Engraver based in Mainz, Germany http://www.petercrighton.de 2013/7/12 Arnold Krille > Hi, > > top-reply because Julien likes that (I heard). > > There is a reason the SM58 is in every tour-case on every stage on > every event in the last 20-30 years. Its a trusted old friend and can > take the heat. If you want something a bit better, take the Beta58 > which has a bit clearer sound. > > If you want your mic for the studio the choice is completely different: > Get a large-diaphragm microphone, maybe the Studio Projects B1 (I got > the little sisters C1 as a stereo-pair and like them), the Rode NT1 > or an AKG in that range. > > You can also use these for micing traditional instruments > (relatively close to cello or violin for example), for acoustic guitars > and in (not direct) front of guitar-amp. Provided the room is nice > too... > > Have fun, > > Arnold > > On Fri, 12 Jul 2013 23:08:23 +0200 (CEST) Julien Claassen > wrote: > > I need to get a vocal microphone. I've had this project several > > times before, but now I'm decided. There are a couple of > > alternatives. there is the Shure SM5x and there is the SE-220, if my > > memory doesn't betray me. Any more suggestions on the topic? I want > > to pay around 100-200 EUR maximum. This microphone will be meant > > solely for singing. No instruments to mic. Warm regards and sorry for > > the OT again Julien > > _______________________________________________ > 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 kevinc at cosgroves.us Sat Jul 13 01:16:43 2013 From: kevinc at cosgroves.us (Kevin Cosgrove) Date: Fri, 12 Jul 2013 18:16:43 -0700 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: Message-ID: <20130713011644.00914BE05B@joseph.cosgroves.us> Last year one of our Pickathon (shameless plug, pickathon.com) sound crew was asked to pick up an SM58 on the way to load-in for the weekend. He did that, but he also picked up a Sennheiser e840 ($130USD). There was no comparison. The Sennheiser was much clearer and was less prone to feedback than the SM58. The 58 went back to the store, and the e840 stayed up, along with the guy buying another e840. Next time I need a vocal mic I'll get an e945 ($220USD), a step up from the e840. I'm also familiar with Beyer Dymamics, which are fabulous, but about $300USD. The e840s have a bit more presence in the vocal sweet spot, and have a hotter signal by 6-10dB. The Beyer's are also much nicer sounding than the SM58. We also use AKG 330s, and the Sennheiser keeps up with them too. The advantage, as far as I can tell, of an SM58, is that a drunk singer can drop them from the stage onto concrete 2m below, and they're likely to still sound the same. Also, they don't sound all that bad to start with, and they're not too expensive ($100USD). For live sound there's a lot to be said for cheap and indestructible. In a studio someone might want a bit more. All the best.... On 12 July 2013 at 23:08, Julien Claassen wrote: > Hello! > I need to get a vocal microphone. I've had this project several times > before, but now I'm decided. There are a couple of alternatives. there is the > Shure SM5x and there is the SE-220, if my memory doesn't betray me. Any more > suggestions on the topic? I want to pay around 100-200 EUR maximum. This > microphone will be meant solely for singing. No instruments to mic. > Warm regards and sorry for the OT again > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Kevin From gheskett at wdtv.com Sat Jul 13 02:52:10 2013 From: gheskett at wdtv.com (Gene Heskett) Date: Fri, 12 Jul 2013 22:52:10 -0400 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: <20130713011644.00914BE05B@joseph.cosgroves.us> References: <20130713011644.00914BE05B@joseph.cosgroves.us> Message-ID: <201307122252.10886.gheskett@wdtv.com> On Friday 12 July 2013 22:29:50 Kevin Cosgrove did opine: > Last year one of our Pickathon (shameless plug, pickathon.com) > sound crew was asked to pick up an SM58 on the way to load-in for > the weekend. He did that, but he also picked up a Sennheiser > e840 ($130USD). There was no comparison. The Sennheiser was > much clearer and was less prone to feedback than the SM58. The > 58 went back to the store, and the e840 stayed up, along with the > guy buying another e840. Next time I need a vocal mic I'll get > an e945 ($220USD), a step up from the e840. > > I'm also familiar with Beyer Dymamics, which are fabulous, but > about $300USD. The e840s have a bit more presence in the vocal > sweet spot, and have a hotter signal by 6-10dB. The Beyer's are > also much nicer sounding than the SM58. We also use AKG 330s, > and the Sennheiser keeps up with them too. > > The advantage, as far as I can tell, of an SM58, is that a drunk > singer can drop them from the stage onto concrete 2m below, > and they're likely to still sound the same. Also, they don't > sound all that bad to start with, and they're not too expensive > ($100USD). For live sound there's a lot to be said for cheap and > indestructible. In a studio someone might want a bit more. > Indestructible is a pretty strong word. Quite a few years ago when I was the CE at a tv station, I'd convinced TPTB that the mic situation in news was untenable, and with a lot of yelling and screaming, got perms to by 6 of them, so we'd have one for each camera crew and one spare. Two things. Every time one of them was laid down while setting up the camera, it would leave, but a radio shack $20 lookalike would come home with the reporter & the reporters didn't know the diff & couldn't be bothered to learn. In 2 months we had one left and it was smashed as if a fire truck had run over it, about 1/2" thick. Maybe it did for all I know. One failed with no visible damage in a week, sent it back, they fixed it, we had the tape on the air about 10 days later when it went away again, sitting on a stand in full view of the rolling camera while interviewing a football coach. You have no clue how destructive a reporter, who is gung ho to get the story, alligators in the pool be damned, can be to even the relatively heavy SM58. Better than 80% of whats out there, at <50% of the price, but not indestructible. 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) My web page: is up! My views BOFH excuse #388: Bad user karma. A pen in the hand of this president is far more dangerous than 200 million guns in the hands of law-abiding citizens. From ralf.mardorf at alice-dsl.net Sat Jul 13 05:24:36 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 13 Jul 2013 07:24:36 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <51E06518.5000105@gmail.com> References: <51E04850.7090207@gmail.com> <51E06518.5000105@gmail.com> Message-ID: <1373693076.697.206.camel@archlinux> On Fri, 2013-07-12 at 16:20 -0400, Ricardus Vincente wrote: > On 07/12/2013 02:36 PM, Harry van Haaren wrote: > > > > wrote: > >> Some people believe in this new gremlim called inter-sample distortion > > or some such thing. > > > > [Side note on gremlins] > > Inter sample peaks: neither peak represents the highest value of the > > wave: the peak exists between the samples. This is a mathematically > > proven phenomena. The distance to keep from 0dB FS depends on the signal > > (due to the inter-sample peaks depending on the signal). > > > > I tend to stay away 3dB from 0dBFS, I think that suffices... -Harry > > I understand that they can exist, but for the waveform to be rendered > by the D/A, 3dB over the sample values (in the peake between the > samples) seems like an unlikely transient. Further, if the D/A has > sufficient headroom (modern good D/A should) I would think it's not > really a problem. > > I still see plenty of mastering engineers normalizing their INCREDIBLY > LOUD mixes to 0, and to -.1 And those mixes do sound good ;)? They mix music in a way that I can't stand to listen it. You like to listen to this music? > So in short, I tend not worry about it. :-) In the past I made many masterings with DAT having a margin of just 0.5, IOW -0.5 dBFS, when recording a live session without limiter even peaks that were > 0 dBFS don't cause audible effects. However, what do the meters show? Full-Scale Sine Wave, then the sine is at 0 dBFS or Full-Scale Square Wave, then the max for the sine should be at -3 dBFS. I really don't know what the meters do display. The sound of DAT recordings with peaks at -6 dBFS or lower -12 dBFS etc. don't sound less good, then recordings with peaks at -3 or 0 dBFS. It's nonsense to take care about "optimized" leveling, there simply should be headroom so that peaks won't reach 0 dBFS or -3 dBFS. As somebody already pointed out, e.g. -20 dBFS, then peaks anyway might reach 0 or -3 dBFS. For the mastering, when we know the peaks we don't need that much headroom, but a peak won't reach always the same level, sometimes it might be at -2.5 dBFS and sometimes at -3 dBFS, so minimal headroom is needed. Even if > 0 dBFS sometimes isn't audible, sometimes it does cause audible effects. Regarding to noise, as Julien already pointed out, the instrument or microphone does cause noise, you won't hear the noise of a good sound card. Julien mentioned the DX7, yes, my DX7 does produce audible noise too, but recording my Matrix-1000 is like recording a soft synth, there isn't noise. FWIW there is a reason that adding noise sometimes does improve the sound quality http://en.wikipedia.org/wiki/Dither : "Usage Dither should be added to any low-amplitude or highly-periodic signal before any quantization or re-quantization process, in order to de-correlate the quantization noise from the input signal and to prevent non-linear behavior (distortion); the lesser the bit depth, the greater the dither must be. The result of the process still yields distortion, but the distortion is of a random nature so the resulting noise is, effectively, de-correlated from the intended signal. Any bit-reduction process should add dither to the waveform before the reduction is performed." From ralf.mardorf at alice-dsl.net Sat Jul 13 05:29:30 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 13 Jul 2013 07:29:30 +0200 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: References: Message-ID: <1373693370.697.207.camel@archlinux> For studio and/or stage? From ralf.mardorf at alice-dsl.net Sat Jul 13 05:47:07 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 13 Jul 2013 07:47:07 +0200 Subject: [LAU] [a bit OT] Vocal microphone In-Reply-To: <201307122252.10886.gheskett@wdtv.com> References: <20130713011644.00914BE05B@joseph.cosgroves.us> <201307122252.10886.gheskett@wdtv.com> Message-ID: <1373694427.697.218.camel@archlinux> On Fri, 2013-07-12 at 22:52 -0400, Gene Heskett wrote: > Every time one of them was laid down while setting up the camera, it would > leave, but a radio shack $20 lookalike would come home with the reporter & > the reporters didn't know the diff & couldn't be bothered to learn. I own such a faked SM58, a Realistic Highball, costs >$20, perhaps >= $45, it does sound less good than the original, but even this fake is indestructible! Around 30 years in usage, it was broken to intermateable pieces and then pieced together, it's still ok. But the sound quality even of the original SM58 is minimal better than the sound of an elCheapo microphone from the supermarket. The fakes that came home don't had switches? My fake has got a switch. From fons at linuxaudio.org Sat Jul 13 11:00:08 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 13 Jul 2013 11:00:08 +0000 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: References: <51E04850.7090207@gmail.com> <51E06518.5000105@gmail.com> Message-ID: <20130713110008.GA30046@linuxaudio.org> On Fri, Jul 12, 2013 at 09:34:27PM +0100, Harry van Haaren wrote: > http://lac.linuxaudio.org/2011/download/lm-pres.pdf > > The wave is pretty much predictable: until a (strange) *double 1.0* value > sample appears. The D/A conversion goes crazy, a spike to value of approx > 3, while the range should be -1 to 1. This is of course an extreme example. In real life the soup isn't eaten that hot. I use dpl1 as a safety net for most of my work, usually it's set to -0.5 dB as the allowed peak level. 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 alice-dsl.net Sat Jul 13 11:10:03 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 13 Jul 2013 13:10:03 +0200 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <20130713110008.GA30046@linuxaudio.org> References: <51E04850.7090207@gmail.com> <51E06518.5000105@gmail.com> <20130713110008.GA30046@linuxaudio.org> Message-ID: <1373713803.700.5.camel@archlinux> On Sat, 2013-07-13 at 11:00 +0000, Fons Adriaensen wrote: > -0.5 dB Exactly the margin I preferred in the past with my AIWA and Sony DAT recorders, but I never noticed anything when I mastered exactly to 0.0 either, however often in the past I mastered between -0.5 and -1.5 or even -2 and today I master in a range from 0.0 to -3.0 and seldom even -6.0. There's nothing bad audible neither at 0.0 and also not if it's very low, e.g. -10.0, only the loudness does differ, at least for my hearing. From atte at youmail.dk Sat Jul 13 11:18:02 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Sat, 13 Jul 2013 13:18:02 +0200 Subject: [LAU] zynaddsubfx as dssi In-Reply-To: <51DF3704.1020305@youmail.dk> References: <51DF3704.1020305@youmail.dk> Message-ID: <51E1376A.6010403@youmail.dk> On 07/12/2013 12:51 AM, Atte Andr? Jensen wrote: > However zynaddsubfx doesn't show up in renoise. I then tried symlinking > to /usr/lib/dssi (maybe renoise forgot to look in /usr/local) Looked in the renoise Log.txt, which shows this: DSSI: Scanning '/usr/lib/dssi/libzynaddsubfx_dssi.so'... DSSI: Instantiate FAILED (failed to open the DLL file '/usr/lib/dssi/libzynaddsubfx_dssi.so: undefined symbol: instance_name')! A fellow renoise users, tried zyn/dssi under rosegarden and it reports the same. So it seems zyn is either not build correctly or not exactly ready for prime time as dssi. The later seems strange, though, since there's a package in debian (that has too-new dependencies for me to install): http://packages.debian.org/search?keywords=zynaddsubfx-dssi Any thoughts? -- Atte http://atte.dk http://modlys.dk From allcoms at gmail.com Sat Jul 13 11:35:58 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Sat, 13 Jul 2013 12:35:58 +0100 Subject: [LAU] zynaddsubfx as dssi In-Reply-To: <51E1376A.6010403@youmail.dk> References: <51DF3704.1020305@youmail.dk> <51E1376A.6010403@youmail.dk> Message-ID: I'm not a renoise user - I know it supports lxvst's but its news to me that it supports DSSI. IIRC the zyn DSSI port lacks a GUI, or at least it lacks a fully functional one. Real zyn also lacks a usable GUI IMO so I dumped it years ago for TAL Noizemaker (and Loomers Aspect) which are good examples of what a GUI synth should be. On Sat, Jul 13, 2013 at 12:18 PM, Atte Andr? Jensen wrote: > On 07/12/2013 12:51 AM, Atte Andr? Jensen wrote: > > However zynaddsubfx doesn't show up in renoise. I then tried symlinking >> to /usr/lib/dssi (maybe renoise forgot to look in /usr/local) >> > > Looked in the renoise Log.txt, which shows this: > > DSSI: Scanning '/usr/lib/dssi/libzynaddsubfx_**dssi.so'... > DSSI: Instantiate FAILED (failed to open the DLL file > '/usr/lib/dssi/libzynaddsubfx_**dssi.so: undefined symbol: > instance_name')! > > A fellow renoise users, tried zyn/dssi under rosegarden and it reports the > same. So it seems zyn is either not build correctly or not exactly ready > for prime time as dssi. The later seems strange, though, since there's a > package in debian (that has too-new dependencies for me to install): > > http://packages.debian.org/**search?keywords=zynaddsubfx-**dssi > > Any thoughts? > > > -- > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sat Jul 13 11:44:10 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 13 Jul 2013 11:44:10 +0000 Subject: [LAU] maximum input level, or normalization and dc offset correction? In-Reply-To: <1373655543.6087.52.camel@cayenne.local> References: <51E04850.7090207@gmail.com> <1373655543.6087.52.camel@cayenne.local> Message-ID: <20130713114410.GB30046@linuxaudio.org> On Fri, Jul 12, 2013 at 02:59:03PM -0400, Bill Gribble wrote: > AFAICT "best practice" with modern equipment is to basically work as if > you were on an analog console; that is, clipping at no less than +20 dB > with respect to your "0 dB" working level. That means you record with > signals working at RMS around -20dBFS and then you just don't worry > about clipping unless your signal is truly hugely dynamic, and in that > case you make allowances as necessary with gain and/or limiting. Agreed 100%. With 24 bits you have a lot of headroom, and any 'digital gain' used post-recording won't affect the real S/N ratio at all. A working level of -20 dB RMS FS will preserve the full dynamic range even if you have to boost the signal later. A high end preamp/AD such as the RME Micstasy will have a S/N ratio of around 120 dB maximum when configured for a high level line input. With typical gain settings for e.g. recording an acoustic instrument it will be around 100 dB. Most mic preamps will give you much less, and we're ignoring mic self noise and ambient noise. In many cases the latter will determine the real S/N ratio you can have. If you're recording electronic instruments they and the DI-box will determine the noise floor. It won't be near anything that 20 bits can't handle. Anyway, for music with a restricted dynamic range (that includes almost everything presented/discussed here) this is completely academic. You could record the tracks at -40 dB FS and nobody would notice. 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 Sat Jul 13 12:52:36 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Sat, 13 Jul 2013 14:52:36 +0200 Subject: [LAU] zynaddsubfx as dssi In-Reply-To: References: <51DF3704.1020305@youmail.dk> <51E1376A.6010403@youmail.dk> Message-ID: <51E14D94.8090708@youmail.dk> On 07/13/2013 01:35 PM, Dan MacDonald wrote: > I'm not a renoise user - I know it supports lxvst's but its news to me > that it supports DSSI. I got a bit further. It seems that it was a bug that was fixed, build from git, and now it runs fine. I cannot access any presets, though :-( > IIRC the zyn DSSI port lacks a GUI, or at least it lacks a fully > functional one.Real zyn also lacks a usable GUI IMO so I dumped it > years ago for TAL Noizemaker (and Loomers Aspect) which are good > examples of what a GUI synth should be. I bought aspect, and it's great. There are however sounds that zyn can do that you can't make in any other synth... -- Atte http://atte.dk http://modlys.dk From csanchezgs at gmail.com Sat Jul 13 19:50:28 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sat, 13 Jul 2013 21:50:28 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting Message-ID: Hello, dear folks. I'm still (constantly) tweaking my raspberry here and there when I can. I have some configurations/sessions that run ok, Looper (jack+sooperlooper), Looper+Fx (jack+sooperlooper+rakarrack/guitarrix). But I'm still having some annoying problems that wanted to share with you: 1) Simple session script, Looper, runs quite well when launched (via ssh -X or from console) logged as PI user. When I run it as a boot/init script (simple launcher script placed in /etc/init.d and then installed with "update-rc.d launcherscript defaults 99"), almost in every first start there's crackles and noise and my loops sound distorted, even with clean guitar; if I restart then it goes well (2nd or 3d time). 2) Next step forward I added Rakarrack in charge of guitar FX routed to the Looper. First, it happens almost same thing as mentioned about noise. Well, I guess It should have something to do with the signal path and things connected to the same power supplier, so I'll have to reserve some time to debug and switch off things until I get a cleaner signal and get rid of the hum and crackles. But the main issue is that there's problems with Rakarrack not been able to start because several problems with not enough Jack ports and Jack and RT. I didn't have any of these problems before. I got to solve the problem with ports changing -p16 to -p32 when launching jack+sooperlooper (-p24 is enough for just Rakarrack). Then I made a launcher script LSB compliant (service script placed in /etc/init.d and then installed with "update-rc.d servicescript defaults", with LSB flags and functions for "start" and "stop", as a proper newer Debian service, and configured to launch last in boot phase) which then call the Jack-config+Rakarrack+Sooperlooper session launcher script. I can see on screen in the init messages this error: "JACK is running in realtime mode, but you are not allowed to use realtime schedule..." But I know RT is already configured, so I guessed it was something to do with the boot/init stage configuration and environment. I tried using "su" with several different parameters to run the launcher script as "pi" user, as It runs quite well when logged as this user (calling just the script and even with "sudo /etc/init.d/script start"). So, maybe you can't get RT in init phase on RaspberryPI and you can only when you get to the login prompt and everything is loaded and ready. But I think there's a "Puredata on Raspberry PI" project that runs some scripts in the init stage; and it seems that in order to get a headless FX+looper station, running some configuration+apps-launcher at boot time is an obvious choice. I'm sure something escapes me after so much try-error-code cicles. Maybe you have some ideas. Thanks as always. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From jeremy at autostatic.com Sat Jul 13 19:59:50 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sat, 13 Jul 2013 21:59:50 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: References: Message-ID: <51E1B1B6.5010609@autostatic.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/13/2013 09:50 PM, Carlos sanchiavedraz wrote: > Hello, dear folks. > > I'm still (constantly) tweaking my raspberry here and there when I > can. I have some configurations/sessions that run ok, Looper > (jack+sooperlooper), Looper+Fx > (jack+sooperlooper+rakarrack/guitarrix). But I'm still having some > annoying problems that wanted to share with you: > > 1) Simple session script, Looper, runs quite well when launched > (via ssh -X or from console) logged as PI user. When I run it as a > boot/init script (simple launcher script placed in /etc/init.d and > then installed with "update-rc.d launcherscript defaults 99"), > almost in every first start there's crackles and noise and my loops > sound distorted, even with clean guitar; if I restart then it goes > well (2nd or 3d time). > Hi Carlos, Try resetting the JACK buffersize with jack_bufsize after starting your program. This does the trick for me on the RPi. > 2) Next step forward I added Rakarrack in charge of guitar FX > routed to the Looper. First, it happens almost same thing as > mentioned about noise. Well, I guess It should have something to do > with the signal path and things connected to the same power > supplier, so I'll have to reserve some time to debug and switch off > things until I get a cleaner signal and get rid of the hum and > crackles. But the main issue is that there's problems with > Rakarrack not been able to start because several problems with not > enough Jack ports and Jack and RT. I didn't have any of these > problems before. > > I got to solve the problem with ports changing -p16 to -p32 when > launching jack+sooperlooper (-p24 is enough for just Rakarrack). > > Then I made a launcher script LSB compliant (service script placed > in /etc/init.d and then installed with "update-rc.d servicescript > defaults", with LSB flags and functions for "start" and "stop", as > a proper newer Debian service, and configured to launch last in > boot phase) which then call the Jack-config+Rakarrack+Sooperlooper > session launcher script. I can see on screen in the init messages > this error: > > "JACK is running in realtime mode, but you are not allowed to use > realtime schedule..." > > But I know RT is already configured, so I guessed it was something > to do with the boot/init stage configuration and environment. I > tried using "su" with several different parameters to run the > launcher script as "pi" user, as It runs quite well when logged as > this user (calling just the script and even with "sudo > /etc/init.d/script start"). So, maybe you can't get RT in init > phase on RaspberryPI and you can only when you get to the login > prompt and everything is loaded and ready. But I think there's a > "Puredata on Raspberry PI" project that runs some scripts in the > init stage; and it seems that in order to get a headless FX+looper > station, running some configuration+apps-launcher at boot time is > an obvious choice. > > I'm sure something escapes me after so much try-error-code cicles. > Maybe you have some ideas. > http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-March/091750.html Regards, Jeremy > Thanks as always. > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with undefined - http://www.enigmail.net/ iQIcBAEBAgAGBQJR4bG2AAoJEF63J34m6LiUDq4P+QHsb/bODOeBtL0dQUvmNrD2 jXeYyU57kAF5Al5Nf7e8vtAKuGSsLuVhF30CYGtcM1VUHYT0PFVfaOkZxlQsLJ6W nM2w36ADyIEgboadWmS9DqoEP+L80XR6EpK9Bz1ZuMYgLsKg5eOP6m94v4rArsjQ KDZTEIB5S4d1F4ZvDHbAnH2vY2wlhaH7Ii0R8IKuNCZwJP3L96abayNB9AjA6Fzu SiKmjrnibujvonXOi3ew/86mES7AbOCJ9Bc4FnP+ZDlLMhvV3LGQQ5/lG5mLXW78 nqwQM958pZJrn7ZHgsU73mhi6i1iBP7hjGTtbdOd6NeUyNcWPeVnhJbHJAM9HCBE xs0NkNVoQJnmRrBnT8yYcrZY3Vk6+8dzC/XbRlo97yvfNixKHEfTaAY2aMptQuCu +zPURKNij22FHU1+r7IzZQR6zvoHIxN2zqs7k8677CQEzYFJ8OOcu3YiixmIhPll q9ZVTiy13fONzgdenhwY9Ow+26vtQF+HVcLt3d5WegaEjjkykxvX/6CLX0nLrlqz 3DwyQ0DD0Wk20DwejIsfVo0zDhmw9XKZY2UPX6zCm9p/JIuD/IxYq5mzvZXiGToS 5KTBrT2NAn3t03591M1HiekA8YEj8p4BHVOM8yVTxI3T5RSiT09kYSuKg2RpAfNJ tp6jusbkXuF0aEZav7K2 =hhBs -----END PGP SIGNATURE----- From atte at youmail.dk Sun Jul 14 09:44:06 2013 From: atte at youmail.dk (=?ISO-8859-1?Q?Atte_Andr=E9_Jensen?=) Date: Sun, 14 Jul 2013 11:44:06 +0200 Subject: [LAU] zynaddsubfx as dssi In-Reply-To: <51E14D94.8090708@youmail.dk> References: <51DF3704.1020305@youmail.dk> <51E1376A.6010403@youmail.dk> <51E14D94.8090708@youmail.dk> Message-ID: <51E272E6.9090107@youmail.dk> On 07/13/2013 02:52 PM, Atte Andr? Jensen wrote: > On 07/13/2013 01:35 PM, Dan MacDonald wrote: >> I'm not a renoise user - I know it supports lxvst's but its news to me >> that it supports DSSI. > > I got a bit further. It seems that it was a bug that was fixed, build > from git, and now it runs fine. I cannot access any presets, though :-( src/Misc/Config.cpp: if(cfg.bankRootDirList[0].empty()) { //banks cfg.bankRootDirList[0] = "~/banks"; cfg.bankRootDirList[1] = "./"; cfg.bankRootDirList[2] = "/usr/share/zynaddsubfx/banks"; cfg.bankRootDirList[3] = "/usr/local/share/zynaddsubfx/banks"; cfg.bankRootDirList[4] = "../banks"; cfg.bankRootDirList[5] = "banks"; } if(cfg.presetsDirList[0].empty()) { //presets cfg.presetsDirList[0] = "./"; cfg.presetsDirList[1] = "../presets"; cfg.presetsDirList[2] = "presets"; cfg.presetsDirList[3] = "/usr/share/zynaddsubfx/presets"; cfg.presetsDirList[4] = "/usr/local/share/zynaddsubfx/presets"; } It works if I place the banks in ~/banks. Unfortunately renoise crashes whith this https://dl.dropboxusercontent.com/u/4343030/Log.txt when shutting down a song that uses and instance of zynaddsubfx as dssi :-( -- Atte http://atte.dk http://modlys.dk From julien at mail.upb.de Sun Jul 14 19:37:46 2013 From: julien at mail.upb.de (Julien Claassen) Date: Sun, 14 Jul 2013 21:37:46 +0200 (CEST) Subject: [LAU] Fixing a MIDI issues Message-ID: Hello everyone! this is perghaps not straight on topic, but I couldn't find a satisfactory answer anywhere else. I'm having trouble with my JV-1080. Only every second note gets played. I checked different input devices and different cables, direct and indirect connections. It seems to be a problem of the Roland itself. Internal sounds, demos and the preview button, work fine. Any idea, what this might be? It looks so systematic, as if it might be by design, but I don't know any feature, which should allow for that. Sorry to spam the list and thanks for following me so far. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From ralf.mardorf at alice-dsl.net Sun Jul 14 20:13:29 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 14 Jul 2013 22:13:29 +0200 Subject: [LAU] Fixing a MIDI issues In-Reply-To: References: Message-ID: <1373832809.665.123.camel@archlinux> On Sun, 2013-07-14 at 21:37 +0200, Julien Claassen wrote: > I'm having trouble with my JV-1080. Only every second note gets > played. What's the setup? Is it simply master keyboard out to JV-1080 in? IIUC it's always, exactly "every second" note. What happens, if you send single controller steps and/or a single controller step and after that a note etc.? I guess you're playing each note by the same MIDI channel? Note off might vs Note on velocity 0 and/or a running status issue raise it's ugly head, IOW if the setup includes some bad programmed stopover between the master keyboard and the JV-1080, this program might misinterpret the MIDI data. You replaced the JV-1080 by another synth for the same setup? To ensure that there isn't something broken close to the MIDI input's opto-coupler, what happens, if you play a note and a second note 10 minutes later? From julien at mail.upb.de Sun Jul 14 20:29:12 2013 From: julien at mail.upb.de (Julien Claassen) Date: Sun, 14 Jul 2013 22:29:12 +0200 (CEST) Subject: [LAU] Fixing a MIDI issues In-Reply-To: <1373832809.665.123.camel@archlinux> References: <1373832809.665.123.camel@archlinux> Message-ID: Hello Ralf! the problem is solved. didn't the mail reach the list or was it after yo replied. Anyway: all is well. But I think you as well for trying to help. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From ralf.mardorf at alice-dsl.net Sun Jul 14 20:52:51 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 14 Jul 2013 22:52:51 +0200 Subject: [LAU] Fixing a MIDI issues In-Reply-To: References: <1373832809.665.123.camel@archlinux> Message-ID: <1373835171.665.144.camel@archlinux> If somebody should run into the same issue: "At least I guess you could get 2 or more and use them in stack mode to get stupidly high polyphony!!!" - http://www.dancecrave.com/jv1080.htm From mariusjanssen at gmx.de Sun Jul 14 21:34:08 2013 From: mariusjanssen at gmx.de (=?ISO-8859-1?Q?Marius_Jan=DFen?=) Date: Sun, 14 Jul 2013 23:34:08 +0200 Subject: [LAU] Fixing a MIDI issues In-Reply-To: References: Message-ID: <51E31950.3030403@gmx.de> Am 14.07.2013 21:37, schrieb Julien Claassen: > Hello everyone! > this is perghaps not straight on topic, but I couldn't find a > satisfactory answer anywhere else. > I'm having trouble with my JV-1080. Only every second note gets > played. I checked different input devices and different cables, direct > and indirect connections. > It seems to be a problem of the Roland itself. Internal sounds, > demos and the preview button, work fine. Any idea, what this might be? > It looks so systematic, as if it might be by design, but I don't know > any feature, which should allow for that. > Sorry to spam the list and thanks for following me so far. > Warm regards > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Dear Julien, dear list, Roland has introduced a feature in the JV-1080 called "stacking" so that up to 8 JVs can share incoming MIDI notes. Every JV-1080 can be told via the front panel how many modules share the incoming events and at which place in the chain it is ( for example the second JV out of five or the forth JV out of eight). Maybe You just turned that mode on accidentally That means with 3 present Roland JV-1080 the coming MIDI events will be played by following order JV 01 plays Voice 01 ; JV 02 plays Voice 02 ; JV 03 plays Voice 03 JV 01 plays Voice 04 ; JV 02 plays Voice 05 ; JV 03 plays Voice 06 JV 01 plays Voice 07 ; JV 02 plays Voice 08 ; JV 03 plays Voice 09 and so forth. You can change this mode back to OFF in Performance Mode, Patch Mode and GM MIDI Mode. This might be the same for the three modes, so here it is for the Performance Mode First You have to press SYSTEM key You can now change the value of 3 settings: 1) Control Channel, usually your MIDI Control Channel of your choice 2) Clock Source, MIDI Tempo clock, either intern or via MIDI 3) Stack, Settings are Off, 1 of 2 up to 8 of 8 I did not find a good computer readable JV1080 pdf manual in German or English online to advise you to the right page. Unfortunately I don't know if a MIDI command exists to change that Stack mode. This would be the easiest way for you to send it from your computer. I hope my answer was helpful. Greetings and God bless, Marius From althompson58 at gmail.com Mon Jul 15 02:55:32 2013 From: althompson58 at gmail.com (Al Thompson) Date: Sun, 14 Jul 2013 22:55:32 -0400 Subject: [LAU] Fixing a MIDI issues In-Reply-To: References: Message-ID: <51E364A4.5030009@gmail.com> On 07/14/2013 03:37 PM, Julien Claassen wrote: > Hello everyone! > this is perghaps not straight on topic, but I couldn't find a > satisfactory answer anywhere else. > I'm having trouble with my JV-1080. Only every second note gets > played. I checked different input devices and different cables, direct > and indirect connections. > It seems to be a problem of the Roland itself. Internal sounds, > demos and the preview button, work fine. Any idea, what this might be? > It looks so systematic, as if it might be by design, but I don't know > any feature, which should allow for that. > Sorry to spam the list and thanks for following me so far. > Warm regards > Julien IIRC, the JV1080 engine has, as one of it's many obscure features, one which will alternate every other note event on each LR output. If you are listening to only the left, for example, you would hear only every other note -- --- My bands, CD projects, music, news, and pictures: http://www.lateralforce.com My blog, with commentary on a variety of things, including audio, mixing, equipment, etc, is at: http://audioandmore.wordpress.com Staat hei?t das k?lteste aller kalten Ungeheuer. Kalt l?gt es auch; und diese L?ge kriecht aus seinem Munde: 'Ich, der Staat, bin das Volk.' - [Friedrich Nietzsche] From grekimj at acousticrefuge.com Mon Jul 15 10:29:46 2013 From: grekimj at acousticrefuge.com (Grekim Jennings) Date: Mon, 15 Jul 2013 06:29:46 -0400 Subject: [LAU] [a bit OT] Vocal microphone Message-ID: <51E3CF1A.7060405@acousticrefuge.com> >Hello! > I need to get a vocal microphone. I've had this project several times >before, but now I'm decided. There are a couple of alternatives. there is the >Shure SM5x and there is the SE-220, if my memory doesn't betray me. Any more >suggestions on the topic? I want to pay around 100-200 EUR maximum. This >microphone will be meant solely for singing. No instruments to mic. > Warm regards and sorry for the OT again > Julien Hi Julien, I think it's the Audix OM-6 that I have here. It is nicely built and has a lot of clarity, maybe too much presence in some cases. Also, check out the Heil PR-20 which I think is a little more neutral sounding and less expensive. And the AKG C700 is a very good value. As with most mics, with a good voice it can sound great. Grekim From gabbe.nord at gmail.com Mon Jul 15 12:13:04 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Mon, 15 Jul 2013 14:13:04 +0200 Subject: [LAU] My second album is done! In-Reply-To: <51E06B13.6030404@youmail.dk> References: <51E06B13.6030404@youmail.dk> Message-ID: Yes Atte, I was only talking about how I mostly use it, which is just like a tremolo. But yeah, it has many uses more than the simple tremolo-like duck-to-the-dance-kick. Arve: Thank you very much for the comment! Alexander: Thanks for your excellent sampling libraries. I'm sorry if I smashed some of the sounds in the drums ;). On Fri, Jul 12, 2013 at 10:46 PM, Atte Andr? Jensen wrote: > On 07/05/2013 12:20 PM, Gabbe Nord wrote: > >> Thinking about it, I think you can achieve more or less the same result >> by using a bus with a tremolo plugin on. Maybe worth checking out if >> sidechaining per say does not work! >> > > Depends if the compression is following a fixed pattern (like a > four-on-the-floor bass drumm) or not. > > Done more subtle, it can be a way of letting things like bass drum cut > through the mix. > > -- > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From csanchezgs at gmail.com Mon Jul 15 12:23:13 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 15 Jul 2013 14:23:13 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: <51E1B1B6.5010609@autostatic.com> References: <51E1B1B6.5010609@autostatic.com> Message-ID: Hi Jeremy, 2013/7/13 Jeremy Jongepier : > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/13/2013 09:50 PM, Carlos sanchiavedraz wrote: >> Hello, dear folks. >> >> I'm still (constantly) tweaking my raspberry here and there when I >> can. I have some configurations/sessions that run ok, Looper >> (jack+sooperlooper), Looper+Fx >> (jack+sooperlooper+rakarrack/guitarrix). But I'm still having some >> annoying problems that wanted to share with you: >> >> 1) Simple session script, Looper, runs quite well when launched >> (via ssh -X or from console) logged as PI user. When I run it as a >> boot/init script (simple launcher script placed in /etc/init.d and >> then installed with "update-rc.d launcherscript defaults 99"), >> almost in every first start there's crackles and noise and my loops >> sound distorted, even with clean guitar; if I restart then it goes >> well (2nd or 3d time). >> > > Hi Carlos, > > Try resetting the JACK buffersize with jack_bufsize after starting > your program. This does the trick for me on the RPi. It seems it did my trick as well, great! For the moment sound seems to be as clean as it can until the moment I debug other inner noises (it sounds like that of physical hard drives turning, but there's none in RPi (?)) when nothing is plugged in the audio interface. > >> 2) Next step forward I added Rakarrack in charge of guitar FX >> routed to the Looper. First, it happens almost same thing as >> mentioned about noise. Well, I guess It should have something to do >> with the signal path and things connected to the same power >> supplier, so I'll have to reserve some time to debug and switch off >> things until I get a cleaner signal and get rid of the hum and >> crackles. But the main issue is that there's problems with >> Rakarrack not been able to start because several problems with not >> enough Jack ports and Jack and RT. I didn't have any of these >> problems before. >> >> I got to solve the problem with ports changing -p16 to -p32 when >> launching jack+sooperlooper (-p24 is enough for just Rakarrack). >> >> Then I made a launcher script LSB compliant (service script placed >> in /etc/init.d and then installed with "update-rc.d servicescript >> defaults", with LSB flags and functions for "start" and "stop", as >> a proper newer Debian service, and configured to launch last in >> boot phase) which then call the Jack-config+Rakarrack+Sooperlooper >> session launcher script. I can see on screen in the init messages >> this error: >> >> "JACK is running in realtime mode, but you are not allowed to use >> realtime schedule..." >> >> But I know RT is already configured, so I guessed it was something >> to do with the boot/init stage configuration and environment. I >> tried using "su" with several different parameters to run the >> launcher script as "pi" user, as It runs quite well when logged as >> this user (calling just the script and even with "sudo >> /etc/init.d/script start"). So, maybe you can't get RT in init >> phase on RaspberryPI and you can only when you get to the login >> prompt and everything is loaded and ready. But I think there's a >> "Puredata on Raspberry PI" project that runs some scripts in the >> init stage; and it seems that in order to get a headless FX+looper >> station, running some configuration+apps-launcher at boot time is >> an obvious choice. >> >> I'm sure something escapes me after so much try-error-code cicles. >> Maybe you have some ideas. >> > > http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-March/091750.html I implemented just the first part that solution: uncomment the line "session required pam_limits.so" in "/etc/pam.d/su". Now my script and applicacionts run with "pi" user. Great again, Jeremy! > > Regards, > > Jeremy > >> Thanks as always. >> >> Now, there's just a minimum issue, but I can live with it. Rackarrack runs now as "pi" user and it should take its configuration from user's $HOME, that is what it should be "/home/pi", but it seems that it doesn't. The difference between running the launcher script when logged in as pi is that it takes the MIDI mapping configured in preferences, but when launched automatically on init phase it takes the default MIDI mapping. I run the launcher script from the service script with: su -l -c $STARTPATH/launcherscript.sh pi ...and "-l" should load the pi environment (I checked it when debugging). Maybe is something specific with Rakarrack. Thanks so much, Jeremy. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From rennabh at gmail.com Mon Jul 15 13:04:45 2013 From: rennabh at gmail.com (Renato) Date: Mon, 15 Jul 2013 15:04:45 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: References: Message-ID: <20130715150445.79aa30b2@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 13 Jul 2013 21:50:28 +0200 Carlos sanchiavedraz wrote: > Hello, dear folks. > > I'm still (constantly) tweaking my raspberry here and there when I > can. I have some configurations/sessions that run ok, Looper > (jack+sooperlooper), Looper+Fx > (jack+sooperlooper+rakarrack/guitarrix). Hi Carlos, just curious - how does the pi manage these loads? I remember some rakarrack effects being quite CPU intensive... Do you use it alongside guitarrix or only one at a time? I'm considering abandoning my home server project for the pi and getting back to one of my "ultimate guitar processor with linux" quests... cheers renato -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.20 (GNU/Linux) iQEcBAEBAgAGBQJR4/NxAAoJEBz6xFdttjrfUDsH/3bHO9aYrgRhkGYh91RKz2av IXTxPlO0aqIhxIZN/gtzyCUsymG5RaBTzO8fHgiS5PROd164tcL7wtFDAwD7QxT5 3Je5PeLgiJ+zM9HlvoG7yXNCoYfgn/Vge6qI6P+rLJy5F5z6tF38ZMh8s5ALIcVb 64nmcWWxnCB1ndsBWV0yfXNvWHOpNQA8oTQIvKWaMD6wxr19ZwSNYAXJcosCjqSd vw948qc1v41BJkV44qitVM7J+frplHX5sVNzfv21lyCNEfnPD/p/os88rgHxh98A DfFJdFUF0dakEQDkhaGFt3wR16r0gOmy8roI2oVzVAmSlu9A59nDla8+dABq+Yc= =HMFS -----END PGP SIGNATURE----- From jeremy at autostatic.com Mon Jul 15 13:47:40 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 15 Jul 2013 15:47:40 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: References: <51E1B1B6.5010609@autostatic.com> Message-ID: <51E3FD7C.5010701@autostatic.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/15/2013 02:23 PM, Carlos sanchiavedraz wrote: > Hi Jeremy, > [...] >> Hi Carlos, >> >> Try resetting the JACK buffersize with jack_bufsize after >> starting your program. This does the trick for me on the RPi. > > It seems it did my trick as well, great! For the moment sound seems > to be as clean as it can until the moment I debug other inner > noises (it sounds like that of physical hard drives turning, but > there's none in RPi (?)) when nothing is plugged in the audio > interface. > What kind of audio interface are you using? You might have to ground it properly (assunimg you're using an USB interface): http://sourceforge.net/apps/mediawiki/guitarix/index.php?title=Guitarix_Embedded_/_ARM_SoC#Fighting_the_Noise [...] > > Now, there's just a minimum issue, but I can live with it. > Rackarrack runs now as "pi" user and it should take its > configuration from user's $HOME, that is what it should be > "/home/pi", but it seems that it doesn't. The difference between > running the launcher script when logged in as pi is that it takes > the MIDI mapping configured in preferences, but when launched > automatically on init phase it takes the default MIDI mapping. > > I run the launcher script from the service script with: su -l -c > $STARTPATH/launcherscript.sh pi > > ...and "-l" should load the pi environment (I checked it when > debugging). Maybe is something specific with Rakarrack. > > Thanks so much, Jeremy. > I'm using 'su -l pi -c /some/command' myself but the order shouldn't matter I guess. Otherwise you could take a look at start-stop-daemon. Examples can be found in /etc/init.d/skeleton Jeremy -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) iQIcBAEBAgAGBQJR4/17AAoJEF63J34m6LiUMf0P/1w0LSRQLVKlSc9Kvacl7f3N ZowKUqcZcPgubZB85XB8ii5u5LUJKTVM7na/gL36l2sjEWafPZR5K/kT1wijYI6m bsTBZzPnEroSQld8Jwcyd3t7Gnp2Xo2amnszUjgadc4Cmu0PTlRzqKOd+Vmwjfnf PGpXbxix+0EyBK4TNiW+XtxwKOBiYisbK/gScMEoXHhcK31cdpC03OVYI89U8Yw9 rrgAC/E5RZtVmHptN2wyXDn0cL6cBCq+GuZtU474jvDfGX9gLeKa0jRErngifMZY S0ZLkrOymujTuB5rYw7qPCzYOGrOg9ItINIKTroGTJ2kC+TUNG8B1VFUdb0yPfF7 JFQlTTrqW9ciA1alAFRFJ9ZXFEoxwgEcIGjCmMNbpDtHFuX8ZRp1v76m3bNNgZn/ LaigRkXUjDilYKf62vQAsmZTRBx9SLZWJLJ360dGdkc/CSO3uAnTeMQMB3yqfSja pyb9e6K1dachCWHyF9jHIvADuXUC7fgSdfRhYllrFT5qMKu03NYXrT0lJjVQh+yD hmiIDF2RmHEbrsFfvsvyEFeiVeoDbJGPdOFzPSrn/8ruHQr83NZWjkUGv/8yjzf7 fES+4FIUuAREagZG1bX08e85l5q4mU0mlV74uBfhLY9D2p4XLU+ikyq3nAWPHlrJ jrn2YzjmbLa1EkaiGxVB =TOji -----END PGP SIGNATURE----- From nicola.di.marzo at vodafone.it Tue Jul 16 00:19:44 2013 From: nicola.di.marzo at vodafone.it (Nicola) Date: Tue, 16 Jul 2013 01:19:44 +0100 Subject: [LAU] raspberry-qmidinet-synth headless Message-ID: <51E491A0.1080309@vodafone.it> Hi all, now that i have some spare time to invest i was trying to control my RPI headless with my tablet (nexus 7 with android). I saw jeremy's video https://www.youtube.com/watch?v=Pycl9Oi1tv8&feature=c4-overview&list=UUby-h1uGg9pj7MyeAEIIf-g (great job man!) that was so inspiring and i've tried to follow his steps. I was able to make it work but i miss only 1 piece to the chain... Is there any possibility to run qmidinet headless without the gui? I can't make a script at boot working in this way 'cause i disabled X on my rpi. Any suggestion? Thank you Nicola From jeremy at autostatic.com Tue Jul 16 07:08:34 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 16 Jul 2013 09:08:34 +0200 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E491A0.1080309@vodafone.it> References: <51E491A0.1080309@vodafone.it> Message-ID: <51E4F172.3040002@autostatic.com> On 07/16/2013 02:19 AM, Nicola wrote: > Hi all, > now that i have some spare time to invest i was trying to control my RPI > headless with my tablet (nexus 7 with android). > I saw jeremy's video > https://www.youtube.com/watch?v=Pycl9Oi1tv8&feature=c4-overview&list=UUby-h1uGg9pj7MyeAEIIf-g > (great job man!) that was so inspiring and i've tried to follow his steps. > I was able to make it work but i miss only 1 piece to the chain... > > Is there any possibility to run qmidinet headless without the gui? > I can't make a script at boot working in this way 'cause i disabled X on > my rpi. > > Any suggestion? > Thank you > Nicola Hi Nicola, I'm using multimidicast myself, it's in my RPI repository. Example script here: https://raw.github.com/AutoStatic/scripts/rpi/amsynth-touchdaw Regards, Jeremy From nachoen79 at hotmail.com Tue Jul 16 10:59:05 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Tue, 16 Jul 2013 12:59:05 +0200 Subject: [LAU] mtc over network Message-ID: Hi everybody. My first mail here. I have a question about send mtc between two different cities over network. Something like the osx rtp-midi for linux. I've heard that you can do it with jack, but I?m not sure if it?s only for a lan network. Just want to know if possible before starting. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien at mail.upb.de Tue Jul 16 11:07:46 2013 From: julien at mail.upb.de (Julien Claassen) Date: Tue, 16 Jul 2013 13:07:46 +0200 (CEST) Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: Hey Nacho! There is aseqnet, this is an ALSA utility to connect MIDI over network. I'm not sure, how good this is for sweetly synchronised MIDI ports over the net for jamming together, but it works and it's small. As far as I'm aware, the JACK network driver is only intended for LANs nowadays. Tipp: If you want to keep your networks secure, you can use an ssh tunnel. Search google for something like "tunnel service over ssh", that's how I found it. I've tried it and it worked like a treat. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From nicola.di.marzo at vodafone.it Tue Jul 16 11:43:02 2013 From: nicola.di.marzo at vodafone.it (Nicola) Date: Tue, 16 Jul 2013 12:43:02 +0100 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E4F172.3040002@autostatic.com> References: <51E491A0.1080309@vodafone.it> <51E4F172.3040002@autostatic.com> Message-ID: <51E531C6.4010405@vodafone.it> On 16/07/13 08:08, Jeremy Jongepier wrote: > On 07/16/2013 02:19 AM, Nicola wrote: >> Hi all, >> now that i have some spare time to invest i was trying to control my RPI >> headless with my tablet (nexus 7 with android). >> I saw jeremy's video >> https://www.youtube.com/watch?v=Pycl9Oi1tv8&feature=c4-overview&list=UUby-h1uGg9pj7MyeAEIIf-g >> (great job man!) that was so inspiring and i've tried to follow his steps. >> I was able to make it work but i miss only 1 piece to the chain... >> >> Is there any possibility to run qmidinet headless without the gui? >> I can't make a script at boot working in this way 'cause i disabled X on >> my rpi. >> >> Any suggestion? >> Thank you >> Nicola > Hi Nicola, > > I'm using multimidicast myself, it's in my RPI repository. Example > script here: https://raw.github.com/AutoStatic/scripts/rpi/amsynth-touchdaw Thank you very much for sharing Jeremy! I was trying to use qmidinet that automatically associates multicast to alsa and jack midi ports so then you can connect quimidinet out to the soundcard output with jack_connect, pretty easy. The matter is that it needs an X session to run (at least AFAIK...) However great script! Nicola > > Regards, > > Jeremy > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From paul at linuxaudiosystems.com Tue Jul 16 11:50:07 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 16 Jul 2013 07:50:07 -0400 Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: On Tue, Jul 16, 2013 at 7:07 AM, Julien Claassen wrote: > Hey Nacho! > There is aseqnet, this is an ALSA utility to connect MIDI over network. > I'm not sure, how good this is for sweetly synchronised MIDI ports over the > net for jamming together, but it works and it's small. > As far as I'm aware, the JACK network driver is only intended for LANs > nowadays. > not correct. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Tue Jul 16 11:51:20 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 16 Jul 2013 13:51:20 +0200 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E531C6.4010405@vodafone.it> References: <51E491A0.1080309@vodafone.it> <51E4F172.3040002@autostatic.com> <51E531C6.4010405@vodafone.it> Message-ID: <51E533B8.6050905@autostatic.com> On 07/16/2013 01:43 PM, Nicola wrote: > I was trying to use qmidinet that automatically associates multicast to alsa and jack midi ports so then you can connect quimidinet out to the soundcard output with jack_connect, pretty easy. > The matter is that it needs an X session to run (at least AFAIK...) Hi Nicola, multimidicast can do this too. It's a CLI tool so it runs headless. I've also looked at qmidinet but it can't run without X (unless you use Xvfb). Jeremy From julien at mail.upb.de Tue Jul 16 11:54:15 2013 From: julien at mail.upb.de (Julien Claassen) Date: Tue, 16 Jul 2013 13:54:15 +0200 (CEST) Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: Hello Paul1 Did you change it again or was theLAN thing only in either of the twoJACK versions? I definitely did want to try and we had to set up a VPN tunnel to use the JACK net driver. Maybe I'm just antiquated in my knowledge. Best regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From nachoen79 at hotmail.com Tue Jul 16 13:29:21 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Tue, 16 Jul 2013 15:29:21 +0200 Subject: [LAU] mtc over network In-Reply-To: References: , , Message-ID: Please, what is not correct? aseqnet for mtc over wan networks or JACK only for LAN networks? Thanks. Date: Tue, 16 Jul 2013 07:50:07 -0400 Subject: Re: [LAU] mtc over network From: paul at linuxaudiosystems.com To: julien at mail.upb.de CC: nachoen79 at hotmail.com; linux-audio-user at lists.linuxaudio.org On Tue, Jul 16, 2013 at 7:07 AM, Julien Claassen wrote: Hey Nacho! There is aseqnet, this is an ALSA utility to connect MIDI over network. I'm not sure, how good this is for sweetly synchronised MIDI ports over the net for jamming together, but it works and it's small. As far as I'm aware, the JACK network driver is only intended for LANs nowadays. not correct. -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Tue Jul 16 13:36:14 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 16 Jul 2013 09:36:14 -0400 Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: personally i think that sending MTC via a WAN is ridiculous, even though it can be done. what are you trying to do? On Tue, Jul 16, 2013 at 9:29 AM, Nacho - wrote: > Please, what is not correct? aseqnet for mtc over wan networks or JACK > only for LAN networks? > Thanks. > > > > ------------------------------ > Date: Tue, 16 Jul 2013 07:50:07 -0400 > Subject: Re: [LAU] mtc over network > From: paul at linuxaudiosystems.com > To: julien at mail.upb.de > CC: nachoen79 at hotmail.com; linux-audio-user at lists.linuxaudio.org > > > > > > On Tue, Jul 16, 2013 at 7:07 AM, Julien Claassen wrote: > > Hey Nacho! > There is aseqnet, this is an ALSA utility to connect MIDI over network. > I'm not sure, how good this is for sweetly synchronised MIDI ports over the > net for jamming together, but it works and it's small. > As far as I'm aware, the JACK network driver is only intended for LANs > nowadays. > > > not correct. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From csanchezgs at gmail.com Tue Jul 16 13:37:08 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 16 Jul 2013 15:37:08 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: <20130715150445.79aa30b2@gmail.com> References: <20130715150445.79aa30b2@gmail.com> Message-ID: Hi, Renato, 2013/7/15 Renato : > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Sat, 13 Jul 2013 21:50:28 +0200 > Carlos sanchiavedraz wrote: > >> Hello, dear folks. >> >> I'm still (constantly) tweaking my raspberry here and there when I >> can. I have some configurations/sessions that run ok, Looper >> (jack+sooperlooper), Looper+Fx >> (jack+sooperlooper+rakarrack/guitarrix). > > Hi Carlos, just curious - how does the pi manage these loads? I > remember some rakarrack effects being quite CPU intensive... Do you use > it alongside guitarrix or only one at a time? > > I'm considering abandoning my home server project for the pi and > getting back to one of my "ultimate guitar processor with linux" > quests... > > cheers > renato > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2.0.20 (GNU/Linux) > > iQEcBAEBAgAGBQJR4/NxAAoJEBz6xFdttjrfUDsH/3bHO9aYrgRhkGYh91RKz2av > IXTxPlO0aqIhxIZN/gtzyCUsymG5RaBTzO8fHgiS5PROd164tcL7wtFDAwD7QxT5 > 3Je5PeLgiJ+zM9HlvoG7yXNCoYfgn/Vge6qI6P+rLJy5F5z6tF38ZMh8s5ALIcVb > 64nmcWWxnCB1ndsBWV0yfXNvWHOpNQA8oTQIvKWaMD6wxr19ZwSNYAXJcosCjqSd > vw948qc1v41BJkV44qitVM7J+frplHX5sVNzfv21lyCNEfnPD/p/os88rgHxh98A > DfFJdFUF0dakEQDkhaGFt3wR16r0gOmy8roI2oVzVAmSlu9A59nDla8+dABq+Yc= > =HMFS > -----END PGP SIGNATURE----- By now I use Rakarrack standalone (FX), and also connected to Sooperlooper (FX+looper). I made some time ago, for a different not-powerful device, preset which have in its signal path just FX that are let's say "simple": Compressor, Echo, Chorus, Flanger... No Convolver or the other stereo versions of those fx. It would be great to have some kind of colored (red-yellow-green) list of FX by CPU consumption, but for now I just guess which plugins could RPI handle. On this preset I chose also Harmonizer, but it turns out that for RPi and jack with buffer up to 256 the voice generated has noises, so is a little unusable, and for higher buffer values you start to notice the delay. I have to try some more to see if I can tweak something (right now it is selected the minimum quality (4) for Harmonizer in Rakarrack preferences). -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From nachoen79 at hotmail.com Tue Jul 16 13:43:15 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Tue, 16 Jul 2013 15:43:15 +0200 Subject: [LAU] mtc over network In-Reply-To: References: , , , , Message-ID: I'm trying to sync a video player on my pc, using the mtc sent from another linux or mac computer. That mtc-computer is on another city. Is that possible with linux? Date: Tue, 16 Jul 2013 09:36:14 -0400 Subject: Re: [LAU] mtc over network From: paul at linuxaudiosystems.com To: nachoen79 at hotmail.com CC: julien at mail.upb.de; linux-audio-user at lists.linuxaudio.org personally i think that sending MTC via a WAN is ridiculous, even though it can be done. what are you trying to do? On Tue, Jul 16, 2013 at 9:29 AM, Nacho - wrote: Please, what is not correct? aseqnet for mtc over wan networks or JACK only for LAN networks? Thanks. Date: Tue, 16 Jul 2013 07:50:07 -0400 Subject: Re: [LAU] mtc over network From: paul at linuxaudiosystems.com To: julien at mail.upb.de CC: nachoen79 at hotmail.com; linux-audio-user at lists.linuxaudio.org On Tue, Jul 16, 2013 at 7:07 AM, Julien Claassen wrote: Hey Nacho! There is aseqnet, this is an ALSA utility to connect MIDI over network. I'm not sure, how good this is for sweetly synchronised MIDI ports over the net for jamming together, but it works and it's small. As far as I'm aware, the JACK network driver is only intended for LANs nowadays. not correct. -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Tue Jul 16 13:39:53 2013 From: len at ovenwerks.net (Len Ovens) Date: Tue, 16 Jul 2013 06:39:53 -0700 Subject: [LAU] mtc over network Message-ID: On Tue, July 16, 2013 4:54 am, Julien Claassen wrote: > Hello Paul1 > Did you change it again or was theLAN thing only in either of the > twoJACK > versions? I definitely did want to try and we had to set up a VPN tunnel > to > use the JACK net driver. Maybe I'm just antiquated in my knowledge. At least one end has to have a "visible" IP. Physical distance of the two ends affects the latency, as do the switches/routers in the path. The lowest bandwidth link in the path determines the maximum bandwidth of information a netjack link deal with. Obviously MTC is low bandwidth, but MTC without some other information (either a midi or audio stream) is of limited value :) -- Len Ovens www.OvenWerks.net From csanchezgs at gmail.com Tue Jul 16 13:47:02 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 16 Jul 2013 15:47:02 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: <51E3FD7C.5010701@autostatic.com> References: <51E1B1B6.5010609@autostatic.com> <51E3FD7C.5010701@autostatic.com> Message-ID: Hi Jeremy, 2013/7/15 Jeremy Jongepier : > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/15/2013 02:23 PM, Carlos sanchiavedraz wrote: >> Hi Jeremy, >> > > [...] > >>> Hi Carlos, >>> >>> Try resetting the JACK buffersize with jack_bufsize after >>> starting your program. This does the trick for me on the RPi. >> >> It seems it did my trick as well, great! For the moment sound seems >> to be as clean as it can until the moment I debug other inner >> noises (it sounds like that of physical hard drives turning, but >> there's none in RPi (?)) when nothing is plugged in the audio >> interface. >> > > What kind of audio interface are you using? You might have to ground > it properly (assunimg you're using an USB interface): > http://sourceforge.net/apps/mediawiki/guitarix/index.php?title=Guitarix_Embedded_/_ARM_SoC#Fighting_the_Noise > What I supposed is exactly what it points in that article: "noise on the USB power line will feed into the preamplifier of the audio interface", or maybe not, but it is probably. But cutting cables and soldering I'm afraid it's not an option for now; don't want to change my audio interface into some Frankenterface and break it. I'll see if I can clean a little bit more without going extremes. > [...] > >> >> Now, there's just a minimum issue, but I can live with it. >> Rackarrack runs now as "pi" user and it should take its >> configuration from user's $HOME, that is what it should be >> "/home/pi", but it seems that it doesn't. The difference between >> running the launcher script when logged in as pi is that it takes >> the MIDI mapping configured in preferences, but when launched >> automatically on init phase it takes the default MIDI mapping. >> >> I run the launcher script from the service script with: su -l -c >> $STARTPATH/launcherscript.sh pi >> >> ...and "-l" should load the pi environment (I checked it when >> debugging). Maybe is something specific with Rakarrack. >> >> Thanks so much, Jeremy. >> > > I'm using 'su -l pi -c /some/command' myself but the order shouldn't > matter I guess. Otherwise you could take a look at start-stop-daemon. > Examples can be found in /etc/init.d/skeleton > > Jeremy My service script is based on Debian norms: http://wiki.debian.org/LSBInitScripts Basically: * First the config for Run-time dependencies in the format indicated * Shell methods for start, stop... On my method start is that command I mentioned: su -l -c >> $STARTPATH/launcherscript.sh pi I've seen that other service run whichever command but preceded by start-stop-daemon -v, but I'm not sure it makes a difference for this issue. And since this is not a proper "service" but a launcher at the very beginning, I didn't use start-stop-daemon. Thanks again. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Tue Jul 16 13:50:29 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 16 Jul 2013 15:50:29 +0200 Subject: [LAU] Raspberry: problems with shell script launcher when booting In-Reply-To: References: <20130715150445.79aa30b2@gmail.com> Message-ID: 2013/7/16 Carlos sanchiavedraz : > Hi, Renato, > > 2013/7/15 Renato : >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> On Sat, 13 Jul 2013 21:50:28 +0200 >> Carlos sanchiavedraz wrote: >> >>> Hello, dear folks. >>> >>> I'm still (constantly) tweaking my raspberry here and there when I >>> can. I have some configurations/sessions that run ok, Looper >>> (jack+sooperlooper), Looper+Fx >>> (jack+sooperlooper+rakarrack/guitarrix). >> >> Hi Carlos, just curious - how does the pi manage these loads? I >> remember some rakarrack effects being quite CPU intensive... Do you use >> it alongside guitarrix or only one at a time? >> >> I'm considering abandoning my home server project for the pi and >> getting back to one of my "ultimate guitar processor with linux" >> quests... >> >> cheers >> renato >> -----BEGIN PGP SIGNATURE----- >> Version: GnuPG v2.0.20 (GNU/Linux) >> >> iQEcBAEBAgAGBQJR4/NxAAoJEBz6xFdttjrfUDsH/3bHO9aYrgRhkGYh91RKz2av >> IXTxPlO0aqIhxIZN/gtzyCUsymG5RaBTzO8fHgiS5PROd164tcL7wtFDAwD7QxT5 >> 3Je5PeLgiJ+zM9HlvoG7yXNCoYfgn/Vge6qI6P+rLJy5F5z6tF38ZMh8s5ALIcVb >> 64nmcWWxnCB1ndsBWV0yfXNvWHOpNQA8oTQIvKWaMD6wxr19ZwSNYAXJcosCjqSd >> vw948qc1v41BJkV44qitVM7J+frplHX5sVNzfv21lyCNEfnPD/p/os88rgHxh98A >> DfFJdFUF0dakEQDkhaGFt3wR16r0gOmy8roI2oVzVAmSlu9A59nDla8+dABq+Yc= >> =HMFS >> -----END PGP SIGNATURE----- > > By now I use Rakarrack standalone (FX), and also connected to > Sooperlooper (FX+looper). > > I made some time ago, for a different not-powerful device, preset > which have in its signal path just FX that are let's say "simple": > Compressor, Echo, Chorus, Flanger... No Convolver or the other stereo > versions of those fx. It would be great to have some kind of colored > (red-yellow-green) list of FX by CPU consumption, but for now I just > guess which plugins could RPI handle. > > On this preset I chose also Harmonizer, but it turns out that for RPi > and jack with buffer up to 256 the voice generated has noises, so is a > little unusable, and for higher buffer values you start to notice the > delay. I have to try some more to see if I can tweak something (right > now it is selected the minimum quality (4) for Harmonizer in Rakarrack > preferences). > > > > -- > Carlos sanchiavedraz > * Musix GNU+Linux > http://www.musix.es Arpie works well also. I would be trying FX to see which of them RPi can/cannot handle. Would love to know you're (Renato and all) experiences about. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From rncbc at rncbc.org Tue Jul 16 14:49:56 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Tue, 16 Jul 2013 15:49:56 +0100 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready Message-ID: <51E55D94.2060705@rncbc.org> hi july ;) yet another batch of the Vee One Suite of old-school software instruments is out: synthv1 [1] polyphonic synthesizer, samplv1 [2] polyphonic sampler and drumkv1 [3] drum-kit sampler, this time presenting their brand new icons while dropping the lamest old-schooler's out, in a master lesson taught by Jarle Richard Akselsen, thanks. all still available in dual form: - a pure stand-alone JACK client with JACK-session, NSM (Non Session management) and both JACK MIDI and ALSA MIDI input support; - a LV2 instrument plugin. all free and open-source Linux Audio software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. here goes the new triumvirate bunch again: [1] synthv1 - an old-school polyphonic synthesizer synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic synthesizer with stereo fx. LV2 URI: http://synthv1.sourceforge.net/lv2 website: http://synthv1.sourceforge.net downloads: http://sourceforge.net/projects/synthv1/files - source tarball: http://download.sourceforge.net/synthv1/synthv1-0.3.4.tar.gz - source package: http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.src.rpm - binary packages: http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.i586.rpm http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.x86_84.rpm [2] samplv1 - an old-school polyphonic sampler samplv1 is an(other) old-school all-digital polyphonic sampler synthesizer with stereo fx. LV2 URI: http://samplv1.sourceforge.net/lv2 website: http://samplv1.sourceforge.net downloads: http://sourceforge.net/projects/samplv1/files - source tarball: http://download.sourceforge.net/samplv1/samplv1-0.3.4.tar.gz - source package: http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.src.rpm - binary packages: http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.i586.rpm http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.x86_84.rpm [3] drumkv1 - an old-school drum-kit sampler drumkv1 is (yet) an(other) old-school all-digital drum-kit sampler synthesizer with stereo fx. LV2 URI: http://drumkv1.sourceforge.net/lv2 website: http://drumkv1.sourceforge.net downloads: http://sourceforge.net/projects/drumkv1/files - source tarball: http://download.sourceforge.net/drumkv1/drumkv1-0.3.4.tar.gz - source package: http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.src.rpm - binary packages: http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.i586.rpm http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.x86_84.rpm see also: http://www.rncbc.org/drupal/node/679 enjoy && summer's ready! -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From fede2001 at hotmail.com Tue Jul 16 15:10:52 2013 From: fede2001 at hotmail.com (Federico Lopez) Date: Tue, 16 Jul 2013 10:10:52 -0500 Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: On 07/16/2013 08:36 AM, Paul Davis wrote: > personally i think that sending MTC via a WAN is ridiculous, even > though it can be done. Could it be an alternate solution to send MTC from stage to Front of house in concerts (typically 30mts) to sync automatization? cheers, federico PD: There is a dedicated hardware solution http://www.midijet.com/ From paul at linuxaudiosystems.com Tue Jul 16 15:20:19 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 16 Jul 2013 11:20:19 -0400 Subject: [LAU] mtc over network In-Reply-To: References: Message-ID: that isn't a WAN. On Tue, Jul 16, 2013 at 11:10 AM, Federico Lopez wrote: > > On 07/16/2013 08:36 AM, Paul Davis wrote: > >> personally i think that sending MTC via a WAN is ridiculous, even though >> it can be done. >> > > Could it be an alternate solution to send MTC from stage to Front of house > in concerts (typically 30mts) to sync automatization? > > > cheers, > > federico > > > PD: There is a dedicated hardware solution http://www.midijet.com/ > > > > ______________________________**_________________ > Linux-audio-user mailing list > Linux-audio-user at lists.**linuxaudio.org > http://lists.linuxaudio.org/**listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fero.kiraly at gmail.com Tue Jul 16 15:29:37 2013 From: fero.kiraly at gmail.com (Fero Kiraly) Date: Tue, 16 Jul 2013 17:29:37 +0200 Subject: [LAU] 6 x MIDI in Message-ID: Hi, maybe not a linux audio question but I try: I need 6 MIDI IN ports ( six separate instruments) connect to my system (presonus 1818vsl) is there some (cheap) solution ? thanx fero -------------- next part -------------- An HTML attachment was scrubbed... URL: From nachoen79 at hotmail.com Tue Jul 16 15:40:04 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Tue, 16 Jul 2013 17:40:04 +0200 Subject: [LAU] mtc over network In-Reply-To: References: , , , , , , Message-ID: I understand then that there is nothing similar (not in development) to rtpmidi for linux? What about jack? When I say something like rtpmidi I'm talking about midi over a wan network. Date: Tue, 16 Jul 2013 11:20:19 -0400 From: paul at linuxaudiosystems.com To: fede2001 at hotmail.com CC: linux-audio-user at lists.linuxaudio.org Subject: Re: [LAU] mtc over network that isn't a WAN. On Tue, Jul 16, 2013 at 11:10 AM, Federico Lopez wrote: On 07/16/2013 08:36 AM, Paul Davis wrote: personally i think that sending MTC via a WAN is ridiculous, even though it can be done. Could it be an alternate solution to send MTC from stage to Front of house in concerts (typically 30mts) to sync automatization? cheers, federico PD: There is a dedicated hardware solution http://www.midijet.com/ _______________________________________________ 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 julien at mail.upb.de Tue Jul 16 16:16:45 2013 From: julien at mail.upb.de (Julien Claassen) Date: Tue, 16 Jul 2013 18:16:45 +0200 (CEST) Subject: [LAU] 6 x MIDI in In-Reply-To: References: Message-ID: Hello Fero! My solution would be to buy as many MIDI -> USB cables and be done with it. If your onboard soundcard has one pair of MIDI jacks, then you'd need five more. I paid about 10 EUR for one cable. Works out of the box and I have them all as separate devices. Very comfortable. If you're in luck a few of your instruments might have a direct MIDI connector, those usually work. My Korg and the MiniBrute work that way and there was no hassle at all. You can of course try a midisport. I don't know how many ports you can get with one. I know the 2x2 and the 4x4. Not sure about the prices, but I assume not too much. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From ilitzroth at gmail.com Tue Jul 16 16:30:46 2013 From: ilitzroth at gmail.com (immanuel litzroth) Date: Tue, 16 Jul 2013 18:30:46 +0200 Subject: [LAU] 6 x MIDI in In-Reply-To: References: Message-ID: I use emagic 8x unitor's or amt which can be got cheap second hand if you''re on the lookout. Don't know where you're from, I have 2 I'm not using right now. Immanuel On Tue, Jul 16, 2013 at 5:29 PM, Fero Kiraly wrote: > Hi, > > maybe not a linux audio question but I try: > > I need 6 MIDI IN ports ( six separate instruments) connect to my system > (presonus 1818vsl) > > is there some (cheap) solution ? > thanx > > fero > > _______________________________________________ > 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 ralf.mardorf at alice-dsl.net Tue Jul 16 18:11:19 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Tue, 16 Jul 2013 20:11:19 +0200 Subject: [LAU] 6 x MIDI in In-Reply-To: References: Message-ID: On Tue, 16 Jul 2013 18:16:45 +0200, Julien Claassen wrote: > My solution would be to buy as many MIDI -> USB cables and be done with > it. USB does cause MIDI jitter, it might be ok for most people, but I guess it should be mentioned. For me it isn't ok, I don't use my USB device, but PCI cards instead. However, do you need 8 inputs at the same time? If not, I would use a switch box. If you need some inputs only to dump sounds etc., than MIDI jitter isn't an issue and USB should be ok for everybody. From rncbc at rncbc.org Wed Jul 17 14:24:09 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Wed, 17 Jul 2013 15:24:09 +0100 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E533B8.6050905@autostatic.com> References: <51E491A0.1080309@vodafone.it> <51E4F172.3040002@autostatic.com> <51E531C6.4010405@vodafone.it> <51E533B8.6050905@autostatic.com> Message-ID: <51E6A909.70504@rncbc.org> On 07/16/2013 12:51 PM, Jeremy Jongepier wrote: > On 07/16/2013 01:43 PM, Nicola wrote: >> I was trying to use qmidinet that automatically associates multicast to alsa and jack midi ports so then you can connect quimidinet out to the soundcard output with jack_connect, pretty easy. >> The matter is that it needs an X session to run (at least AFAIK...) > > Hi Nicola, > > multimidicast can do this too. It's a CLI tool so it runs headless. I've > also looked at qmidinet but it can't run without X (unless you use Xvfb). > sorry to come this late. yes, multimidicast can do the trick of course. however it doesn't deal with sysex at all and it's alsa-midi (aka. alsa-seq) only. if enough demand, i'm sure it won't be that hard to have a gui-less version of qmidinet might be just one nice and really easy summer/week-end/one-night-stand coding project... ;) any takers? -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From nicola.di.marzo at vodafone.it Wed Jul 17 23:01:27 2013 From: nicola.di.marzo at vodafone.it (Nicola) Date: Thu, 18 Jul 2013 00:01:27 +0100 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E6A909.70504@rncbc.org> References: <51E491A0.1080309@vodafone.it> <51E4F172.3040002@autostatic.com> <51E531C6.4010405@vodafone.it> <51E533B8.6050905@autostatic.com> <51E6A909.70504@rncbc.org> Message-ID: <51E72247.6030903@vodafone.it> On 17/07/13 15:24, Rui Nuno Capela wrote: > On 07/16/2013 12:51 PM, Jeremy Jongepier wrote: >> On 07/16/2013 01:43 PM, Nicola wrote: >>> I was trying to use qmidinet that automatically associates multicast >>> to alsa and jack midi ports so then you can connect quimidinet out >>> to the soundcard output with jack_connect, pretty easy. >>> The matter is that it needs an X session to run (at least AFAIK...) >> >> Hi Nicola, >> >> multimidicast can do this too. It's a CLI tool so it runs headless. I've >> also looked at qmidinet but it can't run without X (unless you use >> Xvfb). >> > > sorry to come this late. > > yes, multimidicast can do the trick of course. however it doesn't deal > with sysex at all and it's alsa-midi (aka. alsa-seq) only. > > if enough demand, i'm sure it won't be that hard to have a gui-less > version of qmidinet > > might be just one nice and really easy summer/week-end/one-night-stand > coding project... ;) > > any takers? Hello Rui, Thanks for your interest, that would be a great feature. I'm not a developer myself and i don't know if i can help so much. I usually make some bash script and that's all...but if there's anything else i can do to help(donation:-) ), i'm here. Regards Nicola From jeremy at autostatic.com Thu Jul 18 07:07:48 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 18 Jul 2013 09:07:48 +0200 Subject: [LAU] raspberry-qmidinet-synth headless In-Reply-To: <51E72247.6030903@vodafone.it> References: <51E491A0.1080309@vodafone.it> <51E4F172.3040002@autostatic.com> <51E531C6.4010405@vodafone.it> <51E533B8.6050905@autostatic.com> <51E6A909.70504@rncbc.org> <51E72247.6030903@vodafone.it> Message-ID: <51E79444.4060308@autostatic.com> On 07/18/2013 01:01 AM, Nicola wrote: > On 17/07/13 15:24, Rui Nuno Capela wrote: >> On 07/16/2013 12:51 PM, Jeremy Jongepier wrote: >>> On 07/16/2013 01:43 PM, Nicola wrote: >>>> I was trying to use qmidinet that automatically associates multicast >>>> to alsa and jack midi ports so then you can connect quimidinet out >>>> to the soundcard output with jack_connect, pretty easy. >>>> The matter is that it needs an X session to run (at least AFAIK...) >>> >>> Hi Nicola, >>> >>> multimidicast can do this too. It's a CLI tool so it runs headless. I've >>> also looked at qmidinet but it can't run without X (unless you use >>> Xvfb). >>> >> >> sorry to come this late. >> >> yes, multimidicast can do the trick of course. however it doesn't deal >> with sysex at all and it's alsa-midi (aka. alsa-seq) only. >> >> if enough demand, i'm sure it won't be that hard to have a gui-less >> version of qmidinet >> >> might be just one nice and really easy summer/week-end/one-night-stand >> coding project... ;) >> >> any takers? > Hello Rui, > > Thanks for your interest, that would be a great feature. > I'm not a developer myself and i don't know if i can help so much. > I usually make some bash script and that's all...but if there's anything > else i can do to help(donation:-) ), i'm here. > > Regards > Nicola I'd be interested too as multimidicast only does ALSA MIDI and no JACK MIDI. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: 0x26E8B894.asc Type: application/pgp-keys Size: 3129 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From neilcsmith.net at googlemail.com Thu Jul 18 11:39:08 2013 From: neilcsmith.net at googlemail.com (Neil C Smith) Date: Thu, 18 Jul 2013 12:39:08 +0100 Subject: [LAU] Announcing praxislive.org Message-ID: Hi All, I finally got around to giving Praxis LIVE its own home on the web this week. There's a lot more to come in the near future in terms of help and resources, but this should hopefully give a better overview of what the project's all about. www.praxislive.org Praxis LIVE is an open-source, graphical environment for rapid development of intermedia performance tools, projections and interactive spaces. This replaces and enhances the somewhat outdated information on the Google Code site. Downloads and code are still hosted there, though I'll be looking to host downloads elsewhere in the near future now Google has deprecated this. Thoughts and / or contributions to the further development of the project and its online resources very much appreciated. Thanks to whoever mentioned Praxis LIVE on the open-source musician podcast last month - your slight confusion over what it was for was one of a number of factors that prompted the development of this website! :-) Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source intermedia development - www.praxislive.org Digital Prisoners - interactive spaces and projections - www.digitalprisoners.co.uk OpenEye - the web, managed - www.openeye.info From neilcsmith.net at googlemail.com Thu Jul 18 12:03:10 2013 From: neilcsmith.net at googlemail.com (Neil C Smith) Date: Thu, 18 Jul 2013 13:03:10 +0100 Subject: [LAU] Music(?) made with Linux: Meeting Point Message-ID: Well, as well as improving the web presence for Praxis LIVE this week (see t'other thread), I've also recently built a website for Digital Prisoners. All our projects are created with Praxis LIVE on Linux. However, I thought I'd share this page and video of Meeting Point from earlier this year - an interactive projection that allow 2 people at opposite ends of a public square to dance "together". Audio and visuals are being processed live - a mix of generative and motion triggered effects. Input loops / samples are a mix of stuff. Mostly created and/or edited on Linux, a couple of things off FreeSound, and at least one loop was created a long time ago on an Acorn A5000 (now that's a computer! :-) ) http://www.digitalprisoners.co.uk/projects/meeting-point http://youtu.be/lXLmhCwWSNw Enjoy! Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source intermedia development - www.praxislive.org Digital Prisoners - interactive spaces and projections - www.digitalprisoners.co.uk OpenEye - the web, managed - www.openeye.info From rncbc at rncbc.org Thu Jul 18 15:32:44 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Thu, 18 Jul 2013 16:32:44 +0100 Subject: [LAU] [ANN] Qtractor 0.5.10 - The Kilo Papa is out Message-ID: <51E80A9C.8090402@rncbc.org> Nothing much to say... but anything goes, as long it goes before the proverbial midsummer meltdown... whatever:) Qtractor 0.5.10 (kilo papa) is out! Release highlights: * Edit/Insert, Remove range options (NEW) * LV2 Dyn-manifest support (NEW) * Time, frames, BBT display option in-place menu (NEW) * Audio export track automation (FIX) * Clip/event selection clear/reset (FIX) Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor Downloads: http://sourceforge.net/projects/qtractor/files - source tarball: http://downloads.sourceforge.net/qtractor/qtractor-0.5.10.tar.gz - source package (openSUSE 12.3): http://downloads.sourceforge.net/qtractor/qtractor-0.5.10-7.rncbc.suse123.src.rpm - binary packages (openSUSE 12.3): http://downloads.sourceforge.net/qtractor/qtractor-0.5.10-7.rncbc.suse123.i586.rpm http://downloads.sourceforge.net/qtractor/qtractor-0.5.10-7.rncbc.suse123.x86_64.rpm - quick start guide & user manual: http://downloads.sourceforge.net/qtractor/qtractor-0.5.x-user-manual.pdf Weblog (upstream support): http://www.rncbc.org License: Qtractor is free, open-source software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. Change-log: - Default drum-key note names are now properly showing on MIDI tracks that are assigned to known drum/percussive instrument patches (eg. SoundFont 2 (.sf2) bank 128). - Time display format (frames, clock-time or BBT) may now be changed from the context-menu on any time entry spin-box. - LV2 plugin support is now tightly tied to liblilv; the same tie applies to LV2 plugin UI support and libsuil and vice-versa. - Mixer buses racks (ie. left/input and right/output panes) are now both kept fixed-width when whole mixer window is resized. - Unconditional LV2 Dyn(amic)-manifest support has been added. - Main track-view Edit/Insert,Remove/Range dialog is now being introduced with optional applicability to Clips, Loop, Punch in/out, Automation, Tempo-map and/or Markers. - New range removal editing tool, split/moving clips backward at the specified edit-head/tail interval (Edit/Remove/Range, Track Range)--by Tuomas Airaksinen, thanks. - Andy Fitzsimon's original icon from opencliparts.org makes it through as the default standard scalable format (SVG). - Automation's back in effect on Track/Export Tracks.../Audio. - Reversed shift/ctrl keyboard modifier roles on middle-button clicking over the main track and MIDI clip editor views (aka. piano-roll) in regression to original old semantics. - Color selection actions now have a brand new palette icon. - Make sure main track-view and MIDI clip editor selection is only cleared on specific discrete commands. - Try keeping the original session file in most recent files menu list, despite current version auto-incremental backup mode is in effect. - Fixed non-zero clip offsets upon tempo/time-scale changes. - Some sympathy to extreme dark color (read black) schemes is now indulged on empty backgrounds. See also: http://www.rncbc.org/drupal/node/680 Enjoy && have (lots of) fun. -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From andreas.degert at googlemail.com Thu Jul 18 21:29:13 2013 From: andreas.degert at googlemail.com (Andreas Degert) Date: Thu, 18 Jul 2013 23:29:13 +0200 Subject: [LAU] Guitarix 0.28.0 released Message-ID: The Guitarix developers proudly present Guitarix release 0.28.0 "magic chainsaw trick" *) Yes, we took a chainsaw, cut through the chest of Guitarix, and... it's still alive. But now it's also 2 pieces. You can start it (headless) on an embedded ARM system. You can start it on your laptop and connect the user interface the the headless instance. You can even tap on your smartphone and control Guitarix with the web browser (or just use the tuner). For the uninitiated, Guitarix is a tube amplifier simulation for jack, with effect modules and an additional stereo effect chain. Please refer to our project page for more information: http://guitarix.sourceforge.net/ Download Site: http://sourceforge.net/projects/guitarix/ Forum: http://guitarix.sourceforge.net/forum/ Embedded Guitarix Prototype: http://sourceforge.net/apps/mediawiki/guitarix/index.php?title=Guitarix_Embedded_/_ARM_SoC Please consider visiting our forum or leaving a message on guitarix-developer at lists.sourceforge.net if you plan to work with embedded Guitarix. List of changes: * new french translation contributed by Bajo * new MultiBandCompressor contributed by kokoko3k * new include BestPlugins IR-Packs I-III by David Fau Casquel * new DigitalDelay Effect module * the Guitarix LV2 plugins are now in the default build (but you can configure the build with --no-lv2) * Optimizations for embedded systems / ARM processors with NEON * --faust-vectorize-float, --convolver-ffmpeg * Guitarix headless mode for embedded systems (no X11 / user Interface) * network socket based service for controlling / connecting a user interface * service announcement via Avahi, if available * command line options --rpchost, --rpcport, --nogui * the service can be started in addition to the "local" user interface * Guitarix can now be used as a user interface connecting to a remote instance * no configuration needed if Avahi is available * command line options --onlygui, --rpchost, --rpcport * multiple user interface clients can be started (the displays keep synchronized) * Browser-based user interface (javascript code) * cross platform * usable on small devices like smartphones, and in conjunction with an embedded device * some features still missing (no MIDI controller connections, no presets for single effect units, no polished appearance) * small additional server needed (included), to translate between the socket based interface and WebSockets, which is what (modern) browsers understand * only recent browsers supported, testers needed * released in a separate tarfile *) http://www.youtube.com/watch?v=GPTHdwqG9rw -------------- next part -------------- An HTML attachment was scrubbed... URL: From harryhaaren at gmail.com Fri Jul 19 11:17:00 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 19 Jul 2013 12:17:00 +0100 Subject: [LAU] OpenAV Fabla Message-ID: Hi, Its my pleasure to announce the release of Fabla! After 8 days we have reached the target donation amount, many thanks to all those who contributed! Available here: http://openavproductions.com/fabla Cheers! -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From perodog at gmx.net Fri Jul 19 16:19:07 2013 From: perodog at gmx.net (Dragan Noveski) Date: Fri, 19 Jul 2013 18:19:07 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: Message-ID: <51E966FB.8080403@gmx.net> On 19.07.2013 13:17, Harry van Haaren wrote: > Hi, > > Its my pleasure to announce the release of Fabla! > > After 8 days we have reached the target donation amount, many thanks > to all those who contributed! > > Available here: http://openavproductions.com/fabla > > Cheers! -Harry > hallo harry, i tried to compile fabla, but there is an error: nowhiskey at murija7:~/Desktop/src/openAV-Fabla$ make mkdir -p fabla.lv2/ g++ gui/ui_helpers.cxx gui/fabla.cxx gui/fabla_ui.c -pthread -D_LARGEFILE64_SOURCE -D_FILE_OFFSET_BITS=64 -D_GNU_SOURCE -I/usr/include/cairo -I/usr/include/sigc++-2.0 -I/usr/lib/i386-linux-gnu/sigc++-2.0/include -I/usr/include/glib-2.0 -I/usr/lib/i386-linux-gnu/glib-2.0/include -I/usr/include/pixman-1 -I/usr/include/freetype2 -I/usr/include/libpng12 -I/usr/include/cairomm-1.0 -I/usr/lib/cairomm-1.0/include -I/usr/include/ntk -g -Wall -lsndfile -lcairomm-1.0 -lsigc-2.0 -lntk_images -lntk -lcairo -fPIC -shared -Wl,-z,nodelete -o fabla.lv2/fabla_ui.so gui/fabla_ui.c:473:14: error: ?LV2UI_Idle_Interface? does not name a type gui/fabla_ui.c: In function ?const void* extension_data(const char*)?: gui/fabla_ui.c:479:20: error: ?LV2_UI__idleInterface? was not declared in this scope gui/fabla_ui.c:481:13: error: ?idle_iface? was not declared in this scope gui/fabla_ui.c: At global scope: gui/fabla_ui.c:465:12: warning: ?int idle(LV2UI_Handle)? defined but not used [-Wunused-function] make: *** [fabla.lv2/fabla_ui.so] Fehler 1 nowhiskey at murija7:~/Desktop/src/openAV-Fabla$ i have no idea, what it is meaning.... cheers, doc From harryhaaren at gmail.com Fri Jul 19 16:40:59 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 19 Jul 2013 17:40:59 +0100 Subject: [LAU] OpenAV Fabla In-Reply-To: <51E966FB.8080403@gmx.net> References: <51E966FB.8080403@gmx.net> Message-ID: On Fri, Jul 19, 2013 at 5:19 PM, Dragan Noveski wrote: > i tried to compile fabla, but there is an error: > Lets fix this on the Issues page of Fabla: https://github.com/harryhaaren/openAV-Fabla/issues i have no idea, what it is meaning.... > I've created an issue for this error, see solution available on the page too. https://github.com/harryhaaren/openAV-Fabla/issues/1 Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at quirq.net Fri Jul 19 18:24:52 2013 From: lists at quirq.net (Q) Date: Fri, 19 Jul 2013 19:24:52 +0100 Subject: [LAU] OpenAV Fabla In-Reply-To: References: Message-ID: <51E98474.20107@quirq.net> On 19/07/13 12:17, Harry van Haaren wrote: > > Or if its crafty beat programming your after that?s cool too! > That should be "Or if it's crafty beat programming you're after..." Q From jeremy at autostatic.com Fri Jul 19 19:23:05 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 19 Jul 2013 21:23:05 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: Message-ID: <51E99219.5070205@autostatic.com> On 07/19/2013 01:17 PM, Harry van Haaren wrote: > Hi, > > Its my pleasure to announce the release of Fabla! > > After 8 days we have reached the target donation amount, many thanks to all > those who contributed! > > Available here: http://openavproductions.com/fabla > > Cheers! -Harry Harry, under what license did you release Fabla? 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 Fri Jul 19 19:23:56 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Fri, 19 Jul 2013 21:23:56 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: Message-ID: <51E9924C.9050107@autostatic.com> On 07/19/2013 01:17 PM, Harry van Haaren wrote: > Hi, > > Its my pleasure to announce the release of Fabla! > > After 8 days we have reached the target donation amount, many thanks to all > those who contributed! > > Available here: http://openavproductions.com/fabla > > Cheers! -Harry > License is GPL2+, never mind. 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 ralf.mardorf at alice-dsl.net Fri Jul 19 20:37:29 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 19 Jul 2013 22:37:29 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: <51E966FB.8080403@gmx.net> Message-ID: <1374266249.2343.7.camel@archlinux> Hi Harry, it does build on Arch 64-bit, but there's no GUI if I run it by the script or with Qtractor as host. FWIW Sourcer does run with a GUI, but I didn't test if it really does work. The Fabla window is transparent. On Fri, 2013-07-19 at 17:40 +0100, Harry van Haaren wrote: > https://github.com/harryhaaren/openAV-Fabla/issues/1 [rocketmouse at archlinux openAV-Fabla]$ pacman -Q jalv ntk-git cairo cairomm libsndfile lv2-svn jalv 1.4.0-1 ntk-git 1:81f41f5-1 cairo 1.12.14-4 cairomm 1.10.0-3 libsndfile 1.0.25-2 lv2-svn 882-1 [rocketmouse at archlinux openAV-Fabla]$ ./run.sh Plugin: http://www.openavproductions.com/fabla UI: http://www.openavproductions.com/fabla/gui JACK Name: Fabla Block length: 256 frames MIDI buffers: 32768 bytes Comm buffers: 131072 bytes Update rate: 2 Hz volume = 0.750000 compressor_attack = 0.150000 compressor_decay = 0.300000 compressor_threshold = 0.500000 compressor_ratio = 0.500000 compressor_makeup = 0.500000 compressor_enable = 0.000000 pad_gain_1 = 0.500000 pad_gain_2 = 0.500000 pad_gain_3 = 0.500000 pad_gain_4 = 0.500000 pad_gain_5 = 0.500000 pad_gain_6 = 0.500000 pad_gain_7 = 0.500000 pad_gain_8 = 0.500000 pad_gain_9 = 0.500000 pad_gain_10 = 0.500000 pad_gain_11 = 0.500000 pad_gain_12 = 0.500000 pad_gain_13 = 0.500000 pad_gain_14 = 0.500000 pad_gain_15 = 0.500000 pad_gain_16 = 0.500000 pad_speed_1 = 0.500000 pad_speed_2 = 0.500000 pad_speed_3 = 0.500000 pad_speed_4 = 0.500000 pad_speed_5 = 0.500000 pad_speed_6 = 0.500000 pad_speed_7 = 0.500000 pad_speed_8 = 0.500000 pad_speed_9 = 0.500000 pad_speed_10 = 0.500000 pad_speed_11 = 0.500000 pad_speed_12 = 0.500000 pad_speed_13 = 0.500000 pad_speed_14 = 0.500000 pad_speed_15 = 0.500000 pad_speed_16 = 0.500000 pad_pan_1 = 0.500000 pad_pan_2 = 0.500000 pad_pan_3 = 0.500000 pad_pan_4 = 0.500000 pad_pan_5 = 0.500000 pad_pan_6 = 0.500000 pad_pan_7 = 0.500000 pad_pan_8 = 0.500000 pad_pan_9 = 0.500000 pad_pan_10 = 0.500000 pad_pan_11 = 0.500000 pad_pan_12 = 0.500000 pad_pan_13 = 0.500000 pad_pan_14 = 0.500000 pad_pan_15 = 0.500000 pad_pan_16 = 0.500000 pad_attack_1 = 0.000000 pad_attack_2 = 0.000000 pad_attack_3 = 0.000000 pad_attack_4 = 0.000000 pad_attack_5 = 0.000000 pad_attack_6 = 0.000000 pad_attack_7 = 0.000000 pad_attack_8 = 0.000000 pad_attack_9 = 0.000000 pad_attack_10 = 0.000000 pad_attack_11 = 0.000000 pad_attack_12 = 0.000000 pad_attack_13 = 0.000000 pad_attack_14 = 0.000000 pad_attack_15 = 0.000000 pad_attack_16 = 0.000000 pad_decay_1 = 0.500000 pad_decay_2 = 0.500000 pad_decay_3 = 0.500000 pad_decay_4 = 0.500000 pad_decay_5 = 0.500000 pad_decay_6 = 0.500000 pad_decay_7 = 0.500000 pad_decay_8 = 0.500000 pad_decay_9 = 0.500000 pad_decay_10 = 0.500000 pad_decay_11 = 0.500000 pad_decay_12 = 0.500000 pad_decay_13 = 0.500000 pad_decay_14 = 0.500000 pad_decay_15 = 0.500000 pad_decay_16 = 0.500000 pad_sustain_1 = 1.000000 pad_sustain_2 = 1.000000 pad_sustain_3 = 1.000000 pad_sustain_4 = 1.000000 pad_sustain_5 = 1.000000 pad_sustain_6 = 1.000000 pad_sustain_7 = 1.000000 pad_sustain_8 = 1.000000 pad_sustain_9 = 1.000000 pad_sustain_10 = 1.000000 pad_sustain_11 = 1.000000 pad_sustain_12 = 1.000000 pad_sustain_13 = 1.000000 pad_sustain_14 = 1.000000 pad_sustain_15 = 1.000000 pad_sustain_16 = 1.000000 pad_release_1 = 0.500000 pad_release_2 = 0.500000 pad_release_3 = 0.500000 pad_release_4 = 0.500000 pad_release_5 = 0.500000 pad_release_6 = 0.500000 pad_release_7 = 0.500000 pad_release_8 = 0.500000 pad_release_9 = 0.500000 pad_release_10 = 0.500000 pad_release_11 = 0.500000 pad_release_12 = 0.500000 pad_release_13 = 0.500000 pad_release_14 = 0.500000 pad_release_15 = 0.500000 pad_release_16 = 0.500000 JackEngine::XRun: client = Fabla was not finished, state = Running JackAudioDriver::ProcessGraphAsyncMaster: Process error Regards, Ralf From harryhaaren at gmail.com Fri Jul 19 21:33:23 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Fri, 19 Jul 2013 22:33:23 +0100 Subject: [LAU] OpenAV Fabla In-Reply-To: <1374266249.2343.7.camel@archlinux> References: <51E966FB.8080403@gmx.net> <1374266249.2343.7.camel@archlinux> Message-ID: On Fri, Jul 19, 2013 at 9:37 PM, Ralf Mardorf wrote: > it does build on Arch 64-bit, but there's no GUI if I run it by the > script or with Qtractor as host. FWIW Sourcer does run with a GUI, but I > Sorcer uses GTK, Fabla uses NTK, hence the difference. > On Fri, 2013-07-19 at 17:40 +0100, Harry van Haaren wrote: > > https://github.com/harryhaaren/openAV-Fabla/issues/1 > Please file an issue for this, although it seems you need to update your Jalv install. Cheers, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 19 22:44:51 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 20 Jul 2013 00:44:51 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: <51E966FB.8080403@gmx.net> <1374266249.2343.7.camel@archlinux> Message-ID: <1374273891.7169.19.camel@archlinux> On Fri, 2013-07-19 at 22:33 +0100, Harry van Haaren wrote: > On Fri, Jul 19, 2013 at 9:37 PM, Ralf Mardorf > wrote: > it does build on Arch 64-bit, but there's no GUI if I run it > by the > script or with Qtractor as host. FWIW Sourcer does run with a > GUI, but I > Sorcer uses GTK, Fabla uses NTK, hence the difference. > > > On Fri, 2013-07-19 at 17:40 +0100, Harry van Haaren wrote: > > https://github.com/harryhaaren/openAV-Fabla/issues/1 > > > Please file an issue for this, although it seems you need to update > your Jalv install. Thank you, while you replied I rebuild ntk and then noticed it already was up to date :D. $ git log --pretty=format:"%h - %an, %ar : %s" 81f41f5 - Jonathan Moore Liles, 6 days ago : Fl_Choice: Respect specified boxtypes. Also get rid of old 'scheme' [snip] $ svn co http://svn.drobilla.net/lad/trunk/jalv jalv [snip] Checked out revision 5144. $ yaourt -S jalv-svn [snip] I get "error: target not found: lv2-ui-resize" for the AUR default pkgver=3882 and also if I edit it to 5144. I've got no time to continue now and to file an issue I need to sign up :(. I have to delay this, perhaps I can do it tomorrow or next week. I'm in passing now, sorry. Googling for lv2-ui-resize does show that the AUR package is missing, so I have to file an issue there too. Regards, Ralf From marc at hacklava.net Fri Jul 19 23:14:20 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Fri, 19 Jul 2013 19:14:20 -0400 Subject: [LAU] Zita-ajbridge ? Message-ID: <20130719191420.6a2d5ba4@hacklava.net> Good evening LAU. This is my first post to the list. For some reason, zita-j2a 0.2.2 is silent on my laptop. I tried it with the built-in sound card and also with a USB module (both are 16bit/48KHz). I started jackd with the alsa and the dummy backends, without success. I even recompiled zita-j2a. What could be the cause? Thanks in advance. -- Marc From marc at hacklava.net Sat Jul 20 02:33:26 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Fri, 19 Jul 2013 22:33:26 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: References: <20130719191420.6a2d5ba4@hacklava.net> Message-ID: <20130719223326.67149ede@hacklava.net> Hi Len. I would like to use jackd in 32bit (or is it 24bit, or float?) but output in 16bit (resampled with zita-j2a). This is to play 24bit files with the best possible quality when using multiple 16bit sound cards or USB modules. When starting jackd normally (with the alsa backend and a 16bit sound card), mplayer output is 16bit (that's not what I want). When starting jackd with the dummy backend, or the alsa backend connected to a loop device, jackd works in 32bit float and mplayer output is floatle; jackd doesn't report sync problems and mplayer just plays the file. Then I connect the floatle output of mplayer to zita-j2a, there's no sound. Even when starting jackd with the internal sound card, zita-j2a stays silent. -- Marc Len Ovens a ?crit : > On Fri, 19 Jul 2013, Marc Lavall?e wrote: > > > For some reason, zita-j2a 0.2.2 is silent on my laptop. I tried it > > with the built-in sound card and also with a USB module (both are > > 16bit/48KHz). I started jackd with the alsa and the dummy backends, > > without success. I even recompiled zita-j2a. What could be the > > cause? > > Dummy back end? Jackd has to be synced to something. Normally, the > use for zita-a2j is to add a second audio card. So jackd must first > be connaected to a first card. > > Len From ralf.mardorf at alice-dsl.net Sat Jul 20 07:54:21 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 20 Jul 2013 09:54:21 +0200 Subject: [LAU] OpenAV Fabla In-Reply-To: References: <51E966FB.8080403@gmx.net> <1374266249.2343.7.camel@archlinux> Message-ID: <1374306861.1212.9.camel@archlinux> On Fri, 2013-07-19 at 22:33 +0100, Harry van Haaren wrote: > Please file an issue for this, although it seems you need to update > your Jalv install. https://github.com/harryhaaren/openAV-Fabla/issues/2 https://aur.archlinux.org/packages/jalv-svn/ From harryhaaren at gmail.com Sat Jul 20 08:52:41 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sat, 20 Jul 2013 09:52:41 +0100 Subject: [LAU] OpenAV Fabla In-Reply-To: <1374306861.1212.9.camel@archlinux> References: <51E966FB.8080403@gmx.net> <1374266249.2343.7.camel@archlinux> <1374306861.1212.9.camel@archlinux> Message-ID: On Sat, Jul 20, 2013 at 8:54 AM, Ralf Mardorf wrote: > https://github.com/harryhaaren/openAV-Fabla/issues/2 > Updates and fixing will take place on the github page, I've pushed a commit and a note. Activity on the issue is emailed to me directly, so I'll be notified if anybody responds. I'm looking into what causes this issue, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sat Jul 20 11:42:13 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 20 Jul 2013 11:42:13 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130719191420.6a2d5ba4@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> Message-ID: <20130720114213.GA11334@linuxaudio.org> On Fri, Jul 19, 2013 at 07:14:20PM -0400, Marc Lavall?e wrote: > For some reason, zita-j2a 0.2.2 is silent on my laptop. I tried it > with the built-in sound card and also with a USB module (both are > 16bit/48KHz). I started jackd with the alsa and the dummy backends, > without success. I even recompiled zita-j2a. What could be the cause? First of all, what are you trying to achieve ? Zita-j2a is normally used to add a playback alsa device to Jack. For example, if you use Jack with the internal soundcard, then starting zita-j2a with the USB card will add the outputs of the USB card to jack, you can then play via both cards at the same time. Second, if you want help you need to provide a bit more information. What are the commands you use to start jack and zita-j2a ? What is the output (using the -v option) of zita-j2a ? etc. etc. 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 Sat Jul 20 11:47:43 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 20 Jul 2013 11:47:43 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130719223326.67149ede@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> Message-ID: <20130720114743.GB11334@linuxaudio.org> On Fri, Jul 19, 2013 at 10:33:26PM -0400, Marc Lavall?e wrote: > I would like to use jackd in 32bit (or is it 24bit, or float?) but > output in 16bit (resampled with zita-j2a). This is to play 24bit files > with the best possible quality when using multiple 16bit sound cards or > USB modules. If you use mplayer with jack output (-ao jack) then it will *always* output in floating point format because that is the only format that Jack accepts. And mplayer does not know which soundcard Jack is using. Which means that > When starting jackd normally (with the alsa backend and a 16bit sound > card), mplayer output is 16bit (that's not what I want). can't be true. As said, Jack is *always* using floating point. It will convert the playback channels to whatever the sound card can handle. 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 marc at hacklava.net Sat Jul 20 13:36:22 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sat, 20 Jul 2013 09:36:22 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720114743.GB11334@linuxaudio.org> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> Message-ID: <20130720093622.74687e7e@hacklava.net> Bonjour Fons. Fons Adriaensen a ?crit : > First of all, what are you trying to achieve ? I don't want to use "depth down-sampled" channels with ambdec. > Zita-j2a is normally used to add a playback alsa device to Jack. > For example, if you use Jack with the internal soundcard, then > starting zita-j2a with the USB card will add the outputs of the > USB card to jack, you can then play via both cards at the same time. As I wrote, zita-j2a stays silent when using it as you explained. > As said, Jack is *always* using floating point. Excellent! But when running jackd with my 16bit USB module, Mplayer is converting its output to 16bit. When running Jack with a 24bit sound card or the aloop device, Mplayer's output is floatle (as expected), even when forcing Jack to run in 16bit with the "--short" option. Another problem: when starting Jack with a 2 channels sound card, Mplayer can't output more than 2 channels. So I suppose that Mplayer is misbehaving. > It will convert the playback channels to whatever the sound card can > handle. So far, no sound is coming out of zita-j2a. Here's how I started it: zita-j2a -v -d hw:2 -c 8 playback : nchan : 8 fsamp : 48000 fsize : 256 nfrag : 2 format : S16_LE capture : not enabled Starting synchronisation. Tchao -- Marc From marc at hacklava.net Sat Jul 20 14:08:03 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sat, 20 Jul 2013 10:08:03 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720093622.74687e7e@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> Message-ID: <20130720100803.5d92b56a@hacklava.net> Update: I have no idea why MPlayer now work as expected (with floatle outputs) when starting Jack with my 16 bit USB module: jackd -r -d alsa -d hw:2 -P -o 8 -z shaped Although it would have been nice to use zita-j2a for resampling... What was the problem then? I hope it's not intermittent, and that it's not only related to the device between my ears. :) Then I tried to add the 2 channels internal sound device: zita-j2a -v -d hw:1 (no sound) zita-j2a -v -d hw:1 -L (no sound) Salut -- Marc I wrote: > Bonjour Fons. > > Fons Adriaensen a ?crit : > > First of all, what are you trying to achieve ? > > I don't want to use "depth down-sampled" channels with ambdec. > > > Zita-j2a is normally used to add a playback alsa device to Jack. > > For example, if you use Jack with the internal soundcard, then > > starting zita-j2a with the USB card will add the outputs of the > > USB card to jack, you can then play via both cards at the same time. > > As I wrote, zita-j2a stays silent when using it as you explained. > > > As said, Jack is *always* using floating point. > > Excellent! > > But when running jackd with my 16bit USB module, Mplayer is converting > its output to 16bit. When running Jack with a 24bit sound card or the > aloop device, Mplayer's output is floatle (as expected), even when > forcing Jack to run in 16bit with the "--short" option. > > Another problem: when starting Jack with a 2 channels sound card, > Mplayer can't output more than 2 channels. > > So I suppose that Mplayer is misbehaving. > > > It will convert the playback channels to whatever the sound card can > > handle. > > So far, no sound is coming out of zita-j2a. > Here's how I started it: > zita-j2a -v -d hw:2 -c 8 > playback : > nchan : 8 > fsamp : 48000 > fsize : 256 > nfrag : 2 > format : S16_LE > capture : not enabled > Starting synchronisation. > > Tchao > -- > Marc From fons at linuxaudio.org Sat Jul 20 14:19:47 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 20 Jul 2013 14:19:47 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720100803.5d92b56a@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> Message-ID: <20130720141947.GC11334@linuxaudio.org> On Sat, Jul 20, 2013 at 10:08:03AM -0400, Marc Lavall?e wrote: > Update: > > I have no idea why MPlayer now work as expected (with floatle outputs) > when starting Jack with my 16 bit USB module: > jackd -r -d alsa -d hw:2 -P -o 8 -z shaped Why do you use -r ? Does your system not allow real-time ? Why -o 8 ? Even if you have 8 outputs that is not necessary. > Although it would have been nice to use zita-j2a for resampling... There is no need to resample anything. > What was the problem then? I hope it's not intermittent, > and that it's not only related to the device between my ears. :) > > Then I tried to add the 2 channels internal sound device: > zita-j2a -v -d hw:1 > (no sound) > zita-j2a -v -d hw:1 -L > (no sound) Does hw:1 work with Jack ? Did you connect anything to the zita-j2a ports ? 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 Sat Jul 20 14:23:09 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 20 Jul 2013 14:23:09 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720093622.74687e7e@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> Message-ID: <20130720142309.GD11334@linuxaudio.org> On Sat, Jul 20, 2013 at 09:36:22AM -0400, Marc Lavall?e wrote: > But when running jackd with my 16bit USB module, Mplayer is converting > its output to 16bit. When running Jack with a 24bit sound card or the > aloop device, Mplayer's output is floatle (as expected), even when > forcing Jack to run in 16bit with the "--short" option. As said, unless you have some voodoo SW on your machine, that is not possible. If mplayer uses jack, it doesn not know anything about the sound card. Not the sample format, not the number of channels, only the sample frequency. > Another problem: when starting Jack with a 2 channels sound card, > Mplayer can't output more than 2 channels. Again, mplayer can't know this. Your are probably running on some other soundcard, not using Jack. 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 marc at hacklava.net Sat Jul 20 16:26:33 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sat, 20 Jul 2013 12:26:33 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720141947.GC11334@linuxaudio.org> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> Message-ID: <20130720122633.73cfeee0@hacklava.net> Hi again Fons. Fons Adriaensen a ?crit : > > jackd -r -d alsa -d hw:2 -P -o 8 -z shaped > > Why do you use -r ? Does your system not allow real-time ? Oops. I use a generic 3.7 kernel that allow real-time, so -r is bad. The correct switch is -R (default). > Why -o 8 ? Even if you have 8 outputs that is not necessary. Oops again; all channels are available without -o 8. > There is no need to resample anything. Right... This is your use case for zita-resampler: "The development of zita-resampler was triggered by the need to resample multichannel files (HOA, 25-ish channels), while still keeping some CPU capacity for other tasks." So I don't need it. Less is more! :) > Does hw:1 work with Jack ? Yes. > Did you connect anything to the zita-j2a ports ? Yes. > As said, unless you have some voodoo SW on your machine, that is not > possible. If mplayer uses jack, it doesn not know anything about the > sound card. Not the sample format, not the number of channels, only > the sample frequency. So Mplayer is just reporting the output frequency of Jack? > > Another problem: when starting Jack with a 2 channels sound card, > > Mplayer can't output more than 2 channels. > > Again, mplayer can't know this. Your are probably running on > some other soundcard, not using Jack. I confirm: when using Jack with the internal soundcard (with 2 output channels), Mplayer limit its number of output channels to 2, even if I specify the number of channels of my sound file (4) to Mplayer. Update: - I installed the latest Jackd (version 1.9.10 from Grame): zita-j2a now works! :) after the "Starting synchronisation" message, there's now a continuous sequence of numbers. Suggestion: a note in the documentation about the version of Jackd would help. - I installed the latest Mplayer (2): same behaviour... Thanks for you help. N.B. The descriptions in the man pages are inverted: zita-j2a - Use ALSA capture device as a Jack client, with resampling. zita-a2j - Use ALSA playback device as a Jack client, with resampling. -- Marc From fons at linuxaudio.org Sat Jul 20 17:52:18 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 20 Jul 2013 17:52:18 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720122633.73cfeee0@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> <20130720122633.73cfeee0@hacklava.net> Message-ID: <20130720175218.GE11334@linuxaudio.org> On Sat, Jul 20, 2013 at 12:26:33PM -0400, Marc Lavall?e wrote: > > There is no need to resample anything. > > Right... This is your use case for zita-resampler: > "The development of zita-resampler was triggered by the need to > resample multichannel files (HOA, 25-ish channels), while still keeping > some CPU capacity for other tasks." I mean: there is no need to resample if you want to use mplayer with Jack. If the audio file's sample rate doesn't match that of Jack, mplayer will do the resampling. > So Mplayer is just reporting the output frequency of Jack? Both actually: ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== AO: [jack] 48000Hz 2ch floatle (4 bytes per sample) > > > Another problem: when starting Jack with a 2 channels sound card, > > > Mplayer can't output more than 2 channels. > > > > Again, mplayer can't know this. Your are probably running on > > some other soundcard, not using Jack. > > I confirm: when using Jack with the internal soundcard (with 2 output > channels), Mplayer limit its number of output channels to 2, even if I > specify the number of channels of my sound file (4) to Mplayer. Because it wants to connect to four system:playback ports, and that fails. And when the connection fails, mplayer deletes the output port. Which is insane, but that's how it is. Start Jack with two playback channels, and any program having at least four jack inputs, e.g. ambdec. Then try mplayer -ao jack:port=ambdec -channels 4 YOUR_4_CH_FILE and mplayer will have four output ports and connect them to ambdec's inputs, even if your sound card has only two outputs. The real problem is that mplayer doesn't allow you to specify the connections, they could be in any order. So chances are 1 in 24 that they will be correct in this case. Mplayer is one of the many apps that claim Jack support but get it all wrong. > - I installed the latest Jackd (version 1.9.10 from Grame): zita-j2a now works! :) What was your previous Jack version ? > after the "Starting synchronisation" message, there's now a > continuous sequence of numbers. Probably because you still use -v. See the README for what they mean. > Suggestion: a note in the documentation about the version of > Jackd would help. It shouldn't matter, unless it's *very* old. Zita-ajbridge requires nothing special from Jack, apart from a working DLL. And that was added at least five years ago. 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 alice-dsl.net Sat Jul 20 21:13:22 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 20 Jul 2013 23:13:22 +0200 Subject: [LAU] Screen blanking Message-ID: <1374354802.690.11.camel@archlinux> Hi :) how can screen blanking be disabled? A long time ago I have given up to check this out for my Ubuntu Studio installs. FWIW it's not the monitor, for an old Suse 11.2 install the screen never will be turned off. Now I try to disable screen blanking on Arch Linux, Xfce 4.10. $ uname -rm 3.8.13-rt14-1-rt x86_64 This doesn't work: $ cat ~/.xinitrc #!/bin/sh # # ~/.xinitrc # # Executed by startx (run your window manager from here) if [ -d /etc/X11/xinit/xinitrc.d ]; then for f in /etc/X11/xinit/xinitrc.d/*; do [ -x "$f" ] && . "$f" done unset f fi xset -dpms setterm -blank 0 -powersave off -powerdown 0 xset s off exec startxfce4 # exec xterm Power Manager is enabled when starting a session, but Menu > Settings > Settings Manager > Power Manager > On AC > All "Actions" and "Monitor" settings are set to "never". $ cat /etc/X11/xorg.conf Section "Module" Load "extmod" Load "dri" Load "dbe" Load "dri2" Load "glx" Load "record" Load "GLcore" #Load "v4l" EndSection Section "Monitor" Identifier "Monitor0" VendorName "Plug 'n' Play" ModelName "Plug 'n' Play" DisplaySize 305 230 HorizSync 29-98 VertRefresh 50-120 modeline "1152x864" 128.42 1152 1232 1360 1568 864 865 868 910 Gamma 1.0 EndSection Section "Device" ### Available Driver options are:- ### Values: : integer, : float, : "True"/"False", ### : "String", : " Hz/kHz/MHz" ### [arg]: arg optional #Option "SWcursor" # [] #Option "HWcursor" # [] #Option "NoAccel" # [] #Option "ShadowFB" # [] #Option "VideoKey" # #Option "MergedFB" "off" #old debian/ubuntu Identifier "Card0" Driver "radeon" #Driver "nvidia" #Driver "nv" #Driver "nouveau" #Driver "vesa" #VendorName "nVidia Corporation" #BoardName "G72 [GeForce 7300 SE/7200 GS]" #BusID "PCI:1:0:0" EndSection Section "Screen" Identifier "Screen0" Device "Card0" Monitor "Monitor0" Defaultdepth 24 SubSection "Display" Depth 24 Modes "1152x864" #Virtual 3840 1200 EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 1 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 4 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 8 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 15 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 16 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 24 # EndSubSection EndSection $ ls /etc/X11 xinit xorg.conf xorg.conf.d $ ls /etc/X11/xinit/ xinitrc xinitrc.d xserverrc $ ls /etc/X11/xorg.conf.d/ 10-evdev.conf 10-quirks.conf keyboard.conf $ ls /etc/X11/xinit/xinitrc.d/ 30-dbus 40-libcanberra-gtk-module Regards, Ralf From ralf.mardorf at alice-dsl.net Sat Jul 20 21:43:24 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sat, 20 Jul 2013 23:43:24 +0200 Subject: [LAU] [solved] Screen blanking In-Reply-To: <1374354802.690.11.camel@archlinux> References: <1374354802.690.11.camel@archlinux> Message-ID: <1374356604.692.3.camel@archlinux> I guess it's solved. No screen blanking after 10 minutes, as it was before, after editing ~/.xinitrc and /etc/X11/xorg.conf. $ cat ~/.xinitrc #!/bin/sh # # ~/.xinitrc # # Executed by startx (run your window manager from here) if [ -d /etc/X11/xinit/xinitrc.d ]; then for f in /etc/X11/xinit/xinitrc.d/*; do [ -x "$f" ] && . "$f" done unset f fi xset dpms 0 0 0 setterm -blank 0 -powersave off -powerdown 0 xset s off exec startxfce4 # exec xterm $ cat /etc/X11/xorg.conf Section "Module" Load "extmod" Load "dri" Load "dbe" Load "dri2" Load "glx" Load "record" Load "GLcore" #Load "v4l" EndSection Section "Monitor" Identifier "Monitor0" VendorName "Plug 'n' Play" ModelName "Plug 'n' Play" DisplaySize 305 230 HorizSync 29-98 VertRefresh 50-120 modeline "1152x864" 128.42 1152 1232 1360 1568 864 865 868 910 Gamma 1.0 EndSection Section "Device" ### Available Driver options are:- ### Values: : integer, : float, : "True"/"False", ### : "String", : " Hz/kHz/MHz" ### [arg]: arg optional #Option "SWcursor" # [] #Option "HWcursor" # [] #Option "NoAccel" # [] #Option "ShadowFB" # [] #Option "VideoKey" # #Option "MergedFB" "off" #old debian/ubuntu Identifier "Card0" Driver "radeon" #Driver "nvidia" #Driver "nv" #Driver "nouveau" #Driver "vesa" #VendorName "nVidia Corporation" #BoardName "G72 [GeForce 7300 SE/7200 GS]" #BusID "PCI:1:0:0" EndSection Section "Screen" Identifier "Screen0" Device "Card0" Monitor "Monitor0" Defaultdepth 24 SubSection "Display" Depth 24 Modes "1152x864" #Virtual 3840 1200 EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 1 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 4 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 8 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 15 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 16 # EndSubSection # SubSection "Display" # Viewport 0 0 # Depth 24 # EndSubSection EndSection Section "ServerFlags" Option "BlankTime" "0" Option "StandbyTime" "0" Option "SuspendTime" "0" Option "OffTime" "0" EndSection From willgodfrey at musically.me.uk Sat Jul 20 21:55:06 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Sat, 20 Jul 2013 22:55:06 +0100 Subject: [LAU] [solved] Screen blanking In-Reply-To: <1374356604.692.3.camel@archlinux> References: <1374354802.690.11.camel@archlinux> <1374356604.692.3.camel@archlinux> Message-ID: <20130720225506.2f38ad0a@debian> On Sat, 20 Jul 2013 23:43:24 +0200 Ralf Mardorf wrote: > I guess it's solved. No screen blanking after 10 minutes, as it was > before, after editing ~/.xinitrc and /etc/X11/xorg.conf. > > $ cat ~/.xinitrc > #!/bin/sh > # > # ~/.xinitrc > # > # Executed by startx (run your window manager from here) > > if [ -d /etc/X11/xinit/xinitrc.d ]; then > for f in /etc/X11/xinit/xinitrc.d/*; do > [ -x "$f" ] && . "$f" > done > unset f > fi > > xset dpms 0 0 0 > setterm -blank 0 -powersave off -powerdown 0 > xset s off > > exec startxfce4 > # exec xterm > > $ cat /etc/X11/xorg.conf > Section "Module" > Load "extmod" > Load "dri" > Load "dbe" > Load "dri2" > Load "glx" > Load "record" > Load "GLcore" > #Load "v4l" > EndSection > Section "Monitor" > Identifier "Monitor0" > VendorName "Plug 'n' Play" > ModelName "Plug 'n' Play" > DisplaySize 305 230 > HorizSync 29-98 > VertRefresh 50-120 > modeline "1152x864" 128.42 1152 1232 1360 1568 864 865 868 910 > Gamma 1.0 > EndSection > > Section "Device" > ### Available Driver options are:- > ### Values: : integer, : float, : "True"/"False", > ### : "String", : " Hz/kHz/MHz" > ### [arg]: arg optional > #Option "SWcursor" # [] > #Option "HWcursor" # [] > #Option "NoAccel" # [] > #Option "ShadowFB" # [] > #Option "VideoKey" # > #Option "MergedFB" "off" #old debian/ubuntu > Identifier "Card0" > Driver "radeon" > #Driver "nvidia" > #Driver "nv" > #Driver "nouveau" > #Driver "vesa" > #VendorName "nVidia Corporation" > #BoardName "G72 [GeForce 7300 SE/7200 GS]" > #BusID "PCI:1:0:0" > EndSection > > Section "Screen" > Identifier "Screen0" > Device "Card0" > Monitor "Monitor0" > > Defaultdepth 24 > SubSection "Display" > Depth 24 > Modes "1152x864" > #Virtual 3840 1200 > EndSubSection > > # SubSection "Display" > # Viewport 0 0 > # Depth 1 > # EndSubSection > # SubSection "Display" > # Viewport 0 0 > # Depth 4 > # EndSubSection > # SubSection "Display" > # Viewport 0 0 > # Depth 8 > # EndSubSection > # SubSection "Display" > # Viewport 0 0 > # Depth 15 > # EndSubSection > # SubSection "Display" > # Viewport 0 0 > # Depth 16 > # EndSubSection > # SubSection "Display" > # Viewport 0 0 > # Depth 24 > # EndSubSection > EndSection > > Section "ServerFlags" > Option "BlankTime" "0" > Option "StandbyTime" "0" > Option "SuspendTime" "0" > Option "OffTime" "0" > EndSection > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I can't for the life of me remember what it is, but (for debian at least) there is in fact a very simple one line alteration that does this! -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From harryhaaren at gmail.com Sat Jul 20 22:35:34 2013 From: harryhaaren at gmail.com (Harry van Haaren) Date: Sat, 20 Jul 2013 23:35:34 +0100 Subject: [LAU] [solved] Screen blanking In-Reply-To: <20130720225506.2f38ad0a@debian> References: <1374354802.690.11.camel@archlinux> <1374356604.692.3.camel@archlinux> <20130720225506.2f38ad0a@debian> Message-ID: On Sat, Jul 20, 2013 at 10:55 PM, Will Godfrey wrote: > I can't for the life of me remember what it is, but (for debian at least) there > is in fact a very simple one line alteration that does this! On Arch I have this little script to disable: I'm pretty sure it will work X-distro too. xset is the only command used: https://github.com/harryhaaren/scripts/blob/master/archLinux/screensaverDisable.sh HTH, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Sat Jul 20 23:04:02 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 21 Jul 2013 01:04:02 +0200 Subject: [LAU] [solved] Screen blanking In-Reply-To: References: <1374354802.690.11.camel@archlinux> <1374356604.692.3.camel@archlinux> <20130720225506.2f38ad0a@debian> Message-ID: <1374361442.692.8.camel@archlinux> On Sat, 2013-07-20 at 23:35 +0100, Harry van Haaren wrote: > On Arch I have this little script to disable: I'm pretty sure it will > work X-distro too. xset is the only command used: > https://github.com/harryhaaren/scripts/blob/master/archLinux/screensaverDisable.sh This is quasi what I run by my first .xinitrc and it didn't work. It's definitive solved now. What exactly solved the issue, the new .xinitrc or the new xorg.conf will be tested within the next days. Hopefully it does work for Ubuntu and Debian too. Thank you Will and Harry. Regards, Ralf -- Great guitar riff, 25 or 6 to 4- Chicago: http://www.youtube.com/watch?v=WLiuMkGCOC4 From marc at hacklava.net Sun Jul 21 00:50:35 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sat, 20 Jul 2013 20:50:35 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720175218.GE11334@linuxaudio.org> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> <20130720122633.73cfeee0@hacklava.net> <20130720175218.GE11334@linuxaudio.org> Message-ID: <20130720205035.3c852f6a@hacklava.net> Fons Adriaensen a ?crit : > I mean: there is no need to resample if you want to use mplayer with > Jack. If the audio file's sample rate doesn't match that of Jack, > mplayer will do the resampling. Mmhh... I'd prefer everything in 24bit until the output to alsa; I don't want to provide 16bit channels to ambdec from 24bit amb files. But then, is it a valid concern? I have the feeling that it doesn't matter much... > Start Jack with two playback channels, and any program having at least > four jack inputs, e.g. ambdec. Then try > > mplayer -ao jack:port=ambdec -channels 4 YOUR_4_CH_FILE > > and mplayer will have four output ports and connect them to ambdec's > inputs, even if your sound card has only two outputs. It works as you described. > The real problem is that mplayer doesn't allow you to specify the > connections, they could be in any order. So chances are 1 in 24 that > they will be correct in this case. Mplayer is one of the many apps > that claim Jack support but get it all wrong. For FOA, mapping of amb files is WXYZ, while ambdec inputs are 0w,1y,1z,1x (why?), so the connections are wrong (W-0w,X-1y,Y-1z,Z-1x). Here's how I remapped the outputs: mplayer -loop 0 -channels 4 -af channels=4:4:0:0:1:3:2:1:3:2 -ao jack:port=ambdec AJH_eight-positions.amb I will write a script to support other amb channels mapping with mplayer (up to fff in Malham notation). Now my problem is to connect mplayer to a second instance of ambdec; I need it for lower frequencies, and since I use a triangle (for basic horizontal decoding), I had to change the lower limit for the number of allowed channels in the source code of ambdec. > > - I installed the latest Jackd (version 1.9.10 from Grame): > > zita-j2a now works! :) > > What was your previous Jack version ? Version 0.121. It was kept on my system by some old software. I uninstalled all software with dependencies to Jack, then I was able to install a newer version of Jack. > > after the "Starting synchronisation" message, there's now a > > continuous sequence of numbers. > > Probably because you still use -v. See the README for what they mean. Yes, it's useful to verify if it works. > > Suggestion: a note in the documentation about the version of > > Jackd would help. > > It shouldn't matter, unless it's *very* old. Zita-ajbridge requires > nothing special from Jack, apart from a working DLL. And that was > added at least five years ago. Ubuntu is still providing version 0.121, even in the newest development distribution (saucy), probably for compatibility issues. -- Marc From ralf.mardorf at alice-dsl.net Sun Jul 21 01:04:00 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 21 Jul 2013 03:04:00 +0200 Subject: [LAU] Attention! Attackers have .... Was: Zita-ajbridge ? Message-ID: <1374368640.819.13.camel@archlinux> On Sat, 2013-07-20 at 20:50 -0400, Marc Lavall?e wrote: > Ubuntu is still providing version 0.121 Since you mention Ubuntu, -------- Forwarded Message -------- From: Ali Linx (amjjawad) To: Xubuntu Support and User Discussions Subject: [xubuntu-users] Ubuntu Forum - FYI Date: Sun, 21 Jul 2013 04:38:37 +0400 http://ubuntuforums.org/announce.html -- Attention! Attackers have gotten every user's local username, password, and email address from the Ubuntu Forums database. http://ubuntuforums.org/announce.html From fons at linuxaudio.org Sun Jul 21 11:41:04 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 21 Jul 2013 11:41:04 +0000 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130720205035.3c852f6a@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> <20130720122633.73cfeee0@hacklava.net> <20130720175218.GE11334@linuxaudio.org> <20130720205035.3c852f6a@hacklava.net> Message-ID: <20130721114104.GA20621@linuxaudio.org> On Sat, Jul 20, 2013 at 08:50:35PM -0400, Marc Lavall?e wrote: > > mplayer will do the resampling. > > Mmhh... I'd prefer everything in 24bit until the output to alsa; I > don't want to provide 16bit channels to ambdec from 24bit amb files. > But then, is it a valid concern? I have the feeling that it doesn't > matter much... Resampling means 'computing samples for another sample rate', and not 'changing the sample format'. Anyway, in Jack *everything* is floating point until output to Alsa, just as you want. Again: mplayer, or any other Jack app, does not know the format required by the sound card, and when connected to Jack it always outputs in floating point format. > For FOA, mapping of amb files is WXYZ, while ambdec inputs are > 0w,1y,1z,1x (why?), See below. > Here's how I remapped the outputs: > > mplayer -loop 0 -channels 4 -af channels=4:4:0:0:1:3:2:1:3:2 -ao > jack:port=ambdec AJH_eight-positions.amb > I will write a script to support other amb channels mapping with > mplayer (up to fff in Malham notation). Neat trick, but if it works you are just lucky. ** There is no defined order when Jack provides a list of ports. ** In most cases, Jack will list them in the order they were created, but you can't count on that. The only way to obtain a defined order is by interpreting the port names, which is what e.g. Qjackctl does: it does a lexicographical sort. Which is why it will show Ambdec's ports as 0w, 1x, 1y, 1z. AFAIK, mplayer doesn't do that. > Now my problem is to connect mplayer to a second instance of ambdec; I > need it for lower frequencies, and since I use a triangle (for basic > horizontal decoding), I had to change the lower limit for the number of > allowed channels in the source code of ambdec. There's no need to do that, just set the matrix coefficients for the 4th channel to zero. About Ambisonics channel order ------------------------------ Spherical harmonics are identified by two numbers, 'degree' and 'order', usually notated 'l' and 'm'. [*] I'll use 'L' and 'M' here to avoid confusing 'l' and '1'. This is the mapping using the FuMa channel names: | -3 -2 -1 0 +1 +2 +3 M ---|-------------------------------------- 0 | W 1 | Y Z X 2 | V T R S U 3 | Q O M K L N P | L | The triangle gets wider as L increases, as there are 2 * L + 1 components of degree L. If you read it top to bottom and left to right, and start counting at zero, you get what is called the "Ambisonic channel number" or ACN: W = 0, Y = 1, Z = 2, X = 3, etc. Ambdec uses ACN internally, so it creates ports in that order, and then Jack will (usually) list them in the same order. When Ambdec gets extended to 4th degree (order) and higher, it will also use ACN in the port names - there are no FuMA names for those. The only alternative would be to use port names that explicitly contain L and M. Note that ACN = L^2 + L + M and when given ACN L = sqrt (ACN) rounded down, and then M = ACN - L^2 - L [*] To add to the confusion, in Ambisonics the 'degree' is usually called 'order'. 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 marc at hacklava.net Sun Jul 21 13:14:00 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sun, 21 Jul 2013 09:14:00 -0400 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130721114104.GA20621@linuxaudio.org> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> <20130720122633.73cfeee0@hacklava.net> <20130720175218.GE11334@linuxaudio.org> <20130720205035.3c852f6a@hacklava.net> <20130721114104.GA20621@linuxaudio.org> Message-ID: <20130721091400.46bd82dc@hacklava.net> Fons Adriaensen a ?crit : > Resampling means 'computing samples for another sample rate', and not > 'changing the sample format'. I thought that the sample format was the sample depth plus the sample rate, but sample format and sample depth are the same. Definitions are important. > Neat trick, but if it works you are just lucky. Gee. I just finished my script, and it works. > ** There is no defined order when Jack provides a list of ports. ** What?? Now... Heinseberg law applying to Jack... :-) > In most cases, Jack will list them in the order they were created, > but you can't count on that. I might just count on that. I like being lucky. :-) > The only way to obtain a defined order is by interpreting the port > names, which is what e.g. Qjackctl does: it does a lexicographical > sort. Which is why it will show Ambdec's ports as 0w, 1x, 1y, 1z. > AFAIK, mplayer doesn't do that. Isn't Qjackctl just a wrapper (or controller) for jackd? Qjackctl shows the ports in a different (lexicographical) order, but the order does not change when starting jackd with Qjackctl. > > Now my problem is to connect mplayer to a second instance of > > ambdec; I'll try to use the aloop devices instead of mangling the channels order with a bash script to create a remapping string for Mplayer. I should be able to script the connections between the ambdec decoders and the aloop devices, in order to use MPlayer with ALSA instead of Jack. > I need it for lower frequencies, and since I use a triangle > > (for basic horizontal decoding), I had to change the lower limit > > for the number of allowed channels in the source code of ambdec. > > There's no need to do that, just set the matrix coefficients for the > 4th channel to zero. Nice! I'll fix my triangle decoder; it's cheaper than replacing all my satellite speakers with full-range speakers, only for frequencies below 120Hz. > When Ambdec gets extended to 4th degree (order) and > higher, it will also use ACN in the port names - there are > no FuMA names for those. The only alternative would be to > use port names that explicitly contain L and M. I doubt I'll be using 4th degree anytime soon (my listening room is very small), but there might be a need to create a new amb format using the ACN ordering instead of the FuMA ordering. Or does it already exists? -- Marc From ralf.mardorf at alice-dsl.net Sun Jul 21 13:44:04 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 21 Jul 2013 15:44:04 +0200 Subject: [LAU] Zita-ajbridge ? In-Reply-To: <20130721091400.46bd82dc@hacklava.net> References: <20130719191420.6a2d5ba4@hacklava.net> <20130719223326.67149ede@hacklava.net> <20130720114743.GB11334@linuxaudio.org> <20130720093622.74687e7e@hacklava.net> <20130720100803.5d92b56a@hacklava.net> <20130720141947.GC11334@linuxaudio.org> <20130720122633.73cfeee0@hacklava.net> <20130720175218.GE11334@linuxaudio.org> <20130720205035.3c852f6a@hacklava.net> <20130721114104.GA20621@linuxaudio.org> <20130721091400.46bd82dc@hacklava.net> Message-ID: <1374414244.1445.41.camel@archlinux> On Sun, 2013-07-21 at 09:14 -0400, Marc Lavall?e wrote: > > ** There is no defined order when Jack provides a list of ports. ** > > What?? Now... Heinseberg law applying to Jack... :-) Broken analogy ;) since the ports are there and you can figure out at what position and you don't have another condition to take care about, so it's different from the uncertainty principle :p. SICR. -- Attention! Attackers have gotten every user's local username, password, and email address from the Ubuntu Forums database. http://ubuntuforums.org/announce.html From len at ovenwerks.net Mon Jul 22 00:55:54 2013 From: len at ovenwerks.net (Len Ovens) Date: Sun, 21 Jul 2013 17:55:54 -0700 Subject: [LAU] Headless jackdbus (no X11) Message-ID: I have in the past used something like a jackstart script in a headless situation before, which started dbus and exported it's info so jackdbus would work. It is actually easier to use: dbus-launch screen Then use screen as your CLI session manager. jack_control can then be used to run jack and change settings on the fly (like latency?). Audio applications can be run in other screen terminals to use jack... and if you really want to go crazy you can throw pulse at the mix... though I am really not sure what pulse supporting applications you hope to find that are CLI (I tested it with paplay, but there is no gain using that over jack.play). -- Len Ovens www.OvenWerks.net From p8rpp at aol.com Mon Jul 22 13:00:21 2013 From: p8rpp at aol.com (Peter P.) Date: Mon, 22 Jul 2013 15:00:21 +0200 Subject: [LAU] jack CD ripping: coding failed, err#8 Message-ID: <20130722130021.GA16506@aol.de> Hi, using the jack command line CD ripping program, the extraction from CD audio to wav works fine, but the encoder (pre-set to flac in the .jack3rc config file) seems to fail. Jack gives me "coding failed, err#8" for each CD track. Can't find a -verbose or -debug flag to make it tell me more. Any ideas and help is greatly appreciated! best, Peter From csanchezgs at gmail.com Mon Jul 22 13:48:18 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 22 Jul 2013 15:48:18 +0200 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51E55D94.2060705@rncbc.org> References: <51E55D94.2060705@rncbc.org> Message-ID: 2013/7/16 Rui Nuno Capela : > hi july ;) > > yet another batch of the Vee One Suite of old-school software instruments is > out: synthv1 [1] polyphonic synthesizer, samplv1 [2] polyphonic sampler and > drumkv1 [3] drum-kit sampler, this time presenting their brand new icons > while dropping the lamest old-schooler's out, in a master lesson taught by > Jarle Richard Akselsen, thanks. > > all still available in dual form: > > - a pure stand-alone JACK client with JACK-session, NSM (Non Session > management) and both JACK MIDI and ALSA MIDI input support; > - a LV2 instrument plugin. > > all free and open-source Linux Audio software, distributed under the terms > of the GNU General Public License (GPL) version 2 or later. > > here goes the new triumvirate bunch again: > > > [1] synthv1 - an old-school polyphonic synthesizer > > synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic > synthesizer with stereo fx. > > LV2 URI: http://synthv1.sourceforge.net/lv2 > > website: > http://synthv1.sourceforge.net > > downloads: > http://sourceforge.net/projects/synthv1/files > > - source tarball: > http://download.sourceforge.net/synthv1/synthv1-0.3.4.tar.gz > > - source package: > > http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.src.rpm > > - binary packages: > > http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.i586.rpm > > http://download.sourceforge.net/synthv1/synthv1-0.3.4-11.rncbc.suse123.x86_84.rpm > > > [2] samplv1 - an old-school polyphonic sampler > > samplv1 is an(other) old-school all-digital polyphonic sampler synthesizer > with stereo fx. > > LV2 URI: http://samplv1.sourceforge.net/lv2 > > website: > http://samplv1.sourceforge.net > > downloads: > http://sourceforge.net/projects/samplv1/files > > - source tarball: > http://download.sourceforge.net/samplv1/samplv1-0.3.4.tar.gz > > - source package: > > http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.src.rpm > > - binary packages: > > http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.i586.rpm > > http://download.sourceforge.net/samplv1/samplv1-0.3.4-11.rncbc.suse123.x86_84.rpm > > > [3] drumkv1 - an old-school drum-kit sampler > > drumkv1 is (yet) an(other) old-school all-digital drum-kit sampler > synthesizer with stereo fx. > > LV2 URI: http://drumkv1.sourceforge.net/lv2 > > website: > http://drumkv1.sourceforge.net > > downloads: > http://sourceforge.net/projects/drumkv1/files > > - source tarball: > http://download.sourceforge.net/drumkv1/drumkv1-0.3.4.tar.gz > > - source package: > > http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.src.rpm > > - binary packages: > > http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.i586.rpm > > http://download.sourceforge.net/drumkv1/drumkv1-0.3.4-7.rncbc.suse123.x86_84.rpm > > > see also: > http://www.rncbc.org/drupal/node/679 > > > enjoy && summer's ready! > -- > rncbc aka Rui Nuno Capela > rncbc at rncbc.org > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Hi, Rui. I've downloaded samplv1 in order to see if it can be executed headless and controlled via MIDI (after being configured with GUI). I couldn't compile, configure fails when checking Qt version on the machine I was logged. I see in README that Qt is mandatory, so I guess headless is not an option. Is that right? Thanks again for all your great work. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From kaspar.bumke at gmail.com Mon Jul 22 14:32:58 2013 From: kaspar.bumke at gmail.com (Kaspar Bumke) Date: Mon, 22 Jul 2013 15:32:58 +0100 Subject: [LAU] Headless jackdbus (no X11) In-Reply-To: References: Message-ID: Thanks for the tip. On 22 July 2013 01:55, Len Ovens wrote: > I have in the past used something like a jackstart script in a headless > situation before, which started dbus and exported it's info so jackdbus > would work. It is actually easier to use: > dbus-launch screen > > Then use screen as your CLI session manager. jack_control can then be used > to run jack and change settings on the fly (like latency?). Audio > applications can be run in other screen terminals to use jack... and if > you really want to go crazy you can throw pulse at the mix... though I am > really not sure what pulse supporting applications you hope to find that > are CLI (I tested it with paplay, but there is no gain using that over > jack.play). > > > -- > Len Ovens > www.OvenWerks.net > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy at autostatic.com Mon Jul 22 18:03:08 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Mon, 22 Jul 2013 20:03:08 +0200 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: References: <51E55D94.2060705@rncbc.org> Message-ID: <51ED73DC.6050207@autostatic.com> On 07/22/2013 03:48 PM, Carlos sanchiavedraz wrote: > I see in README that Qt is mandatory, so I guess headless is not an > option. Is that right? Hi Carlos, If you use jalv or modhost you can run samplv1 with no GUI. 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 rncbc at rncbc.org Mon Jul 22 21:38:28 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Mon, 22 Jul 2013 22:38:28 +0100 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51ED73DC.6050207@autostatic.com> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> Message-ID: <51EDA654.8010203@rncbc.org> On 07/22/2013 07:03 PM, Jeremy Jongepier wrote: > On 07/22/2013 03:48 PM, Carlos sanchiavedraz wrote: >> I see in README that Qt is mandatory, so I guess headless is not an >> option. Is that right? > > Hi Carlos, > > If you use jalv or modhost you can run samplv1 with no GUI. > yes. qt4 is mandatory for build and for runtime standalone samplv1_jack. however, as Jeremy already suggested, you can run the lv2 plugin without the gui with jalv, but again, you'll always need a (fake) X server anyhow so, the authoritarian answer is, no. sorry. any of the vee ones are not meant to run headless, and sure there are workarounds but i can't really patronize ways to do so, sorry :) cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From paul at linuxaudiosystems.com Mon Jul 22 21:50:39 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Mon, 22 Jul 2013 17:50:39 -0400 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51EDA654.8010203@rncbc.org> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> <51EDA654.8010203@rncbc.org> Message-ID: why would a fake X server be needed if no GUI is ever instantiated for the plugin? On Mon, Jul 22, 2013 at 5:38 PM, Rui Nuno Capela wrote: > On 07/22/2013 07:03 PM, Jeremy Jongepier wrote: > >> On 07/22/2013 03:48 PM, Carlos sanchiavedraz wrote: >> >>> I see in README that Qt is mandatory, so I guess headless is not an >>> option. Is that right? >>> >> >> Hi Carlos, >> >> If you use jalv or modhost you can run samplv1 with no GUI. >> >> > yes. qt4 is mandatory for build and for runtime standalone samplv1_jack. > > however, as Jeremy already suggested, you can run the lv2 plugin without > the gui with jalv, but again, you'll always need a (fake) X server anyhow > > so, the authoritarian answer is, no. sorry. any of the vee ones are not > meant to run headless, and sure there are workarounds but i can't really > patronize ways to do so, sorry :) > > cheers > > -- > rncbc aka Rui Nuno Capela > rncbc at rncbc.org > ______________________________**_________________ > Linux-audio-user mailing list > Linux-audio-user at lists.**linuxaudio.org > http://lists.linuxaudio.org/**listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rncbc at rncbc.org Mon Jul 22 23:35:48 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Tue, 23 Jul 2013 00:35:48 +0100 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> <51EDA654.8010203@rncbc.org> Message-ID: <51EDC1D4.7010800@rncbc.org> On 07/22/2013 10:50 PM, Paul Davis wrote: > why would a fake X server be needed if no GUI is ever instantiated for > the plugin? > 'coz the dsp code _and_ the gui code are currently residing on the same shared object (.so) and that, when linked to libqt4(-gui) as is, tends to do their nasty things on dload: often check if a x server is around, even though one hasn't come close to the plugin gui widget request yeah i could have made it into separate dsp and gui .so's which is something in scope of the lv2 spec. but i didn't, at least yet; there's this "DSSI-zing" call 'round here ie. implement complete isolation between dsp and gui process code, also getting rid of evil lv2 instance-access... but until then... :) cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From jeremy at autostatic.com Tue Jul 23 06:35:22 2013 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 23 Jul 2013 08:35:22 +0200 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51EDC1D4.7010800@rncbc.org> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> <51EDA654.8010203@rncbc.org> <51EDC1D4.7010800@rncbc.org> Message-ID: <51EE242A.8090506@autostatic.com> On 07/23/2013 01:35 AM, Rui Nuno Capela wrote: > 'coz the dsp code _and_ the gui code are currently residing on the same > shared object (.so) and that, when linked to libqt4(-gui) as is, tends > to do their nasty things on dload: often check if a x server is around, > even though one hasn't come close to the plugin gui widget request Does this also apply to drumkv1? Because I can rum drumkv1 without X when using jalv. 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 Tue Jul 23 08:27:37 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 23 Jul 2013 10:27:37 +0200 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51ED73DC.6050207@autostatic.com> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> Message-ID: 2013/7/22 Jeremy Jongepier : > On 07/22/2013 03:48 PM, Carlos sanchiavedraz wrote: >> I see in README that Qt is mandatory, so I guess headless is not an >> option. Is that right? > > Hi Carlos, > > If you use jalv or modhost you can run samplv1 with no GUI. > > Jeremy > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Ok, samplv1 as LV2 instrument with jalv. But I was curious if it could be also just standalone and headless. Thanks Jeremy. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From csanchezgs at gmail.com Tue Jul 23 09:00:58 2013 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 23 Jul 2013 11:00:58 +0200 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51EDA654.8010203@rncbc.org> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> <51EDA654.8010203@rncbc.org> Message-ID: 2013/7/22 Rui Nuno Capela : > On 07/22/2013 07:03 PM, Jeremy Jongepier wrote: >> >> On 07/22/2013 03:48 PM, Carlos sanchiavedraz wrote: >>> >>> I see in README that Qt is mandatory, so I guess headless is not an >>> option. Is that right? >> >> >> Hi Carlos, >> >> If you use jalv or modhost you can run samplv1 with no GUI. >> > > yes. qt4 is mandatory for build and for runtime standalone samplv1_jack. > > however, as Jeremy already suggested, you can run the lv2 plugin without the > gui with jalv, but again, you'll always need a (fake) X server anyhow > > so, the authoritarian answer is, no. sorry. any of the vee ones are not > meant to run headless, and sure there are workarounds but i can't really > patronize ways to do so, sorry :) > > cheers > > -- > rncbc aka Rui Nuno Capela > rncbc at rncbc.org > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user As LV2 is OK for me. Thanks Rui. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es From rncbc at rncbc.org Tue Jul 23 14:58:44 2013 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Tue, 23 Jul 2013 15:58:44 +0100 Subject: [LAU] [ANN] Vee One Suite 0.3.4 - Brand new icon ready In-Reply-To: <51EE9060.8060504@autostatic.com> References: <51E55D94.2060705@rncbc.org> <51ED73DC.6050207@autostatic.com> <51EDA654.8010203@rncbc.org> <51EDC1D4.7010800@rncbc.org> <51EE242A.8090506@autostatic.com> <51EE8CDB.5010103@rncbc.org> <51EE9060.8060504@autostatic.com> Message-ID: <51EE9A24.4020409@rncbc.org> On 07/23/2013 03:17 PM, Jeremy Jongepier wrote: > On 07/23/2013 04:02 PM, Rui Nuno Capela wrote: >> On 07/23/2013 07:35 AM, Jeremy Jongepier wrote: >>> On 07/23/2013 01:35 AM, Rui Nuno Capela wrote: >>>> 'coz the dsp code _and_ the gui code are currently residing on the same >>>> shared object (.so) and that, when linked to libqt4(-gui) as is, tends >>>> to do their nasty things on dload: often check if a x server is around, >>>> even though one hasn't come close to the plugin gui widget request >>> >>> Does this also apply to drumkv1? Because I can rum drumkv1 without X >>> when using jalv. >>> >> >> maybe my mistake and applies only to the standalone version, not to the >> lv2 plugin in general--you certainly need xlib and qt4-gui for build. >> >> you don't necessarily have to have an effectively running Xserverbut you >> certainly need Xlib et al. for the lv2 runtime. ntl. >> >> hth. > > Hi Rui, > > You only sent this mail to me or was that intentional? > ouch. thanks for the heads-up. i'm sending this from a sunbed by the sea shore, and some mobile clients are somewhat hard to master between dives ;) cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From fons at linuxaudio.org Wed Jul 24 13:18:35 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Wed, 24 Jul 2013 13:18:35 +0000 Subject: [LAU] TEST-IGNORE Message-ID: <20130724131835.GA28544@linuxaudio.org> TEST MESSAGE, PLEASE IGNORE -- 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 julien at mail.upb.de Wed Jul 24 16:36:32 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 24 Jul 2013 18:36:32 +0200 (CEST) Subject: [LAU] Trouble with mod-host Message-ID: Hello everyone! I just found, that there were quite a few updates for mod-host in the git repository. But after having built the software, I discovered a few problems. When starting mod-host I always get this message: error: PROTOCOL_MAX_COMMANDS reached (reconfigure it) In mod-host's interactive shell (mod-host -i), I can get help, but when typing quite, I get the message: not found. The command can even be completed, so a typo is unlikely. :-) Any ideas or fixes for this? Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From brunogola at gmail.com Wed Jul 24 16:49:21 2013 From: brunogola at gmail.com (Bruno Gola) Date: Wed, 24 Jul 2013 13:49:21 -0300 Subject: [LAU] Trouble with mod-host In-Reply-To: References: Message-ID: Hi Julien, Are you sure you have the latest version from github? I believe the PROTOCOL_MAX_COMMANDS issue was fixed a few months ago. About the "not found" problem, my guess is that it is related to the PROTOCOL_MAX_COMMANDS bug. Can you please try to do a git pull and try again? I just tried with the latest version and it works fine for me. Cheers, On Wed, Jul 24, 2013 at 1:36 PM, Julien Claassen wrote: > Hello everyone! > I just found, that there were quite a few updates for mod-host in the git > repository. But after having built the software, I discovered a few > problems. > When starting mod-host I always get this message: > error: PROTOCOL_MAX_COMMANDS reached (reconfigure it) > In mod-host's interactive shell (mod-host -i), I can get help, but when > typing quite, I get the message: > not found. > The command can even be completed, so a typo is unlikely. :-) > Any ideas or fixes for this? > Warm regards > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -- Bruno Gola http://bgo.la/ | +55 11 9-5552-3599 From julien at mail.upb.de Wed Jul 24 16:53:53 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 24 Jul 2013 18:53:53 +0200 (CEST) Subject: [LAU] Trouble with mod-host In-Reply-To: References: Message-ID: Hello Bruno! I had the latest version, but somehow the build process got tangled. A few minutes fidling around and I cloned again and everything worked fine. So some issue on my side. Thanks for putting my mind at ease. Now I can work with it again and have fun with it. :-) Best wishes Julien ---------------------------------------- http://juliencoder.de/nama/music.html From ricardo.crudo at gmail.com Thu Jul 25 14:26:43 2013 From: ricardo.crudo at gmail.com (Ricardo Crudo) Date: Thu, 25 Jul 2013 11:26:43 -0300 Subject: [LAU] Trouble with mod-host In-Reply-To: References: Message-ID: Hey Julien! Maybe the "not found" problem was extra spaces at end of line, I fix it. And, taking advantage, two new features: - commands now support quotation marks - load / save commands added (you can save and load a preset in 'commands history' format) Regards, Ricardo Crudo. 2013/7/24 Julien Claassen > Hello Bruno! > I had the latest version, but somehow the build process got tangled. A > few minutes fidling around and I cloned again and everything worked fine. > So some issue on my side. Thanks for putting my mind at ease. Now I can > work with it again and have fun with it. :-) > Best wishes > > Julien > > ------------------------------**---------- > http://juliencoder.de/nama/**music.html > ______________________________**_________________ > 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 julien at mail.upb.de Thu Jul 25 14:58:09 2013 From: julien at mail.upb.de (Julien Claassen) Date: Thu, 25 Jul 2013 16:58:09 +0200 (CEST) Subject: [LAU] Trouble with mod-host In-Reply-To: References: Message-ID: Hello Ricardo! The save/load feature is a very nice and helpful one. It can take a great deal of time to set up one of the larger plugins completely and with that, it won't be just for one session. Good feature! Good work! Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From cracauer at cons.org Thu Jul 25 18:21:50 2013 From: cracauer at cons.org (Martin Cracauer) Date: Thu, 25 Jul 2013 14:21:50 -0400 Subject: [LAU] Gain and clipping wav -> lame Message-ID: <20130725182150.GA97830@cons.org> Does anybody know why wav files that do not have clipping would clip when encoding them with lame? I have a bunch of self-made clips that for various reasons don't have enough gain. I wrote a wav->wav conversion that determines maximum gain (as in highest or lowest absolute value) and uses sox' volume argument to boost. Inspecting it with same gain tools and listening to it shows the expected gain raise, but not clipping. Then, when encoding in lame it gets clipped. Lame warns about it and the values have been boosted to the max. When doing the boost, with the same factor, directly in lame (feeding in the unboosted wav) I get the same problem. Very roughly a clip with 3% headroom ends up being too high by 16%, or about a 20% boost. What's going on? I assume there is something about mp3 encoding I don't quite understand. Is there anything more efficient that I can do about that doing one lame run to get an appropriate boost value and then doing another encoding pass? Here are the stats of a boosted wav file and the resulting lame: t02c02.tmp.wav_maxgain_tmp.wav: Samples read: 36900960 Length (seconds): 384.385000 Scaled by: 2147483647.0 Maximum amplitude: 0.969208 Minimum amplitude: -0.970673 Midline amplitude: -0.000732 Mean norm: 0.213396 Mean amplitude: 0.000000 RMS amplitude: 0.274775 Maximum delta: 1.675323 Minimum delta: 0.000000 Mean delta: 0.193639 RMS delta: 0.245325 Rough frequency: 6820 Volume adjustment: 1.030 mp3: Samples read: 36907776 Length (seconds): 384.456000 Scaled by: 2147483647.0 Maximum amplitude: 1.000000 Minimum amplitude: -1.000000 Midline amplitude: -0.000000 Mean norm: 0.203464 Mean amplitude: 0.000010 RMS amplitude: 0.262007 Maximum delta: 1.636044 Minimum delta: 0.000000 Mean delta: 0.185782 RMS delta: 0.235408 Rough frequency: 6863 Volume adjustment: 1.000 /opt/good-sox/bin/sox WARN sox: `-' input clipped 725 samples /opt/good-sox/bin/sox WARN sox: `/tmp/cracauer/l.wav' output clipped 360 samples; decrease volume? Encoding output: lame -h -b 160 --replaygain-accurate --clipdetect t02c02.tmp.wav_maxgain_tmp.wav t02c02.tmp.wav_maxgain_tmp.mp3_tmp '&&' mv t02c02.tmp.wav_maxgain_tmp.mp3_tmp t02c02.tmp.wav_maxgain_tmp.mp3 + sh LAME version 3.96.1 (http://lame.sourceforge.net/) Using polyphase lowpass filter, transition band: 18000 Hz - 18581 Hz Encoding t02c02.tmp.wav_maxgain_tmp.wav to t02c02.tmp.wav_maxgain_tmp.mp3_tmp Encoding as 48 kHz 160 kbps j-stereo MPEG-1 Layer III (9.6x) qval=2 Frame | CPU time/estim | REAL time/estim | play/CPU | ETA 16015/16018 (100%)| 0:22/ 0:22| 0:22/ 0:22| 17.121x| 0:00 average: 160.0 kbps LR: 2551 (15.93%) MS: 13467 (84.07%) Writing LAME Tag...done ReplayGain: -9.1dB WARNING: clipping occurs at the current gain. Set your decoder to decrease the gain by at least 1.3dB or encode again using --scale (For a suggestion on the optimal value of encode with --scale 1 first) -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer http://www.cons.org/cracauer/ From fons at linuxaudio.org Thu Jul 25 19:40:53 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 25 Jul 2013 19:40:53 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725182150.GA97830@cons.org> References: <20130725182150.GA97830@cons.org> Message-ID: <20130725194053.GA20598@linuxaudio.org> On Thu, Jul 25, 2013 at 02:21:50PM -0400, Martin Cracauer wrote: > Does anybody know why wav files that do not have clipping would clip > when encoding them with lame? Because the peak sample value says nothing about the real level. Take a square wave with peak values +/-1. It is the sum of a number of sine waves, with frequencies 1,3,5,7,... times the frequency of the square wave. The first one has a peak value of about +/-1.27, that is +2 dB. So any encoding that looks at the spectrum (and mp3 does) will see a level that is +2 dB. If you really want to normalise on the peak level, use a lower one. If you want all your samples to have the same loudness, use RMS instead of peak, or a real loudness measurement such as provided by ebumeter. 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 cracauer at cons.org Thu Jul 25 20:14:20 2013 From: cracauer at cons.org (Martin Cracauer) Date: Thu, 25 Jul 2013 16:14:20 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725194053.GA20598@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> Message-ID: <20130725201420.GA58212@cons.org> Fons Adriaensen wrote on Thu, Jul 25, 2013 at 07:40:53PM +0000: > On Thu, Jul 25, 2013 at 02:21:50PM -0400, Martin Cracauer wrote: > > > Does anybody know why wav files that do not have clipping would clip > > when encoding them with lame? > > Because the peak sample value says nothing about the real > level. Take a square wave with peak values +/-1. It is the > sum of a number of sine waves, with frequencies 1,3,5,7,... > times the frequency of the square wave. The first one has > a peak value of about +/-1.27, that is +2 dB. So any encoding > that looks at the spectrum (and mp3 does) will see a level > that is +2 dB. Ah. Makes sense. Thanks so much! Is there a rule of thumb how many db less I should give music to avoid this? What would be the value for pink noise starting at 40 Hz? > If you really want to normalise on the peak level, use a > lower one. If you want all your samples to have the same > loudness, use RMS instead of peak, or a real loudness > measurement such as provided by ebumeter. Is there a way to hook up ebumeter to just an audio file or a stream not associated with real time? It seems to come in a jack package only. Thanks again Martin -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer http://www.cons.org/cracauer/ From marc at hacklava.net Thu Jul 25 20:45:48 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Thu, 25 Jul 2013 16:45:48 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725201420.GA58212@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> Message-ID: <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> Martin Cracauer a ?crit : > Is there a way to hook up ebumeter to just an audio file or a stream > not associated with real time? It seems to come in a jack package > only. > > Thanks again > Martin http://r128gain.sourceforge.net/ -- Marc From willgodfrey at musically.me.uk Thu Jul 25 21:09:07 2013 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Thu, 25 Jul 2013 22:09:07 +0100 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725201420.GA58212@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> Message-ID: <20130725220907.7a01e4a5@debian> On Thu, 25 Jul 2013 16:14:20 -0400 Martin Cracauer wrote: > Fons Adriaensen wrote on Thu, Jul 25, 2013 at 07:40:53PM +0000: > > On Thu, Jul 25, 2013 at 02:21:50PM -0400, Martin Cracauer wrote: > > > > > Does anybody know why wav files that do not have clipping would clip > > > when encoding them with lame? > > > > Because the peak sample value says nothing about the real > > level. Take a square wave with peak values +/-1. It is the > > sum of a number of sine waves, with frequencies 1,3,5,7,... > > times the frequency of the square wave. The first one has > > a peak value of about +/-1.27, that is +2 dB. So any encoding > > that looks at the spectrum (and mp3 does) will see a level > > that is +2 dB. > > Ah. Makes sense. Thanks so much! > > Is there a rule of thumb how many db less I should give music to avoid > this? What would be the value for pink noise starting at 40 Hz? I always finalise a wav file in Audacity with the peak level at -1dB. Seems to work for me :) -- 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 fons at linuxaudio.org Thu Jul 25 21:44:16 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 25 Jul 2013 21:44:16 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725201420.GA58212@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> Message-ID: <20130725214416.GB20598@linuxaudio.org> On Thu, Jul 25, 2013 at 04:14:20PM -0400, Martin Cracauer wrote: > Is there a rule of thumb how many db less I should give music > to avoid this? Depends entirely on what you are trying to do, what sort of signals/music you have, how the clips are going to be used, and where you stand in the 'loudness wars'. 'Acoustic' music has a peak/RMS ratio of around 20 dB. The EBU standard means to aim for an average RMS level of -23 dB. Most popular music productions try to have a much higher average level, and hence a lower peak/RMS ratio, something like 10 dB, or 6 dB for the more agressive ones, and as low as 3 dB if the target audience is the braindead. OTOH, if your clips are e.g. unprocessed drum samples, you could have a peak/RMS ratio that is much higher than 20 dB. So it depends, use your judgement. > What would be the value for pink noise starting at 40 Hz? Impossible to say. Gaussian noise (analog noise is Gaussian) has in theory an infinite peak/RMS ratio. In practice it will be limited of course. To get an idea of this, you could measure the pink noise provided by e.g. japa with jkmeter (which shows RMS and peak). The RMS value is -20 dB. If you wait long enough, the peak memory can take any value, there is no limit. In practice, if one or few samples per second get clipped you won't notice. > Is there a way to hook up ebumeter to just an audio file or a stream > not associated with real time? It seems to come in a jack package only. If you have ebumeter installed, try ebur128 --full --lufs filename.wav The ebur128 program is included in the source distribution of ebumeter, I don't know if it's available in binary packages. 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 davidtweed2003 at yahoo.com Thu Jul 25 23:59:06 2013 From: davidtweed2003 at yahoo.com (tweed) Date: Thu, 25 Jul 2013 16:59:06 -0700 (PDT) Subject: [LAU] Trouble with mod-host In-Reply-To: References: Message-ID: <1374796746.47581.YahooMailNeo@web161701.mail.bf1.yahoo.com> Hello , trying mod-host for the 1st time.? do i just get param names from .ttl file?? or is there a way to retreive them from within mod-host? interactively? thanks so much for your work on this. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 26 04:29:03 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 26 Jul 2013 06:29:03 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725214416.GB20598@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725214416.GB20598@linuxaudio.org> Message-ID: <1374812943.773.21.camel@archlinux> On Thu, 2013-07-25 at 21:44 +0000, Fons Adriaensen wrote: > as low as 3 dB if the target audience ^ and the engineer and the producer are the braindead. No engineer or producer is forced to do bad recordings. The target group is naive and only sometimes braindead. Engineers and producers permanently doing this are the braindead. From kvutter at frii.com Fri Jul 26 04:40:16 2013 From: kvutter at frii.com (Kevin Utter) Date: Thu, 25 Jul 2013 22:40:16 -0600 Subject: [LAU] MIDI Port names in Jack Message-ID: Hi all! I'm wondering why so many of my MIDI ports in Jack are named so similarly, and if they can be changed? I don't remember this being a problem on my previous Linux system, but there are a few differences. I'm avoiding GUIs and need to use CLI tools, so no Qjackctl suggestions, please. I'm using an M-Audio MIDI sport Uno USB midi interface, and the virMIDI device for internal patching, as well as MIDISH. In jack, all of those, along with the main system ports like through are all listed as system midi playback or capture and a number. I've tried both raw and seq jack MIDI modes. I can see which is which with Jack_lsp with the -A or -t options set. But I was hoping to use something like Jack Plumbing for reconnecting, and since the real client names don't show up, I can't get jack plumbing to see them. Since the port numbers aren't always the same, I can't depend on the suffix digits to identify the right port. Then, using MIDISH, where the ports come and go when it starts and stops, its port numbers seem to increment each time as well. Should I perhaps be using A2Jmidid instead of the jack MIDI modes? Is there something else I'm missing? Or is this a difference between Jack 1 and Jack 2? Or differences from Ubuntu 9-10 and 12-10? Sorry to be so wordy, but wanted to be clear. Thanks for any help. Kevin From espiritocz at gmail.com Fri Jul 26 05:19:52 2013 From: espiritocz at gmail.com (Milan Lazecky) Date: Fri, 26 Jul 2013 13:19:52 +0800 Subject: [LAU] linearly increasing velocity Message-ID: Hi, I will play one song to my sister's wedding in 14 days :) the idea is that it starts slow and then every measure goes little faster (+XX bpm) than previous one. Then it remains few measures in the final (fast) speed. Does anybody have experience (with any linux utility) for this? Thank you. Milan -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 26 05:25:35 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 26 Jul 2013 07:25:35 +0200 Subject: [LAU] linearly increasing tempo (velocity ;) In-Reply-To: References: Message-ID: <1374816335.736.3.camel@archlinux> On Fri, 2013-07-26 at 13:19 +0800, Milan Lazecky wrote: > the idea is that it starts slow and then every measure goes little > faster (+XX bpm) than previous one. Then it remains few measures in > the final (fast) speed. > > > Does anybody have experience (with any linux utility) for this? This can be done with Qtractor, Ardour and for sure with others too. From julien at mail.upb.de Fri Jul 26 06:19:24 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 26 Jul 2013 08:19:24 +0200 (CEST) Subject: [LAU] Trouble with mod-host In-Reply-To: <1374796746.47581.YahooMailNeo@web161701.mail.bf1.yahoo.com> References: <1374796746.47581.YahooMailNeo@web161701.mail.bf1.yahoo.com> Message-ID: Hello David! You can get the names from within mod-host. You can type something like: add http://nickbailey.co.nr/triceratops 0 param_set 0 Problem is, you might get quite a few results. Another external option is to use lv2info in another terminal. Nicer than the ttl file and more helpful than the output in mod-host . I hope this helps a little. Kindly yours Julien ---------------------------------------- http://juliencoder.de/nama/music.html From julien at mail.upb.de Fri Jul 26 06:29:06 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 26 Jul 2013 08:29:06 +0200 (CEST) Subject: [LAU] MIDI Port names in Jack In-Reply-To: References: Message-ID: Hello Kevin! I'm not sure about your portnames, but with -Xseq (in JACK 1), my MIDI port names look like this: alsa_pcm:Emu10k1-WaveTable/midi_playback_4 This is very long, but for automated purposes this is OK. I could cut© and paste it into any script. But there's jack_alias, this could rename ports for you. Since your ports do change names or numbers, this might not be the best solution neither. As far as I know, jack_alias can rename a whole client, not a single port. With j2amidi_bridge you could patch a JACK port thought to ALSA. You'd still have to connect a softsynth port to it. But soft synth ports should have the usual names. Other hardware ports can be connected directly through ALSA sequencer. I can send you my example .midishrc off-list. There's more than just a device setup, there are some convenience functions to connect new ports on the fly. I hope, this did answer your question, at least partly. Kind regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From davidtweed2003 at yahoo.com Fri Jul 26 09:32:44 2013 From: davidtweed2003 at yahoo.com (tweed) Date: Fri, 26 Jul 2013 02:32:44 -0700 (PDT) Subject: [LAU] Trouble with mod-host In-Reply-To: References: <1374796746.47581.YahooMailNeo@web161701.mail.bf1.yahoo.com> Message-ID: <1374831164.4839.YahooMailNeo@web161705.mail.bf1.yahoo.com> Thanks so much for your help Julien, I didn't know about mod-host so I'm glad I saw your post.? cool how lv2info shows min, max, default. Thanks again. >________________________________ > From: Julien Claassen >To: tweed >Cc: linux-audio-user xc >Sent: Friday, July 26, 2013 2:19 AM >Subject: Re: [LAU] Trouble with mod-host > > >Hello David! >? You can get the names from within mod-host. You can type something like: >add http://nickbailey.co.nr/triceratops 0 >param_set 0 >? Problem is, you might get quite a few results. Another external option is to >use lv2info in another terminal. Nicer than the ttl file and more helpful than >the output in mod-host . >? I hope this helps a little. >? Kindly yours >? ? ? ? ? Julien > >---------------------------------------- >http://juliencoder.de/nama/music.html > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at alice-dsl.net Fri Jul 26 10:33:16 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 26 Jul 2013 12:33:16 +0200 Subject: [LAU] Session managers Message-ID: <1374834796.799.36.camel@archlinux> Hi, do session managers take care about the configurations of audio apps, e.g. do they save and restore ~/.config/rncbc.org/ too? Can they launch apps by a terminal emulation as I can do it using a script? E.g. xfce4-terminal --maximize -T "? jackd" -e "jackd --sync -dalsa -r$sample_rate -p$frames_period" xfce4-terminal --maximize -T "? qtractor" -e qtractor\ $song_path/qtr/$song_name-$song_version_qtr.qtr Somebody on the Arch general mailing list mentioned to use session managers, when I ask "How to safe configs to another path than ~" - https://mailman.archlinux.org/pipermail/arch-general/2013-July/thread.html#start I'm looking for a solution, not only for audio sessions, but for audio sessions it's the most important. Regards, Ralf From brummer- at web.de Fri Jul 26 11:32:41 2013 From: brummer- at web.de (hermann meyer) Date: Fri, 26 Jul 2013 13:32:41 +0200 Subject: [LAU] Session managers In-Reply-To: <1374834796.799.36.camel@archlinux> References: <1374834796.799.36.camel@archlinux> Message-ID: <51F25E59.9090505@web.de> Am 26.07.2013 12:33, schrieb Ralf Mardorf: > Hi, > > do session managers take care about the configurations of audio apps, > e.g. do they save and restore ~/.config/rncbc.org/ too? > > Can they launch apps by a terminal emulation as I can do it using a > script? > > E.g. > xfce4-terminal --maximize -T "? jackd" -e "jackd --sync -dalsa -r$sample_rate -p$frames_period" > xfce4-terminal --maximize -T "? qtractor" -e qtractor\ $song_path/qtr/$song_name-$song_version_qtr.qtr > > Somebody on the Arch general mailing list mentioned to use session > managers, when I ask "How to safe configs to another path than ~" - > https://mailman.archlinux.org/pipermail/arch-general/2013-July/thread.html#start > > I'm looking for a solution, not only for audio sessions, but for audio > sessions it's the most important. > > Regards, > Ralf > > Applications can only save there config files by themselves, a sessionmanager cant do that ever. All, a sessionmanager can do, is to ask a application to save it now. If implemented in the application, it will do so. As well the sessionmanager could give a path to save to, but the same rule apply as before. From ralf.mardorf at alice-dsl.net Fri Jul 26 11:52:16 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Fri, 26 Jul 2013 13:52:16 +0200 Subject: [LAU] Session managers In-Reply-To: <51F25E59.9090505@web.de> References: <1374834796.799.36.camel@archlinux> <51F25E59.9090505@web.de> Message-ID: <1374839536.799.48.camel@archlinux> On Fri, 2013-07-26 at 13:32 +0200, hermann meyer wrote: > Applications can only save there config files by themselves, a > sessionmanager cant do that ever. Thank you Hermann, that was the thought I had. With good luck the HOME variable might do the job. HOME=/path/to/session/directory application but the thread on the Arch Linux mailing list already showed, that the only way seems to be to write a function call that will replace the getpwuid() calls by LD_PRELOAD. If a session manager would do this, I suspect issues for applications that always should use the original home directory, it at least would cause a race condition. Btw. I'm not willing to relearn C at the moment and so I'm unable to use LD_PRELOAD, so I still need to search the config and than copy and restore it, when switching between sessions. Regards, Ralf From paul at linuxaudiosystems.com Fri Jul 26 12:07:51 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 26 Jul 2013 08:07:51 -0400 Subject: [LAU] MIDI Port names in Jack In-Reply-To: References: Message-ID: On Fri, Jul 26, 2013 at 12:40 AM, Kevin Utter wrote: > > stops, its port numbers seem to increment each time as well. Should I > perhaps be using A2Jmidid instead of the jack MIDI modes? yes, absolutely. the only reason that -X raw and -X seq have not been removed is that the code to replace them isn't ready. always use a2jmidid (with -e to pick up external h/w ports). -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Fri Jul 26 13:04:09 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 26 Jul 2013 13:04:09 +0000 Subject: [LAU] MIDI Port names in Jack In-Reply-To: References: Message-ID: <20130726130409.GA29498@linuxaudio.org> On Fri, Jul 26, 2013 at 08:07:51AM -0400, Paul Davis wrote: > yes, absolutely. the only reason that -X raw and -X seq have not been > removed is that the code to replace them isn't ready. always use a2jmidid > (with -e to pick up external h/w ports). This made me have a look at the sources. Up to ten minutes ago I was believing that a2jmidid used timestamps etc. to reduce jitter, and that this made it preferable over Jack's -X. But apparently it doesn't, all events in the same Jack period are just bunched together. There's even a comment in the source code: /* -FIX- this should have the frame time of the event, instead of '0': */ So what is the advantage ? 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 hollunder at lavabit.com Fri Jul 26 13:26:21 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Fri, 26 Jul 2013 15:26:21 +0200 Subject: [LAU] Looking for advice on a field recording device Message-ID: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Hi there, I finally want to buy a small field recording device, a zoom H* or similar, and I hope that some of you have experience with some of those devices and can give me some advice. I want to use this device to record interesting sounds wherever I am, so it should be relatively small and and fast to start recording. I also intend to record lots of quiet or relatively far away sources, I want to record bird songs rather than guitars. As far as I've seen those devices use SSDs or similar storage media, so I assume they don't make any noise on their own, right? I guess the microphone should also have a low self-noise. Any recommendations? Regards, Philipp From paul at linuxaudiosystems.com Fri Jul 26 13:53:06 2013 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 26 Jul 2013 09:53:06 -0400 Subject: [LAU] MIDI Port names in Jack In-Reply-To: <20130726130409.GA29498@linuxaudio.org> References: <20130726130409.GA29498@linuxaudio.org> Message-ID: the last time i touched a2jmidi, it did precisely what you were looking for - 1 period of latency, zero jitter. however, i do not maintain a2jmidid (it is not my project) and i have no idea what has happened to it since I last did any work on it. On Fri, Jul 26, 2013 at 9:04 AM, Fons Adriaensen wrote: > On Fri, Jul 26, 2013 at 08:07:51AM -0400, Paul Davis wrote: > > > yes, absolutely. the only reason that -X raw and -X seq have not been > > removed is that the code to replace them isn't ready. always use a2jmidid > > (with -e to pick up external h/w ports). > > This made me have a look at the sources. > > Up to ten minutes ago I was believing that a2jmidid used timestamps etc. > to reduce jitter, and that this made it preferable over Jack's -X. > > But apparently it doesn't, all events in the same Jack period are > just bunched together. There's even a comment in the source code: > > /* -FIX- this should have the frame time of the event, instead of '0': */ > > So what is the advantage ? > > Ciao, > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Fri Jul 26 14:44:35 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 26 Jul 2013 14:44:35 +0000 Subject: [LAU] MIDI Port names in Jack In-Reply-To: References: <20130726130409.GA29498@linuxaudio.org> Message-ID: <20130726144435.GB29498@linuxaudio.org> On Fri, Jul 26, 2013 at 09:53:06AM -0400, Paul Davis wrote: > the last time i touched a2jmidi, it did precisely what you were looking for > - 1 period of latency, zero jitter. > > however, i do not maintain a2jmidid (it is not my project) and i have no > idea what has happened to it since I last did any work on it. The code in jack.c seems to do the right thing, the one in a2jmidi_bridge.c and j2amidi_bridge.c (which is where I looked first) certainly doesn'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 julien at mail.upb.de Fri Jul 26 15:04:08 2013 From: julien at mail.upb.de (Julien Claassen) Date: Fri, 26 Jul 2013 17:04:08 +0200 (CEST) Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: Hello Philipp! I have a Zoom h4n now and it is very good. It's not the tiniest of implements, but certainly easyenough to carry around. It takes a minute to start up, but then recording is instantaneous. It has two built-in microphones and you can change the angle between those two from 90 to 120 degrees. The self-noise is present, but OK. You can go to my website and check outmy field recordings to get your own impressions of its recording quality. A friend recently bought a Zoom H2n. It's much smaller, doesn't have as many fancy recording options. I don't know, how fast this can boot up and I don't know the difference in sound. I can ask myfriend, what he thinks. He knows the Zoom H4n in comparison, since it was his. :-) As to their own noise levels, you assume correctly. Since they write on SD cards, they are silent. Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From jamesmstone at gmail.com Fri Jul 26 15:24:19 2013 From: jamesmstone at gmail.com (James Stone) Date: Fri, 26 Jul 2013 16:24:19 +0100 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: On Jul 26, 2013 2:26 PM, "Philipp ?berbacher" wrote: > > Hi there, > > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. > > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. > > Any recommendations? > Not sure about recommendations.. I had a zoom h1 for recording band rehearsals. Was amazingly good sound quality and able to cope with very loud sound as well as very quiet. Don't remember any bad internal noise. Fast startip - almost instantaneous i think. However, it felt plasticky and after approx 9 months stopped booting following no obvious trauma. I got a refund and didn't replace it. However I wonder if I was unlucky. Half tempted to buy another one to see... J -------------- next part -------------- An HTML attachment was scrubbed... URL: From brendan.jones.it at gmail.com Fri Jul 26 15:38:10 2013 From: brendan.jones.it at gmail.com (Brendan Jones) Date: Fri, 26 Jul 2013 17:38:10 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <51F297E2.6000901@gmail.com> On 07/26/2013 05:04 PM, Julien Claassen wrote: > Hello Philipp! > I have a Zoom h4n now and it is very good. It's not the tiniest of > implements, but certainly easyenough to carry around. It takes a minute > to start up, but then recording is instantaneous. It has two built-in > microphones and you can change the angle between those two from 90 to > 120 degrees. The self-noise is present, but OK. You can go to my website > and check outmy field recordings to get your own impressions of its > recording quality. > A friend recently bought a Zoom H2n. It's much smaller, doesn't have > as many fancy recording options. I don't know, how fast this can boot up > and I don't know the difference in sound. I can ask myfriend, what he > thinks. He knows the Zoom H4n in comparison, since it was his. :-) > As to their own noise levels, you assume correctly. Since they write > on SD cards, they are silent. No direct recommendations here but I've also heard good reports from the Zoom. Toshiba is another option if you require stereo From jkroll at lavabit.com Fri Jul 26 15:55:07 2013 From: jkroll at lavabit.com (Johannes Kroll) Date: Fri, 26 Jul 2013 17:55:07 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130726175507.7f4531d4@sampi> On Fri, 26 Jul 2013 15:26:21 +0200 Philipp ?berbacher wrote: > Hi there, > > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. I tried the H2N some time ago. I was impressed by the sound quality, I liked the 4 built-in microphones which you can use in different configurations. But I was underwhelmed by what is probably the self-noise of the thing. Whenever I tried to record low-volume stuff, like someone talking quietly a meter away or so, or animal noises in some distance, and later digitally amplified that to a normal level, white noise became audible quickly. I thought these devices would have a better signal-to-noise ratio. I tried different settings (uncompressed, 16/24 bits, different sampling rates) with no luck. Maybe I overestimated what devices in that price range can do. In the end I returned it. Want to buy another one, but not sure which. From kevinc at cosgroves.us Fri Jul 26 16:54:02 2013 From: kevinc at cosgroves.us (Kevin Cosgrove) Date: Fri, 26 Jul 2013 09:54:02 -0700 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130726165402.63CF9BE05B@joseph.cosgroves.us> On 26 July 2013 at 15:26, Philipp ?berbacher wrote: > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. > > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. I like everything about my Zoom H2, except for one thing. The line in jack is not very durable, and mine is intermittent on one side of the stereo pair. It's got 4 mics, and I can select either pair, or both pairs for recording. I'm pretty good now at selecting a good sounding place in a room to record music. I've used it at family gatherings to record conversations, etc. I've had it for about 5 years now, and recorded 400-500 hours of material. The SD card makes it super easy to transfer to my computer. If I had a chance to start over, I'd buy a Zoom H4n for what I hope would be more rugged line in jacks. That would mean I'd need to forgo the 2nd pair of microphones. I would be willing to part with my H2. Good luck with your search.... P.S.: You could use a smart phone and a microphone for this. -- Kevin From nachoen79 at hotmail.com Fri Jul 26 16:54:24 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Fri, 26 Jul 2013 18:54:24 +0200 Subject: [LAU] live audio on netjack1 Message-ID: Hi everybody. I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidtweed2003 at yahoo.com Fri Jul 26 22:14:16 2013 From: davidtweed2003 at yahoo.com (tweed) Date: Fri, 26 Jul 2013 15:14:16 -0700 (PDT) Subject: [LAU] live audio on netjack1 In-Reply-To: References: Message-ID: <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> Hi Nacho, which direction are you trying to send live audio.?? master to slave or slave to master??? if slave to master use alsa_in on slave machine to have that audio interface as a jack client.? hope this helps. ? www.the-temp-agency.com/lollipopfactory ? >________________________________ > From: Nacho - >To: "linux-audio-user at lists.linuxaudio.org" >Sent: Friday, July 26, 2013 12:54 PM >Subject: [LAU] live audio on netjack1 > > > > >Hi everybody. > >I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? > >Thanks! > >_______________________________________________ >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 cracauer at cons.org Fri Jul 26 22:35:02 2013 From: cracauer at cons.org (Martin Cracauer) Date: Fri, 26 Jul 2013 18:35:02 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> Message-ID: <20130726223502.GA73801@cons.org> Thanks again. Things are looking much better now. I embraced the concept of not being able to predict this. Turns out I have old screwed up things that really benefitted from revisiting them and cleaning them up. Marc Lavall??e wrote on Thu, Jul 25, 2013 at 04:45:48PM -0400: > Martin Cracauer a ??crit : > > Is there a way to hook up ebumeter to just an audio file or a stream > > not associated with real time? It seems to come in a jack package > > only. > > > > Thanks again > > Martin > > http://r128gain.sourceforge.net/ That works. Comes in the Debian ebumeter package, BTW. But how do I translate the output to a required db or ratio adjustment? I have a file in front of me that was maxed out amplitude wise by somebody else. According to lame I needed --scale 0.64 to get it not to clip in lame (should be 3.9 db). Which, BTW was not the value it first estimated that I would need. ebur128 --lufs: Integrated loudness: -4.9 LUFS Loudness range: 5.5 LU Integrated threshold: -13.0 LUFS Range threshold: -25.0 LUFS Range min: -8.6 LUFS Range max: -3.1 LUFS Momentary max: -1.2 LUFS Short term max: -2.4 LUFS I don't see that any of the value correspond with what lame needed to not clip over a collection of different loudness clips. (means: some clips that needed less --scale have higher numbers here and others have lower) I assume the ebumeter output is more for making things sound even (between different pieces) and not directly a tool to max out anything, is that right? Martin -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer http://www.cons.org/cracauer/ From looplog at gmail.com Sat Jul 27 01:37:53 2013 From: looplog at gmail.com (michael noble) Date: Sat, 27 Jul 2013 10:37:53 +0900 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: If mic self-noise is truly a concern for you, then look beyond Zoom. This page provides a good comparison: http://www.avisoft.com/recordertests.htm I have the Sony PCM-M10 and a Tascam DR-680 for different purposes, and both are solid units for there price. Keep in mind that most portable recorders will not give pleasing results for "far away sources" unless you mean general ambient recording. For distant sources you would need a specialized microphone. For most versatility, I personally would recommend choosing a recorder with XLR inputs such that you can buy additional mics in the future. The Olympus LS-100 seems to be a favorite among recordists on a budget for this reason. If you want the advice of people far more experienced with these recorders and techniques than myself, I'd also suggest signing up the for the nature recordists mailing list. It's a very useful resource in general. http://tech.groups.yahoo.com/group/naturerecordists/ On Fri, Jul 26, 2013 at 10:26 PM, Philipp ?berbacher wrote: > Hi there, > > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. > > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. > > Any recommendations? > > Regards, > Philipp > > _______________________________________________ > 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 hollunder at lavabit.com Sat Jul 27 07:10:20 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sat, 27 Jul 2013 09:10:20 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130726223502.GA73801@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> Message-ID: <20130727091020.7aa32573@eeyore.mozart.uni-klu.ac.at> On Fri, 26 Jul 2013 18:35:02 -0400 Martin Cracauer wrote: > Thanks again. Things are looking much better now. I embraced the > concept of not being able to predict this. Turns out I have old > screwed up things that really benefitted from revisiting them and > cleaning them up. > > Marc Lavall??e wrote on Thu, Jul 25, 2013 at 04:45:48PM -0400: > > Martin Cracauer a ??crit : > > > Is there a way to hook up ebumeter to just an audio file or a > > > stream not associated with real time? It seems to come in a jack > > > package only. > > > > > > Thanks again > > > Martin > > > > http://r128gain.sourceforge.net/ > > That works. Comes in the Debian ebumeter package, BTW. > > But how do I translate the output to a required db or ratio > adjustment? > > I have a file in front of me that was maxed out amplitude wise by > somebody else. According to lame I needed --scale 0.64 to get it not > to clip in lame (should be 3.9 db). Which, BTW was not the value it > first estimated that I would need. > > ebur128 --lufs: > Integrated loudness: -4.9 LUFS > Loudness range: 5.5 LU > Integrated threshold: -13.0 LUFS > Range threshold: -25.0 LUFS > Range min: -8.6 LUFS > Range max: -3.1 LUFS > Momentary max: -1.2 LUFS > Short term max: -2.4 LUFS > > I don't see that any of the value correspond with what lame needed to > not clip over a collection of different loudness clips. (means: some > clips that needed less --scale have higher numbers here and others > have lower) > > I assume the ebumeter output is more for making things sound even > (between different pieces) and not directly a tool to max out > anything, is that right? > > Martin A good place to find out about this is the talk by Fons and J?rn given at the LAC 2011, here's the link to the program page where you can find the paper, slides and the video recording: http://lac.linuxaudio.org/2011/?page=program&mode=list Regards, Philipp From hollunder at lavabit.com Sat Jul 27 07:37:42 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sat, 27 Jul 2013 09:37:42 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130727093742.189879a5@eeyore.mozart.uni-klu.ac.at> On Sat, 27 Jul 2013 10:37:53 +0900 michael noble wrote: > If mic self-noise is truly a concern for you, then look beyond Zoom. > This page provides a good comparison: > > http://www.avisoft.com/recordertests.htm > > I have the Sony PCM-M10 and a Tascam DR-680 for different purposes, > and both are solid units for there price. Keep in mind that most > portable recorders will not give pleasing results for "far away > sources" unless you mean general ambient recording. For distant > sources you would need a specialized microphone. For most > versatility, I personally would recommend choosing a recorder with > XLR inputs such that you can buy additional mics in the future. The > Olympus LS-100 seems to be a favorite among recordists on a budget > for this reason. > > If you want the advice of people far more experienced with these > recorders and techniques than myself, I'd also suggest signing up the > for the nature recordists mailing list. It's a very useful resource > in general. > > http://tech.groups.yahoo.com/group/naturerecordists/ Thanks everyone, I spent quite some time sifting through reviews on the thomann homepage, but I simply can't trust those, your advice is much appreciated. Julien, I especially enjoyed the 'Toilet visit' :) I think the noise in the recordings there is mostly environmental, so it's not as bad as I thought. In the rather quiet toilet visit recording I think I hear some amount of self-noise. However, the vorbis compression might have something to do with that, I'm not sure. Thanks a lot Michael, this group/list looks like a treasure trove for my purposes. Regards, Philipp From fons at linuxaudio.org Sat Jul 27 09:17:54 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 27 Jul 2013 09:17:54 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130726223502.GA73801@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> Message-ID: <20130727091754.GA823@linuxaudio.org> On Fri, Jul 26, 2013 at 06:35:02PM -0400, Martin Cracauer wrote: > ebur128 --lufs: > Integrated loudness: -4.9 LUFS > Loudness range: 5.5 LU > Integrated threshold: -13.0 LUFS > Range threshold: -25.0 LUFS > Range min: -8.6 LUFS > Range max: -3.1 LUFS > Momentary max: -1.2 LUFS > Short term max: -2.4 LUFS This seems to be in the 'braindead' category... > I don't see that any of the value correspond with what lame needed to > not clip over a collection of different loudness clips. (means: some > clips that needed less --scale have higher numbers here and others > have lower) You can't expect that. Lame's limit are specific for the algorithm it uses. That would be the case for any lossy encoding system. > I assume the ebumeter output is more for making things sound even > (between different pieces) and not directly a tool to max out > anything, is that right? That is absolutely right. Systems such as EBU-R128 exist because it makes perfect sense to make things sound even (avoiding your listeners having to adjust the volume all the time), while 'maxing out' serves no useful purpose at all - it just destroys the sound if taken too far. In other words, such systems exist to *stop* you 'maxing out' everything. Lame's 'replaygain' meausurement has the same purpose. Note the value for the file in your original post: -9.1 dB. That means that an intelligent mp3 player will *reduce* the level of this file by 9.1 dB when playing it - it is already much too loud. 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 piedraarena at lavabit.com Sat Jul 27 16:51:23 2013 From: piedraarena at lavabit.com (profiles) Date: Sat, 27 Jul 2013 09:51:23 -0700 Subject: [LAU] linux compatible USB Audio Interface Message-ID: <1374943883.1899.28.camel@localhost.localdomain> Hello, I am a musician and recently joined this mailing list. I use Fedora18 and would like to use it to create music. I am having trouble getting off the ground as I have not identified a USB Audio Interface that is compatible with Fedora. I have visited the ALSA soundcard matrix wiki and a number of forums in hopes of finding a device. Unfortunately many of the models and makes talked about are a few years old now and, frequently, are no longer available from distributors. I imagine a few people on this list use some sort of soundcard to test/use all of the great software that is being created for linux users. I am inviting suggestions from people with experience in the application of audio hardware in the fedora environment. I am looking for a device that uses an independent power plug, connects with usb 2.0, and has both MIDI and analogue inputs and outputs. A device similar to this would be great: http://www.presonus.com/products/AudioBox-44VSL/media Thanks and I look forward to offering feedback on software once I am up and running. -Occhi From nachoen79 at hotmail.com Sat Jul 27 17:47:41 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Sat, 27 Jul 2013 19:47:41 +0200 Subject: [LAU] live audio on netjack1 In-Reply-To: <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> References: , <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> Message-ID: Hi David, thanks for the help, but when I change from jack to alsa on the slave machine I can listen only on that slave machine, not in the master. I don't know what I'm doing wrong. Date: Fri, 26 Jul 2013 15:14:16 -0700 From: davidtweed2003 at yahoo.com To: linux-audio-user at lists.linuxaudio.org Subject: Re: [LAU] live audio on netjack1 Hi Nacho, which direction are you trying to send live audio. master to slave or slave to master? if slave to master use alsa_in on slave machine to have that audio interface as a jack client. hope this helps. www.the-temp-agency.com/lollipopfactory From: Nacho - To: "linux-audio-user at lists.linuxaudio.org" Sent: Friday, July 26, 2013 12:54 PM Subject: [LAU] live audio on netjack1 Hi everybody. I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? Thanks! _______________________________________________ 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 davidtweed2003 at yahoo.com Sat Jul 27 22:16:45 2013 From: davidtweed2003 at yahoo.com (tweed) Date: Sat, 27 Jul 2013 15:16:45 -0700 (PDT) Subject: [LAU] live audio on netjack1 In-Reply-To: References: , <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> Message-ID: <1374963405.29371.YahooMailNeo@web161703.mail.bf1.yahoo.com> Hi Nacho, Which machine (master or slave) has the sound card through which you are running live audio? ? www.the-temp-agency.com/lollipopfactory ? >________________________________ > From: Nacho - >To: tweed ; "linux-audio-user at lists.linuxaudio.org" >Sent: Saturday, July 27, 2013 1:47 PM >Subject: RE: [LAU] live audio on netjack1 > > > > >Hi David, thanks for the help, but when I change from jack to alsa on the slave machine I can listen only on that slave machine, not in the master. >I don't know what I'm doing wrong. > > > >________________________________ >Date: Fri, 26 Jul 2013 15:14:16 -0700 >From: davidtweed2003 at yahoo.com >To: linux-audio-user at lists.linuxaudio.org >Subject: Re: [LAU] live audio on netjack1 > > >Hi Nacho, which direction are you trying to send live audio.?? master to slave or slave to master??? if slave to master use alsa_in on slave machine to have that audio interface as a jack client.? hope this helps. > >? >www.the-temp-agency.com/lollipopfactory >? > > > >>________________________________ >> From: Nacho - >>To: "linux-audio-user at lists.linuxaudio.org" >>Sent: Friday, July 26, 2013 12:54 PM >>Subject: [LAU] live audio on netjack1 >> >> >> >> >>Hi everybody. >> >>I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? >> >>Thanks! >> >>_______________________________________________ >>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 moshwe at gmail.com Sat Jul 27 22:33:19 2013 From: moshwe at gmail.com (Moshe Werner) Date: Sat, 27 Jul 2013 22:33:19 +0000 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130727093742.189879a5@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <20130727093742.189879a5@eeyore.mozart.uni-klu.ac.at> Message-ID: Hi Phillip, one very good device that wasn't mentioned here is the Sound Devices recorder. I used several of their recorders on film sets and had very good experiences. Also used zoom h2n on several occasions and have to say its not bad and gets a lot of things done for a reasonable price. But as mentioned it's not top notch sound quality. Hope that helps a bit. Moshe On 27 ?Jul 2013 10:37, "Philipp ?berbacher" wrote: > On Sat, 27 Jul 2013 10:37:53 +0900 > michael noble wrote: > > > If mic self-noise is truly a concern for you, then look beyond Zoom. > > This page provides a good comparison: > > > > http://www.avisoft.com/recordertests.htm > > > > I have the Sony PCM-M10 and a Tascam DR-680 for different purposes, > > and both are solid units for there price. Keep in mind that most > > portable recorders will not give pleasing results for "far away > > sources" unless you mean general ambient recording. For distant > > sources you would need a specialized microphone. For most > > versatility, I personally would recommend choosing a recorder with > > XLR inputs such that you can buy additional mics in the future. The > > Olympus LS-100 seems to be a favorite among recordists on a budget > > for this reason. > > > > If you want the advice of people far more experienced with these > > recorders and techniques than myself, I'd also suggest signing up the > > for the nature recordists mailing list. It's a very useful resource > > in general. > > > > http://tech.groups.yahoo.com/group/naturerecordists/ > > > Thanks everyone, > I spent quite some time sifting through reviews on the thomann > homepage, but I simply can't trust those, your advice is much > appreciated. > > Julien, I especially enjoyed the 'Toilet visit' :) > I think the noise in the recordings there is mostly environmental, so > it's not as bad as I thought. In the rather quiet toilet visit recording > I think I hear some amount of self-noise. However, the vorbis > compression might have something to do with that, I'm not sure. > > Thanks a lot Michael, this group/list looks like a treasure trove for > my purposes. > > Regards, > Philipp > > _______________________________________________ > 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 emiliano.grilli at gmail.com Sat Jul 27 22:48:49 2013 From: emiliano.grilli at gmail.com (Emiliano Grilli) Date: Sun, 28 Jul 2013 00:48:49 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: Hello, You might want to check also the "pocketrak" line from Yamaha, they are very very lightweight (I have the 2g and it's only 49 grams). The recordings are very good and the price fair. I'm really satisfied with mine. The small form factor of course limits the options for connecting external microphones (no phantom, no XLR) but as a self contained unit, it'great value. Hope this helps Regards Il giorno 26/lug/2013 15:26, "Philipp ?berbacher" ha scritto: > Hi there, > > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. > > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. > > Any recommendations? > > Regards, > Philipp > > _______________________________________________ > 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 marc at hacklava.net Sun Jul 28 04:45:00 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sun, 28 Jul 2013 00:45:00 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130726223502.GA73801@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> Message-ID: <20130728004500.1bff106d@hacklava.net> Hi Martin. The "r128gain" software is not the same than "ebur128" that comes with ebumeter. R128gain can be used to convert your files to mp3 in one pass, while ebur128 analyse and report. My understanding of EBU R128: what's important is the "integrated loudness"; the recommended value is -23 LUFS. If you run ebur128 without --lusfs, the integrated loudness would be -4.9 - -23 = 18.1 So you would have to reduce the volume by 18.1dB. -- Marc Martin Cracauer a ?crit : > Thanks again. Things are looking much better now. I embraced the > concept of not being able to predict this. Turns out I have old > screwed up things that really benefitted from revisiting them and > cleaning them up. > > Marc Lavall??e wrote on Thu, Jul 25, 2013 at 04:45:48PM -0400: > > Martin Cracauer a ??crit : > > > Is there a way to hook up ebumeter to just an audio file or a > > > stream not associated with real time? It seems to come in a jack > > > package only. > > > > > > Thanks again > > > Martin > > > > http://r128gain.sourceforge.net/ > > That works. Comes in the Debian ebumeter package, BTW. > > But how do I translate the output to a required db or ratio > adjustment? > > I have a file in front of me that was maxed out amplitude wise by > somebody else. According to lame I needed --scale 0.64 to get it not > to clip in lame (should be 3.9 db). Which, BTW was not the value it > first estimated that I would need. > > ebur128 --lufs: > Integrated loudness: -4.9 LUFS > Loudness range: 5.5 LU > Integrated threshold: -13.0 LUFS > Range threshold: -25.0 LUFS > Range min: -8.6 LUFS > Range max: -3.1 LUFS > Momentary max: -1.2 LUFS > Short term max: -2.4 LUFS > > I don't see that any of the value correspond with what lame needed to > not clip over a collection of different loudness clips. (means: some > clips that needed less --scale have higher numbers here and others > have lower) > > I assume the ebumeter output is more for making things sound even > (between different pieces) and not directly a tool to max out > anything, is that right? > > Martin From nachoen79 at hotmail.com Sun Jul 28 08:53:20 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Sun, 28 Jul 2013 10:53:20 +0200 Subject: [LAU] live audio on netjack1 In-Reply-To: <1374963405.29371.YahooMailNeo@web161703.mail.bf1.yahoo.com> References: <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> <1374963405.29371.YahooMailNeo@web161703.mail.bf1.yahoo.com> Message-ID: Hi david. The live audio is running on the slave machine. I want receive it on the master but at this moment I can not do it, only works with prerecorded audio or synths. Thanks! El 28/07/2013, a las 00:16, "tweed" escribi?: > Hi Nacho, > Which machine (master or slave) has the sound card through which you are running live audio? > > > www.the-temp-agency.com/lollipopfactory > > > From: Nacho - > To: tweed ; "linux-audio-user at lists.linuxaudio.org" > Sent: Saturday, July 27, 2013 1:47 PM > Subject: RE: [LAU] live audio on netjack1 > > Hi David, thanks for the help, but when I change from jack to alsa on the slave machine I can listen only on that slave machine, not in the master. > I don't know what I'm doing wrong. > > Date: Fri, 26 Jul 2013 15:14:16 -0700 > From: davidtweed2003 at yahoo.com > To: linux-audio-user at lists.linuxaudio.org > Subject: Re: [LAU] live audio on netjack1 > > Hi Nacho, which direction are you trying to send live audio. master to slave or slave to master? if slave to master use alsa_in on slave machine to have that audio interface as a jack client. hope this helps. > > www.the-temp-agency.com/lollipopfactory > > > From: Nacho - > To: "linux-audio-user at lists.linuxaudio.org" > Sent: Friday, July 26, 2013 12:54 PM > Subject: [LAU] live audio on netjack1 > > Hi everybody. > > I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? > > Thanks! > > _______________________________________________ > 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 hollunder at lavabit.com Sun Jul 28 09:06:05 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sun, 28 Jul 2013 11:06:05 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <20130727093742.189879a5@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130728110605.367baa37@eeyore.mozart.uni-klu.ac.at> On Sat, 27 Jul 2013 22:33:19 +0000 Moshe Werner wrote: > Hi Phillip, > one very good device that wasn't mentioned here is the Sound Devices > recorder. I used several of their recorders on film sets and had very > good experiences. Also used zoom h2n on several occasions and have to > say its not bad and gets a lot of things done for a reasonable price. > But as mentioned it's not top notch sound quality. > Hope that helps a bit. > Moshe Thanks, I've seen talk about those in the nature recordist mail archive, and while apparently the quality is good, they are also incredibly expensive. Regards, Philipp From jannis_achstetter at web.de Sun Jul 28 09:21:09 2013 From: jannis_achstetter at web.de (Jannis Achstetter) Date: Sun, 28 Jul 2013 11:21:09 +0200 Subject: [LAU] linux compatible USB Audio Interface In-Reply-To: <1374943883.1899.28.camel@localhost.localdomain> References: <1374943883.1899.28.camel@localhost.localdomain> Message-ID: <51F4E285.1050100@web.de> Am 27.07.2013 18:51, schrieb profiles: > Hello, > I am a musician and recently joined this mailing list. Welcome :) > I am inviting suggestions from people with experience in the application > of audio hardware in the fedora environment. I am looking for a device > that uses an independent power plug, connects with usb 2.0, and has both > MIDI and analogue inputs and outputs. A device similar to this would be > great: > > http://www.presonus.com/products/AudioBox-44VSL/media The Roland/Edirol UA-101 works well under linux and should have everything you need. They can be bought at a reasonable price online (used) and the developer of the linux kernel driver can be found on this mailing list in case there should be problems using the device. Best reagrds, Jannis From hollunder at lavabit.com Sun Jul 28 09:26:25 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sun, 28 Jul 2013 11:26:25 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130728112625.363a8140@eeyore.mozart.uni-klu.ac.at> On Sun, 28 Jul 2013 00:48:49 +0200 Emiliano Grilli wrote: > Hello, > You might want to check also the "pocketrak" line from Yamaha, they > are very very lightweight (I have the 2g and it's only 49 grams). > > The recordings are very good and the price fair. I'm really satisfied > with mine. > > The small form factor of course limits the options for connecting > external microphones (no phantom, no XLR) > but as a self contained unit, it'great value. > > Hope this helps Thanks Emiliano, seems like it's really small, which is nice, but I rather have something a little larger and a little better. I've seen that some nature recordists use a Rode NT1-A, which I happen to own, so an XLR input would be nice. Regards, Philipp From hollunder at lavabit.com Sun Jul 28 09:42:49 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Sun, 28 Jul 2013 11:42:49 +0200 Subject: [LAU] linux compatible USB Audio Interface In-Reply-To: <1374943883.1899.28.camel@localhost.localdomain> References: <1374943883.1899.28.camel@localhost.localdomain> Message-ID: <20130728114249.7897e979@eeyore.mozart.uni-klu.ac.at> On Sat, 27 Jul 2013 09:51:23 -0700 profiles wrote: > Hello, > > I am a musician and recently joined this mailing list. I use Fedora18 > and would like to use it to create music. I am having trouble getting > off the ground as I have not identified a USB Audio Interface that is > compatible with Fedora. I have visited the ALSA soundcard matrix wiki > and a number of forums in hopes of finding a device. Unfortunately > many of the models and makes talked about are a few years old now and, > frequently, are no longer available from distributors. I imagine a few > people on this list use some sort of soundcard to test/use all of the > great software that is being created for linux users. > > I am inviting suggestions from people with experience in the > application of audio hardware in the fedora environment. I am looking > for a device that uses an independent power plug, connects with usb > 2.0, and has both MIDI and analogue inputs and outputs. A device > similar to this would be great: > > http://www.presonus.com/products/AudioBox-44VSL/media > > Thanks and I look forward to offering feedback on software once I am > up and running. > > -Occhi Hi Occhi, my experience is limited to a device that doesn't fit your bill (Edirol UA-25), but at least it simply works. However, USB might not be the best way to go if you have other options. You can achieve lower latencies with pretty much everything else, firewire or however those card bus things are called. If you're just recording or playing back, that's no problem, but as soon as you need something like monitoring, the latency can bite you. My device has some direct monitoring option, probably for exactly that reason, but it is somewhat unsatisfactory. I guess if you tell us a bit more about your requirements, someone will be able to give you good advice. Regards, Philipp From fons at linuxaudio.org Sun Jul 28 11:27:34 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 28 Jul 2013 11:27:34 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728004500.1bff106d@hacklava.net> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> Message-ID: <20130728112733.GB31250@linuxaudio.org> On Sun, Jul 28, 2013 at 12:45:00AM -0400, Marc Lavall?e wrote: > The "r128gain" software is not the same than "ebur128" that comes > with ebumeter. And where is the source code ? The SF packages contain a copy of the GPL, and *only* binaries and shared libraries (which probably shouldn't get installed that way on any system). The SVN links don't work. Something stinks here. 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 alice-dsl.net Sun Jul 28 11:47:03 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 28 Jul 2013 13:47:03 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728112733.GB31250@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> Message-ID: <1375012023.712.9.camel@archlinux> On Sun, 2013-07-28 at 11:27 +0000, Fons Adriaensen wrote: > And where is the source code ? > The SVN links don't work. For the other projects the SVN addresses don't exist too http://sourceforge.net/u/pbelkner/profile/ From edogawa at aon.at Sun Jul 28 12:00:04 2013 From: edogawa at aon.at (Edgar Aichinger) Date: Sun, 28 Jul 2013 14:00:04 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728112733.GB31250@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> Message-ID: <137503478.CER9Kdcr00@edhp.site> Am Sonntag, 28. Juli 2013, 11:27:34 schrieb Fons Adriaensen: > On Sun, Jul 28, 2013 at 12:45:00AM -0400, Marc Lavall?e wrote: > > > The "r128gain" software is not the same than "ebur128" that comes > > with ebumeter. > > And where is the source code ? The SF packages contain a copy > of the GPL, and *only* binaries and shared libraries (which > probably shouldn't get installed that way on any system). http://sourceforge.net/projects/r128gain/files/r128gain/1.0/ lists a src.tar.gz that is downloadable via http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-src.tar.gz/download Edgar > The SVN links don't work. Something stinks here. > > Ciao, > > From ralf.mardorf at alice-dsl.net Sun Jul 28 12:21:01 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 28 Jul 2013 14:21:01 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <137503478.CER9Kdcr00@edhp.site> References: <20130725182150.GA97830@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> <137503478.CER9Kdcr00@edhp.site> Message-ID: <1375014061.712.11.camel@archlinux> On Sun, 2013-07-28 at 14:00 +0200, Edgar Aichinger wrote: > a src.tar.gz that is downloadable JFTR $ make does download additional sources From nettings at stackingdwarves.net Sun Jul 28 12:44:02 2013 From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=) Date: Sun, 28 Jul 2013 14:44:02 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728112733.GB31250@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> Message-ID: <51F51212.1000604@stackingdwarves.net> On 07/28/2013 01:27 PM, Fons Adriaensen wrote: > On Sun, Jul 28, 2013 at 12:45:00AM -0400, Marc Lavall?e wrote: > >> The "r128gain" software is not the same than "ebur128" that comes >> with ebumeter. > > And where is the source code ? http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-src.tar.gz/download haven't tried to build it, but it looks like a sane and complete source package after a cursory glance. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From ralf.mardorf at alice-dsl.net Sun Jul 28 13:20:54 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Sun, 28 Jul 2013 15:20:54 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <51F51212.1000604@stackingdwarves.net> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> <51F51212.1000604@stackingdwarves.net> Message-ID: <1375017654.712.17.camel@archlinux> On Sun, 2013-07-28 at 14:44 +0200, J?rn Nettingsmeier wrote: > On 07/28/2013 01:27 PM, Fons Adriaensen wrote: > > On Sun, Jul 28, 2013 at 12:45:00AM -0400, Marc Lavall?e wrote: > > > >> The "r128gain" software is not the same than "ebur128" that comes > >> with ebumeter. > > > > And where is the source code ? > > http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-src.tar.gz/download > > haven't tried to build it, but it looks like a sane and complete source > package after a cursory glance. Isn't there the need to provide all used sources instead of downloading them? However, I don't care about it. $ grep DOWNLOAD Makefile define DOWNLOAD $(call DOWNLOAD,ftp://ftp.nluug.nl/pub/ImageMagick/$(@F)) # $(call DOWNLOAD,ftp://mirror.checkdomain.de/imagemagick/$(@F)) $(call DOWNLOAD,http://www.ijg.org/files/jpegsrc.v$(JPEG_VER).tar.gz) $(call DOWNLOAD,http://ffmpeg.org/releases/$(@F)) $(call DOWNLOAD,$(call SFURL,sox,sox/$(SOXVER)/$(@F))) $(call DOWNLOAD,http://www.wavpack.com/$(@F)) $(call DOWNLOAD,$(call SFURL,flac,flac-src/$(FLAC)-src/$(@F))) $(call DOWNLOAD,ftp://ftp.videolan.org/pub/videolan/x264/snapshots/$(@F)) $(call DOWNLOAD,http://webm.googlecode.com/files/$(@F)) $(call DOWNLOAD,http://www.mega-nerd.com/libsndfile/files/$(@F)) $(call DOWNLOAD,http://downloads.xiph.org/releases/opus/$(@F)) $(call DOWNLOAD,http://downloads.xiph.org/releases/vorbis/$(@F)) $(call DOWNLOAD,http://downloads.xiph.org/releases/ogg/$(@F)) $(call DOWNLOAD,$(call SFURL,lame,lame/$(LAME_MAJOR)/$(@F))) $(call DOWNLOAD,ftp://sourceware.org/pub/pthreads-win32/$(@F)) $(call DOWNLOAD,http://www.tortall.net/projects/yasm/releases/$(@F)) $(call DOWNLOAD,http://pkgconfig.freedesktop.org/releases//$(@F)) FWIW it didn't build by the first trial on Arch Linux: $ wget http://kent.dl.sourceforge.net/project/r128gain/r128gain/1.0/r128gain-1.0-src.tar.gz [snip] cc1: all warnings being treated as errors make[2]: *** [gtkgui.o] Error 1 rm r128gain_mkdir.o r128gain_s_rg.o r128gain_sink.o r128gain_cmd.o r128gain_y.o r128gain_reg.o r128gain_f_ff.o r128gui_io.o r128gain_s.o r128gain_peak.o r128gui_util.o r128gain_f_sox.o r128gain_ps.o r128gui_ps.o r128gain_s_r128.o r128gain_meta.o r128gain_f.o make[2]: Leaving directory `/usr/src/r128gain-1.0/full/gtk3/build/r128gain' make[1]: *** [full/gtk3/build/r128gain/r128gain] Error 2 make[1]: Leaving directory `/usr/src/r128gain-1.0' make: *** [all-full-gtk3] Error 2 Regards, Ralf From marc at hacklava.net Sun Jul 28 14:21:29 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sun, 28 Jul 2013 10:21:29 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <1375017654.712.17.camel@archlinux> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> <51F51212.1000604@stackingdwarves.net> <1375017654.712.17.camel@archlinux> Message-ID: <20130728102129.31cbb6ce@telecino> Ralf Mardorf a ?crit : > Isn't there the need to provide all used sources instead of > downloading them? However, I don't care about it. I tried to compile it. Here's what I did. # I downloaded the source files in the same folder: http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-src.tar.gz http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-tools.tar.gz # I unarchived them: tar -xzf r128gain-1.0-src.tar.gz tar -xzf r128gain-1.0-tools.tar.gz # I jumped into the resulting source folder: cd r128gain-1.0 # I start rebuilding the world (it's a llloong process): ./configure cmake make # I tried to install it: sudo make install But it didn't work, complaining about a missing sox development package. So I stopped. You're welcome to give it a try. There's probably a way to compile r128gain without compiling all the required libraries, in order to use the already installed system libraries. As a casual user, the binaries works fine, although I had to symlink the liblame library in the r128gain-tools folder (liblame is probably missing for copyright issues); for now I believe it's the easiest way to install r128gain, until official packages appears in popular Linux based distributions. -- Marc From pedro.lopez.cabanillas at gmail.com Sun Jul 28 15:49:47 2013 From: pedro.lopez.cabanillas at gmail.com (Pedro Lopez-Cabanillas) Date: Sun, 28 Jul 2013 17:49:47 +0200 Subject: [LAU] [ANN] KMidimon 0.7.5 released In-Reply-To: <201009171903.54989.pedro.lopez.cabanillas@gmail.com> References: <201009171903.54989.pedro.lopez.cabanillas@gmail.com> Message-ID: <2520423.yskEhavifp@boccanegra.localdomain> KMidimon is a MIDI monitor for Linux using ALSA sequencer and KDE4 user interface. Changes in release 0.7.5 * Japanese translation, by Oota Toshiya * Fixed a crash at exit when playing Copyright (C) 2005-2013, Pedro Lopez-Cabanillas License: GPL v2 More info http://kmidimon.sourceforge.net Sources http://sourceforge.net/projects/kmidimon/files/0.7.5/ Regards, Pedro From davidtweed2003 at yahoo.com Sun Jul 28 22:04:43 2013 From: davidtweed2003 at yahoo.com (tweed) Date: Sun, 28 Jul 2013 15:04:43 -0700 (PDT) Subject: [LAU] live audio on netjack1 In-Reply-To: References: <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> <1374963405.29371.YahooMailNeo@web161703.mail.bf1.yahoo.com> Message-ID: <1375049083.41008.YahooMailNeo@web161705.mail.bf1.yahoo.com> Hi Nacho, Start netjack on both machines as you described in your first post. On the slave machine run the command "alsa_in" in a terminal with your slave's sound card as argument ("alsa_in -d hw:0" if the soundcard you want to use is the first device -? '0' is for the 1st device). this will simply create a jack client for sound card on the slave machine. You can then connect this client in the qjackctl connections window to other clients (in your case probably to 'system' since that is the netjack client on slave).? run "aplay -l" (thats an L) in a terminal to see the the soundcard number (or in qjackctl setup).? run 'man alsa_in' in a terminal to learn more.? hope this helps. ? www.the-temp-agency.com/lollipopfactory ? >________________________________ > From: Nacho - To: tweed >Cc: tweed ; "linux-audio-user at lists.linuxaudio.org" >Sent: Sunday, July 28, 2013 4:53 AM >Subject: Re: [LAU] live audio on netjack1 > > > >Hi david. The live audio is running on the slave machine. I want receive it on the master but at this moment I can not do it, only works with prerecorded audio or synths. > > >Thanks! > > > > > >El 28/07/2013, a las 00:16, "tweed" escribi?: > > >Hi Nacho, >>Which machine (master or slave) has the sound card through which you are running live audio? >> >> >> >>? >>www.the-temp-agency.com/lollipopfactory >>? >> >> >> >>>________________________________ >>> From: Nacho - >>>To: tweed ; "linux-audio-user at lists.linuxaudio.org" >>>Sent: Saturday, July 27, 2013 1:47 PM >>>Subject: RE: [LAU] live audio on netjack1 >>> >>> >>> >>> >>>Hi David, thanks for the help, but when I change from jack to alsa on the slave machine I can listen only on that slave machine, not in the master. >>>I don't know what I'm doing wrong. >>> >>> >>> >>>________________________________ >>>Date: Fri, 26 Jul 2013 15:14:16 -0700 >>>From: davidtweed2003 at yahoo.com >>>To: linux-audio-user at lists.linuxaudio.org >>>Subject: Re: [LAU] live audio on netjack1 >>> >>> >>>Hi Nacho, which direction are you trying to send live audio.?? master to slave or slave to master??? if slave to master use alsa_in on slave machine to have that audio interface as a jack client.? hope this helps. >>> >>>? >>>www.the-temp-agency.com/lollipopfactory >>>? >>> >>> >>> >>>>________________________________ >>>> From: Nacho - >>>>To: "linux-audio-user at lists.linuxaudio.org" >>>>Sent: Friday, July 26, 2013 12:54 PM >>>>Subject: [LAU] live audio on netjack1 >>>> >>>> >>>> >>>> >>>>Hi everybody. >>>> >>>>I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? >>>> >>>>Thanks! >>>> >>>>_______________________________________________ >>>>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 marc at hacklava.net Sun Jul 28 23:50:44 2013 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Sun, 28 Jul 2013 19:50:44 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130727091754.GA823@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> Message-ID: <20130728195044.740eb5b0@telecino> I'm sure it's been answered many times, but here's my question: For loudness evaluation, how's EBU R128 "better" than ReplayGain? (Or why use EBU R128 instead of ReplayGain?) (A link to the suitable InterWeb resource would be appropriate) B.T.W, there's another EBU R128 utility to normalize audio files: https://github.com/jiixyj/libebur128 It can add ReplayGain tags to existing mp3 files. Nice! I add to checkout commit 1c0e8da to avoid linking errors, as mentioned here: https://github.com/jiixyj/libebur128/issues/20 -- Marc Fons Adriaensen a ?crit : > > I assume the ebumeter output is more for making things sound even > > (between different pieces) and not directly a tool to max out > > anything, is that right? > > That is absolutely right. Systems such as EBU-R128 exist because > it makes perfect sense to make things sound even (avoiding your > listeners having to adjust the volume all the time), while 'maxing > out' serves no useful purpose at all - it just destroys the sound > if taken too far. In other words, such systems exist to *stop* you > 'maxing out' everything. > > Lame's 'replaygain' meausurement has the same purpose. Note the > value for the file in your original post: -9.1 dB. That means > that an intelligent mp3 player will *reduce* the level of this > file by 9.1 dB when playing it - it is already much too loud. -- Marc From nettings at stackingdwarves.net Mon Jul 29 01:27:17 2013 From: nettings at stackingdwarves.net (=?UTF-8?B?SsO2cm4gTmV0dGluZ3NtZWllcg==?=) Date: Mon, 29 Jul 2013 03:27:17 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728195044.740eb5b0@telecino> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> Message-ID: <51F5C4F5.1010900@stackingdwarves.net> On 07/29/2013 01:50 AM, Marc Lavall?e wrote: > > I'm sure it's been answered many times, but here's my question: > For loudness evaluation, how's EBU R128 "better" than ReplayGain? > (Or why use EBU R128 instead of ReplayGain?) > (A link to the suitable InterWeb resource would be appropriate) ReplayGain is a laudable effort at solving the issue of arbitrary loudness jumps for audio consumers. More power to them :) However, I believe the psychoacoustic "model" of EBU R128 is superior, although certainly not perfect (colleagues have mentioned to me ways of cheating R128 to achieve higher programme loudness for constant LUFS readings, but I've yet to see an actual demonstration). But regardless of technical merit, I would always choose EBU 128 over ReplayGain, because it has the magic letters "EBU" in it, and the underlying measurement scheme has the even more magic "ITU". Don't forget that any perceptive loudness normalization proposal is basically a "non-agression pact" in the loudness war. If you don't nuke my speakers (and wreck my music), I won't nuke yours. So it makes sense to flock around the biggest fish in the pond, because their involvement and statement of intent carries weight. Until all broadcasters follow suit, those who behave are at a commercial disadvantage, so the whole process has all the interesting properties of a game theory problem. For an euro-centric bastard like me, the European Broadcast Union is the mightiest normative entity in the universe. Some weird people think differently, but those take solace in the fact that the actual loudness determination is governed by an International Telecommunication Union standard (and that should be good enough for everyone :) So yes, R128 is a very safe bet indeed. -- 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 Mon Jul 29 01:29:23 2013 From: nettings at stackingdwarves.net (=?UTF-8?B?SsO2cm4gTmV0dGluZ3NtZWllcg==?=) Date: Mon, 29 Jul 2013 03:29:23 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <1375017654.712.17.camel@archlinux> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> <51F51212.1000604@stackingdwarves.net> <1375017654.712.17.camel@archlinux> Message-ID: <51F5C573.7060202@stackingdwarves.net> On 07/28/2013 03:20 PM, Ralf Mardorf wrote: > On Sun, 2013-07-28 at 14:44 +0200, J?rn Nettingsmeier wrote: >> On 07/28/2013 01:27 PM, Fons Adriaensen wrote: >>> On Sun, Jul 28, 2013 at 12:45:00AM -0400, Marc Lavall?e wrote: >>> >>>> The "r128gain" software is not the same than "ebur128" that comes >>>> with ebumeter. >>> >>> And where is the source code ? >> >> http://sourceforge.net/projects/r128gain/files/r128gain/1.0/r128gain-1.0-src.tar.gz/download >> >> haven't tried to build it, but it looks like a sane and complete source >> package after a cursory glance. > > Isn't there the need to provide all used sources instead of downloading > them? no, there isn't. it is neither required by the gpl, nor would it be useful except in very rare cases. > However, I don't care about it. sigh. -- J?rn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister f?r Veranstaltungstechnik (B?hne/Studio) Tonmeister VDT http://stackingdwarves.net From ralf.mardorf at alice-dsl.net Mon Jul 29 06:00:40 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 29 Jul 2013 08:00:40 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <51F5C573.7060202@stackingdwarves.net> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130728004500.1bff106d@hacklava.net> <20130728112733.GB31250@linuxaudio.org> <51F51212.1000604@stackingdwarves.net> <1375017654.712.17.camel@archlinux> <51F5C573.7060202@stackingdwarves.net> Message-ID: <1375077640.2711.29.camel@q> On Mon, 2013-07-29 at 03:29 +0200, J?rn Nettingsmeier wrote: > On 07/28/2013 03:20 PM, Ralf Mardorf wrote: > > On Sun, 2013-07-28 at 14:44 +0200, J?rn Nettingsmeier wrote: > > Isn't there the need to provide all used sources instead of downloading > > them? > > no, there isn't. it is neither required by the gpl, nor would it be > useful except in very rare cases. Ok. I'm aware that a coder can add a list of dependencies, but I thought if it ignores installed dependencies and include it to the own source code, it also must be provided by the own source code. It was the first time I saw "make" downloading dependencies and I'm surprised, since the dependencies were already installed. Thank you J?rn, Ralf From fons at linuxaudio.org Mon Jul 29 09:25:31 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 29 Jul 2013 09:25:31 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130728195044.740eb5b0@telecino> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> Message-ID: <20130729092531.GA28352@linuxaudio.org> On Sun, Jul 28, 2013 at 07:50:44PM -0400, Marc Lavall?e wrote: > I'm sure it's been answered many times, but here's my question: > For loudness evaluation, how's EBU R128 "better" than ReplayGain? > (Or why use EBU R128 instead of ReplayGain?) > (A link to the suitable InterWeb resource would be appropriate) http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_2.0_specification The proposed RG2 spec uses the same algorithm as R-128. 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 nachoen79 at hotmail.com Mon Jul 29 09:41:57 2013 From: nachoen79 at hotmail.com (Nacho -) Date: Mon, 29 Jul 2013 11:41:57 +0200 Subject: [LAU] live audio on netjack1 In-Reply-To: <1375049083.41008.YahooMailNeo@web161705.mail.bf1.yahoo.com> References: <1374876856.688.YahooMailNeo@web161706.mail.bf1.yahoo.com> <1374963405.29371.YahooMailNeo@web161703.mail.bf1.yahoo.com> , <1375049083.41008.YahooMailNeo@web161705.mail.bf1.yahoo.com> Message-ID: Hi David.Solved. That worked perfect.Many thanks! Date: Sun, 28 Jul 2013 15:04:43 -0700 From: davidtweed2003 at yahoo.com Subject: Re: [LAU] live audio on netjack1 To: nachoen79 at hotmail.com; tweed at lollipopfactory.com CC: linux-audio-user at lists.linuxaudio.org Hi Nacho, Start netjack on both machines as you described in your first post. On the slave machine run the command "alsa_in" in a terminal with your slave's sound card as argument ("alsa_in -d hw:0" if the soundcard you want to use is the first device - '0' is for the 1st device). this will simply create a jack client for sound card on the slave machine. You can then connect this client in the qjackctl connections window to other clients (in your case probably to 'system' since that is the netjack client on slave). run "aplay -l" (thats an L) in a terminal to see the the soundcard number (or in qjackctl setup). run 'man alsa_in' in a terminal to learn more. hope this helps. www.the-temp-agency.com/lollipopfactory From: Nacho - Cc: tweed ; "linux-audio-user at lists.linuxaudio.org" Sent: Sunday, July 28, 2013 4:53 AM Subject: Re: [LAU] live audio on netjack1 Hi david. The live audio is running on the slave machine. I want receive it on the master but at this moment I can not do it, only works with prerecorded audio or synths. Thanks! El 28/07/2013, a las 00:16, "tweed" escribi?: Hi Nacho, Which machine (master or slave) has the sound card through which you are running live audio? www.the-temp-agency.com/lollipopfactory From: Nacho - To: tweed ; "linux-audio-user at lists.linuxaudio.org" Sent: Saturday, July 27, 2013 1:47 PM Subject: RE: [LAU] live audio on netjack1 Hi David, thanks for the help, but when I change from jack to alsa on the slave machine I can listen only on that slave machine, not in the master. I don't know what I'm doing wrong. Date: Fri, 26 Jul 2013 15:14:16 -0700 From: davidtweed2003 at yahoo.com To: linux-audio-user at lists.linuxaudio.org Subject: Re: [LAU] live audio on netjack1 Hi Nacho, which direction are you trying to send live audio. master to slave or slave to master? if slave to master use alsa_in on slave machine to have that audio interface as a jack client. hope this helps. www.the-temp-agency.com/lollipopfactory From: Nacho - To: "linux-audio-user at lists.linuxaudio.org" Sent: Friday, July 26, 2013 12:54 PM Subject: [LAU] live audio on netjack1 Hi everybody. I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible? Thanks! _______________________________________________ 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 hollunder at lavabit.com Mon Jul 29 09:45:01 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 29 Jul 2013 11:45:01 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729092531.GA28352@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> <20130729092531.GA28352@linuxaudio.org> Message-ID: <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> On Mon, 29 Jul 2013 09:25:31 +0000 Fons Adriaensen wrote: > On Sun, Jul 28, 2013 at 07:50:44PM -0400, Marc Lavall?e wrote: > > > I'm sure it's been answered many times, but here's my question: > > For loudness evaluation, how's EBU R128 "better" than ReplayGain? > > (Or why use EBU R128 instead of ReplayGain?) > > (A link to the suitable InterWeb resource would be appropriate) > > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_2.0_specification > > The proposed RG2 spec uses the same algorithm as R-128. > > Ciao, > One of the first things I thought when I saw the R-128 talk was" Could this be used for a better replaygain?". Very nice to see that such an effort is underway. Now the obvious question is: Will this work better than the old replay gain, and by how much? Has anyone done tests? Regards, Philipp From fons at linuxaudio.org Mon Jul 29 10:13:16 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 29 Jul 2013 10:13:16 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> <20130729092531.GA28352@linuxaudio.org> <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130729101316.GB28352@linuxaudio.org> On Mon, Jul 29, 2013 at 11:45:01AM +0200, Philipp ?berbacher wrote: > On Mon, 29 Jul 2013 09:25:31 +0000 Fons Adriaensen wrote: > > > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification > > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_2.0_specification > > > > The proposed RG2 spec uses the same algorithm as R-128. > > One of the first things I thought when I saw the R-128 talk was" Could > this be used for a better replaygain?". Very nice to see that such an > effort is underway. > > Now the obvious question is: Will this work better than the old replay > gain, and by how much? Has anyone done tests? See the Dolby paper referenced in the second link. 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 alice-dsl.net Mon Jul 29 10:24:17 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 29 Jul 2013 12:24:17 +0200 Subject: [LAU] alsamixer Message-ID: <1375093457.675.12.camel@archlinux> Hi :) several times I got the reply that all settings of RME sound cards are available by alsamixer, so seemingly I'm doing something wrong. This isn't a request regarding to my RME sound card, just a question about how to use alsamixer. I'm missing how to set up the routing. You for sure remember that using TotaMix on Windows I can access all 8 ADAT channels, but using TotalMix on Linux, aka hdspmixer, I only can use ADAT channels 1 and 2, the others aren't available by jack. hdspmixer shows a signal for all 8 ADAT channels, the signal only is missing for jack. So how can I use alsamixer to ensure that e.g. ADAT channel 3 is jack capture_9? ADAT channel 1 is available by jack capture_7, channel 2 by capture_8, but than it stops, the other captures don't get the audio signal. $ amixer scontents Simple mixer control 'ADAT Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'ADAT SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'No Lock' Simple mixer control 'AES Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'AES SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'No Lock' Simple mixer control 'Chn',1 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 24 [38%] Simple mixer control 'Chn',2 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',3 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',4 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',5 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',6 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',7 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',8 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 0 [0%] Simple mixer control 'Chn',9 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',10 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',11 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',12 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',13 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 19 [30%] Simple mixer control 'Chn',14 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',15 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'Chn',16 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 64 Mono: 64 [100%] Simple mixer control 'External Rate',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'Internal Clock',0 Capabilities: enum Items: '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: '48 kHz' Simple mixer control 'Preferred Sync Reference',0 Capabilities: enum Items: 'Word Clock' 'ADAT' 'AES' 'SPDIF' 'Sync In' Item0: 'Word Clock' Simple mixer control 'SPDIF Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'SPDIF SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'No Lock' Simple mixer control 'SYNC IN Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'SYNC IN SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'No Lock' Simple mixer control 'System Clock Mode',0 Capabilities: enum Items: 'Master' 'AutoSync' Item0: 'Master' Simple mixer control 'System Sample Rate',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 27000 - 207000 Mono: 48000 [12%] Simple mixer control 'TCO Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'TCO SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'N/A' Simple mixer control 'WC Frequency',0 Capabilities: enum Items: 'No Lock' '32 kHz' '44.1 kHz' '48 kHz' '64 kHz' '88.2 kHz' '96 kHz' '128 kHz' '176.4 kHz' '192 kHz' Item0: 'No Lock' Simple mixer control 'WC SyncCheck',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' 'N/A' Item0: 'No Lock' Regards, Ralf From julien at mail.upb.de Mon Jul 29 10:42:22 2013 From: julien at mail.upb.de (Julien Claassen) Date: Mon, 29 Jul 2013 12:42:22 +0200 (CEST) Subject: [LAU] alsamixer In-Reply-To: <1375093457.675.12.camel@archlinux> References: <1375093457.675.12.camel@archlinux> Message-ID: Hello Ralf! You say, that hdpsmixer is showing all channels and you can access them, right? If so, this looks more like a problem of your card's configuration then or of your JACK server, than a problem concerning the mixer application as such. If people say, that you can use alsamixer to setup your card, then you probably can. You know how to start it. You can press f5 to see all items and then slowly walk through the list. Left/right move between items and up/down change values. Tehre's is help available directly from within the mixer. It's not the best utility for such a complex card, since it will not map the structure to the screen. I suppose, that hdspmixer shows you more of the real structure of your card. In alsamixer routing is a one-dimensional thing. You need to know, which item does what. Sometimes names can be terribly unexplicative or at least unhelpful. Still, give it a try and see, how far you get. You probably have a manual for your card, which might help you with some questions. But I really think, that you might have a closer look at your JACK setup. Does JACK show all channels? Do you have ports uto capture_16 or whatever it should be? Do you also have the correct number of playback ports as such? I'm not talking about signals, I'm just asking for the existance. Well, I'm probably the wrong person to ask, since I won't be able to discuss this much further with you. I only know the systematic basics and I don't have a lot of experience with internal routing of cards. I'm glad my two cards work now. :-) Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From ralf.mardorf at alice-dsl.net Mon Jul 29 10:56:33 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 29 Jul 2013 12:56:33 +0200 Subject: [LAU] alsamixer In-Reply-To: References: <1375093457.675.12.camel@archlinux> Message-ID: <1375095393.675.21.camel@archlinux> On Mon, 2013-07-29 at 12:42 +0200, Julien Claassen wrote: > Do you also have the correct number of playback ports as such? I'm not > talking about signals, I'm just asking for the existance. capture 1 to 14 2 analog ins 2 SPDIF ins 2 AES ins 8 ADAT ins ------------ 14 So they are all there. playback 1 to 16 same as above + perhaps phones out So they are there too. On Windows it's possible to set up a matrix, e.g. if you have several RME cards, you can readjust this matrix, this isn't available for _my_ Linux. TotalMix and hdpmixer provides "submixes" and I can't see how to handle this or how to set the matrix for the routing using alsamixer. Thank you, Ralf From hollunder at lavabit.com Mon Jul 29 13:27:33 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 29 Jul 2013 15:27:33 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729101316.GB28352@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> <20130729092531.GA28352@linuxaudio.org> <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> <20130729101316.GB28352@linuxaudio.org> Message-ID: <20130729152733.662425a4@eeyore.mozart.uni-klu.ac.at> On Mon, 29 Jul 2013 10:13:16 +0000 Fons Adriaensen wrote: > On Mon, Jul 29, 2013 at 11:45:01AM +0200, Philipp ?berbacher wrote: > > > On Mon, 29 Jul 2013 09:25:31 +0000 Fons Adriaensen > > wrote: > > > > > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification > > > http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_2.0_specification > > > > > > The proposed RG2 spec uses the same algorithm as R-128. > > > > One of the first things I thought when I saw the R-128 talk was" > > Could this be used for a better replaygain?". Very nice to see that > > such an effort is underway. > > > > Now the obvious question is: Will this work better than the old > > replay gain, and by how much? Has anyone done tests? > > See the Dolby paper referenced in the second link. > > Ciao, > Thanks Fons, while the paper (http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/ AES128-Loudness-Normalization-Portable-Media-Players.pdf) is interesting, it's not what I was looking for. What needs testing is whether the loudness adjustment of RG2 compared to RG1 is perceived as 'closer to equal loudness'. When some sort of actual listening test shows that RG2 performs significantly better, good, otherwise I don't see the point. I do find their recommended conversion equation questionable, they came up with it based on their sample set plus thumb measure. There's no telling what the results will be, but I don't expect them to be any good. Regards, Philipp From fons at linuxaudio.org Mon Jul 29 14:15:28 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 29 Jul 2013 14:15:28 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729152733.662425a4@eeyore.mozart.uni-klu.ac.at> References: <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> <20130729092531.GA28352@linuxaudio.org> <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> <20130729101316.GB28352@linuxaudio.org> <20130729152733.662425a4@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130729141528.GC28352@linuxaudio.org> On Mon, Jul 29, 2013 at 03:27:33PM +0200, Philipp ?berbacher wrote: > while the paper > (http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/ > AES128-Loudness-Normalization-Portable-Media-Players.pdf) > is interesting, it's not what I was looking for. What needs testing is > whether the loudness adjustment of RG2 compared to RG1 is perceived as > 'closer to equal loudness'. When some sort of actual listening test > shows that RG2 performs significantly better, good, otherwise I don't > see the point. ITU-1770 on which both R-128 and RG2 are based has been tested extensively over a long time and shown to work very well. AFAIK there was no comparable amount of testing done on RG1. > I do find their recommended conversion equation questionable, they came > up with it based on their sample set plus thumb measure. There's no > telling what the results will be, but I don't expect them to be any > good. Why not ? The tests on the sample set (which is not small) show quite a good agreement between the two systems. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From hollunder at lavabit.com Mon Jul 29 16:21:59 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 29 Jul 2013 18:21:59 +0200 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729141528.GC28352@linuxaudio.org> References: <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130728195044.740eb5b0@telecino> <20130729092531.GA28352@linuxaudio.org> <20130729114501.0d5552cc@eeyore.mozart.uni-klu.ac.at> <20130729101316.GB28352@linuxaudio.org> <20130729152733.662425a4@eeyore.mozart.uni-klu.ac.at> <20130729141528.GC28352@linuxaudio.org> Message-ID: <20130729182159.417feeb9@eeyore.mozart.uni-klu.ac.at> On Mon, 29 Jul 2013 14:15:28 +0000 Fons Adriaensen wrote: > On Mon, Jul 29, 2013 at 03:27:33PM +0200, Philipp ?berbacher wrote: > > > while the paper > > (http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/ > > AES128-Loudness-Normalization-Portable-Media-Players.pdf) > > is interesting, it's not what I was looking for. What needs testing > > is whether the loudness adjustment of RG2 compared to RG1 is > > perceived as 'closer to equal loudness'. When some sort of actual > > listening test shows that RG2 performs significantly better, good, > > otherwise I don't see the point. > > ITU-1770 on which both R-128 and RG2 are based has been tested > extensively over a long time and shown to work very well. AFAIK there > was no comparable amount of testing done on RG1. Really? I can't find any proper references, at least not in the latest version of the ITU-1770 paper (http://www.itu.int/rec/R-REC-BS.1770-3-201208-I/en). They only roughly describe some tests, first as mostly female speech (p. 13-14), later they claim it was a broad base of material (p. 18). I can't see any obvious reference to details of those tests, the second referenced paper (Evaluation of Objective Loudness Meters) seems to describe it, but it seems this paper is not publicly accessible. However, my point here is that it has to show its performance and especially superiority over the established RG1 in practice, everything else is of little significance. > > I do find their recommended conversion equation questionable, they > > came up with it based on their sample set plus thumb measure. > > There's no telling what the results will be, but I don't expect > > them to be any good. > > Why not ? The tests on the sample set (which is not small) show quite > a good agreement between the two systems. > > Ciao, According to the R-128 paper the standard deviation of their samples from the line they plotted there is 1dB, no idea what it is from the slope (-1) they guessed there. Still, given to songs it could well be 2dB or more off, but my main point is that this is still with _their_ data set. I guess that with another data set the differences could be larger and significant to the point where the error is large enough for the conversion to be a bad thing?. I wonder whether a better, albeit slightly more complicated way of conversion can be found, this one was apparently chosen primarily for its simplicity. Don't get me wrong, I like ReplayGain, and I'd like to see further adoption, I'm just not sure the R-128 based one will actually be better and cause further adoption. I fear it could cause confusion and further problems instead. Sometimes I also like to argue and see the bad parts in everything. On the upside, the temperature just dropped below 30? for a change. Have a nice day, Philipp From djbarney at djbarney.org Mon Jul 29 17:50:56 2013 From: djbarney at djbarney.org (Barney Holmes) Date: Mon, 29 Jul 2013 18:50:56 +0100 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> Hi, The Sony Minidisc (remember those?) was considered a desirable item for live recording in its time. I'm currently experimenting with a Sony Walkman Minidisc Net MD (Walkman MZ-N707 Type-R) that I picked up in a charity shop for ~6 pounds. Its probably a collectors item, it was first introduced in 2002. But I'm curious if it lives up to all the recommendations I heard at the time from live sound recordists. Did some seaside and other recordings just using a basic mono microphone. Sounds OK when I play back on headphones from the device. I'm currently transferring to my PC. The problem with the Minidisc is that it has too much DRM on it (which is one of the reasons it failed as a technology). Fibre optic connection only works INTO the device. Recordings cannot be transferred off the device in digital form and have to be re-recorded through an analog output - but the quality loss is negligible I think. Currently looking at "linux-minidisc" for some possible ways around the DRM. Who knows what other second had devices are out there ? It's amazing what you can find in charity shops as well ;) djbarney On Fri, July 26, 2013 2:26 pm, Philipp ??berbacher wrote: > Hi there, > > I finally want to buy a small field recording device, a zoom H* or > similar, and I hope that some of you have experience with some of those > devices and can give me some advice. > > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. > > Any recommendations? > > Regards, > Philipp > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > ~~~ Home site - http://djbarney.org From gianfranco at portalmod.com.br Mon Jul 29 18:07:02 2013 From: gianfranco at portalmod.com.br (Gianfranco Ceccolini) Date: Mon, 29 Jul 2013 15:07:02 -0300 Subject: [LAU] [ANN] - MOD Pitch Shifter Message-ID: Hi everybody We are very happy to announce the MOD Pitch Shifter https://github.com/portalmod/mod-pitchshifter.git After some unsuccessfull time looking for a nice pitch shifter to offer in the MOD Cloud we decided to make one ourselves. Kudos for Andre Coutinho who did most of the coding. We tried VocProc and Rubberband but none gave satisfactory results, the first being too complicated and the latter yelding too much latency. It is very simple to use - a simple semitones shift and a quality lever - and sounds pretty nice. The team is working on it to make it even nicer but this first version sounds pretty good already. It uses the fftw3 lib which is pretty common and can be installed via the main linux repositories (ubuntu, debian, arch, etc) Hope you all enjoy Kind Regards Gianfranco The MOD Team -------------- next part -------------- An HTML attachment was scrubbed... URL: From cracauer at cons.org Mon Jul 29 18:07:20 2013 From: cracauer at cons.org (Martin Cracauer) Date: Mon, 29 Jul 2013 14:07:20 -0400 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> Message-ID: <20130729180720.GA19591@cons.org> Philipp ??berbacher wrote on Fri, Jul 26, 2013 at 03:26:21PM +0200: > I want to use this device to record interesting sounds wherever I am, > so it should be relatively small and and fast to start recording. I > also intend to record lots of quiet or relatively far away sources, I > want to record bird songs rather than guitars. As far as I've seen > those devices use SSDs or similar storage media, so I assume they don't > make any noise on their own, right? I guess the microphone should also > have a low self-noise. You can't use the Zoom H4n then. I really like mine but it has some weaknesses that will matter more to you, specially: - long boot time - deep and plain menu system, takes a while to start anything. Crappy UI hardware doesn't help because it gets in the way of muscle memory which usually helps with bad UIs - the maximum gain with the built in microphones is very low. You'll only get decent levels for things that are normal volume to you standing there. Everything that sound quiet to you will not be able to level out. No way to go after birds with it without using external microphones - automatic gain is overly dumb. Any loud impulse will set the gain lower, permanently. E.g. you record something and bump into the thing. Now your gain has been permanently lowered and won't go back up Of course I have no idea whether the alternatives are any better but I'm sure the boot times and menu issue are better in e.g. the Tascams. The Tascams have their own problems, reviews on Amazon are enlightning and they have no top end unit with all their features (top unit is one refresh cycle behind). For me the Zoom is very good. It works and I even use it as a USB soundcard in Linux and FreeBSD. Good sound quality, reliable, lots of options, correct input impedance for passive guitar/bass. Martin -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer http://www.cons.org/cracauer/ From cracauer at cons.org Mon Jul 29 18:13:29 2013 From: cracauer at cons.org (Martin Cracauer) Date: Mon, 29 Jul 2013 14:13:29 -0400 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130727091754.GA823@linuxaudio.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> Message-ID: <20130729181329.GB19591@cons.org> To get back to the original discussion - I feel very enlightened now and have to re-do everything I did in the last years. Thanks, guys, or something :-) I have a question left, and that is that from my observation lame is unable to predict the "correct" gain level. Lame gives you a db value that you can use as adjustment on the input so that you don't clip, or how many db you are under, respectively. I find that lowering the gain by what lame suggested in the first run will still randomly lead to clipping and lame going back suggesting even lower values. Is this an inherit problem that lame can't possibly do an exact job? I just wish lame had an option to just read the whole clip and noodle around until it gets to a level it's happy with. Martin -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer http://www.cons.org/cracauer/ From jh at brainiac.com Mon Jul 29 18:31:04 2013 From: jh at brainiac.com (Joe Hartley) Date: Mon, 29 Jul 2013 14:31:04 -0400 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> Message-ID: <20130729143104.782197f50b601f55708c219e@brainiac.com> On Mon, 29 Jul 2013 18:50:56 +0100 "Barney Holmes" wrote: > The Sony Minidisc (remember those?) was considered a desirable item for > live recording in its time. I'm currently experimenting with a Sony > Walkman Minidisc Net MD (Walkman MZ-N707 Type-R) that I picked up in a > charity shop for ~6 pounds. I had a Sony MD for a while and it made some good recordings, but I really got tired of the restrictions that meant I could not pull the recordings off the disk in anything but realtime. As far as I know, there's no way around that restriction. My understanding is that the unit lacks the ability to make an outward digital connection completely. Sony could have ruled the market, but they were so paranoid about cutting into their music division's sales that they crippled all but the most expensive "pro" MD decks because HOME TAPING IS KILLING MUSIC! (Remember that blast from the past?) Sony's remained paranoid about hardware that can make exact digital copies of media for a long time; their rootkit is another example of it. Those two things have kept me Sony-free ever since, except for a stereo mic that I got with the MD (an ECM-MS907) that consistently surprises me with excellent live recordings. After I got tired of the MD I got an MAudio Microtrack 2496, which is nice for really portable recording, though my older model uses a CF+ Microdrive for storage and it is showing its age. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From fons at linuxaudio.org Mon Jul 29 19:28:38 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 29 Jul 2013 19:28:38 +0000 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130729143104.782197f50b601f55708c219e@brainiac.com> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> Message-ID: <20130729192838.GA27373@linuxaudio.org> On Mon, Jul 29, 2013 at 02:31:04PM -0400, Joe Hartley wrote: > Those two things have kept me Sony-free ever since, except for a stereo > mic that I got with the MD (an ECM-MS907) that consistently surprises me > with excellent live recordings. Sony is one of those outfits I have very mixed feelings about. I still have their flagship short wave receiver ICF-2001D which by now is 30 years old and still working perfectly. One of the reasons for that is that I also got the service manual, from the same main street HiFi shop at the equivalent of 2 Euro or so. It has the complete circuit diagrams. Try to find that for any piece of equipment today. I also have a Minidisk, MZ-R70 which I've used as portable music player for a long time. A few years ago it dropped from the table, turning most of the display into a black blob, but otherwise it still works. I certainly prefer the quality over anything mp3. And yes, you could make digital copies of CDs with it, but any attempt to copy the output digitally has failed so far. They also made some very good speakers and microphones. I've had little experience with the mics, but I liked them every time I've used them. Sony became sort of evil when it turned from an (excellent) equipment manufacturer into a media multinational, with things like the CD rootkit and SACD as the result. 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 alice-dsl.net Mon Jul 29 19:41:27 2013 From: ralf.mardorf at alice-dsl.net (Ralf Mardorf) Date: Mon, 29 Jul 2013 21:41:27 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130729192838.GA27373@linuxaudio.org> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> Message-ID: <1375126887.675.188.camel@archlinux> On Mon, 2013-07-29 at 19:28 +0000, Fons Adriaensen wrote: > It has the complete circuit diagrams. Try to find that for > any piece of equipment today. For old German consumer HiFi gear and TVs it was included in the delivery, no need to pay the equivalent of even 2 EUR. Nowadays you have to pay more than 30 EUR for a rotten xerox by Schaltungsdienst Lange oHG. Throwaway society, it wouldn't make sense to include circuit diagrams, exploded diagram etc., most people can't repair their gear anymore and even if they would be able to do it, they anyway want something new every year. From fons at linuxaudio.org Mon Jul 29 19:56:16 2013 From: fons at linuxaudio.org (Fons Adriaensen) Date: Mon, 29 Jul 2013 19:56:16 +0000 Subject: [LAU] Gain and clipping wav -> lame In-Reply-To: <20130729181329.GB19591@cons.org> References: <20130725182150.GA97830@cons.org> <20130725194053.GA20598@linuxaudio.org> <20130725201420.GA58212@cons.org> <20130725164548.5592899f@HQ-3591-En.nfbonf.nfb.ca> <20130726223502.GA73801@cons.org> <20130727091754.GA823@linuxaudio.org> <20130729181329.GB19591@cons.org> Message-ID: <20130729195616.GB27373@linuxaudio.org> On Mon, Jul 29, 2013 at 02:13:29PM -0400, Martin Cracauer wrote: > I find that lowering the gain by what lame suggested in the first run > will still randomly lead to clipping and lame going back suggesting > even lower values. > > Is this an inherit problem that lame can't possibly do an exact job? Maybe. > I just wish lame had an option to just read the whole clip and noodle > around until it gets to a level it's happy with. You could probably avoid all these issues by not trying to get the maximum level in the first place. If you record/master at normal levels lame won't have any problems with your signals 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 djbarney at djbarney.org Mon Jul 29 20:26:56 2013 From: djbarney at djbarney.org (Barney Holmes) Date: Mon, 29 Jul 2013 21:26:56 +0100 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130729143104.782197f50b601f55708c219e@brainiac.com> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> Message-ID: <60b1dd33918f6034d3cd3bcf56ff86fb.squirrel@djbarney.org> Any recommendations for Minidisc players that can spit out the digital recording ? My local electronics shop still sells minidiscs :) Looked up the Audio Microtrack 2496 - 5 pounds on ebay, but 7 days left. djbarney On Mon, July 29, 2013 7:31 pm, Joe Hartley wrote: > On Mon, 29 Jul 2013 18:50:56 +0100 > "Barney Holmes" wrote: >> The Sony Minidisc (remember those?) was considered a desirable item for >> live recording in its time. I'm currently experimenting with a Sony >> Walkman Minidisc Net MD (Walkman MZ-N707 Type-R) that I picked up in a >> charity shop for ~6 pounds. > > I had a Sony MD for a while and it made some good recordings, but I really > got tired of the restrictions that meant I could not pull the recordings > off the disk in anything but realtime. As far as I know, there's no way > around that restriction. My understanding is that the unit lacks the > ability to make an outward digital connection completely. > > Sony could have ruled the market, but they were so paranoid about cutting > into their music division's sales that they crippled all but the most > expensive "pro" MD decks because HOME TAPING IS KILLING MUSIC! (Remember > that blast from the past?) > > Sony's remained paranoid about hardware that can make exact digital copies > of media for a long time; their rootkit is another example of it. > > Those two things have kept me Sony-free ever since, except for a stereo > mic that I got with the MD (an ECM-MS907) that consistently surprises me > with excellent live recordings. After I got tired of the MD I got an > MAudio Microtrack 2496, which is nice for really portable recording, > though > my older model uses a CF+ Microdrive for storage and it is showing its > age. > > > > -- > ====================================================================== > Joe Hartley - UNIX/network Consultant - jh at brainiac.com > Without deviation from the norm, "progress" is not possible. - FZappa > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > ~~~ Home site - http://djbarney.org From gnome at hawaii.rr.com Tue Jul 30 08:19:28 2013 From: gnome at hawaii.rr.com (david) Date: Mon, 29 Jul 2013 22:19:28 -1000 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130729192838.GA27373@linuxaudio.org> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> Message-ID: <51F77710.4030406@hawaii.rr.com> On 07/29/2013 09:28 AM, Fons Adriaensen wrote: > Sony became sort of evil when it turned from an (excellent) > equipment manufacturer into a media multinational, with > things like the CD rootkit and SACD as the result. The only part of Sony making money is its insurance division. The equipment and media divisions are both running losses. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From hollunder at lavabit.com Tue Jul 30 08:56:04 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Tue, 30 Jul 2013 10:56:04 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130729180720.GA19591@cons.org> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <20130729180720.GA19591@cons.org> Message-ID: <20130730105604.512da586@eeyore.mozart.uni-klu.ac.at> On Mon, 29 Jul 2013 14:07:20 -0400 Martin Cracauer wrote: > Philipp ??berbacher wrote on Fri, Jul 26, 2013 at 03:26:21PM +0200: > > I want to use this device to record interesting sounds wherever I > > am, so it should be relatively small and and fast to start > > recording. I also intend to record lots of quiet or relatively far > > away sources, I want to record bird songs rather than guitars. As > > far as I've seen those devices use SSDs or similar storage media, > > so I assume they don't make any noise on their own, right? I guess > > the microphone should also have a low self-noise. > > You can't use the Zoom H4n then. > > I really like mine but it has some weaknesses that will matter more to > you, specially: > - long boot time > - deep and plain menu system, takes a while to start anything. Crappy > UI hardware doesn't help because it gets in the way of muscle memory > which usually helps with bad UIs > - the maximum gain with the built in microphones is very low. You'll > only get decent levels for things that are normal volume to you > standing there. Everything that sound quiet to you will not be able > to level out. No way to go after birds with it without using > external microphones > - automatic gain is overly dumb. Any loud impulse will set the gain > lower, permanently. E.g. you record something and bump into the > thing. Now your gain has been permanently lowered and won't go back > up > > Of course I have no idea whether the alternatives are any better but > I'm sure the boot times and menu issue are better in e.g. the > Tascams. The Tascams have their own problems, reviews on Amazon are > enlightning and they have no top end unit with all their features (top > unit is one refresh cycle behind). > > For me the Zoom is very good. It works and I even use it as a USB > soundcard in Linux and FreeBSD. Good sound quality, reliable, lots of > options, correct input impedance for passive guitar/bass. > > Martin Thanks Martin, given what I read by you and others I wont buy a zoom. I'm not sure yet what it will be. I wanted to ask on the nature recordist mailing list, but membership is moderated and there has been no response yet. Guess I'll dig deeper into their mail archive. Regards, Philipp From hollunder at lavabit.com Tue Jul 30 09:06:08 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Tue, 30 Jul 2013 11:06:08 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> Message-ID: <20130730110608.3e40cd07@eeyore.mozart.uni-klu.ac.at> On Mon, 29 Jul 2013 18:50:56 +0100 "Barney Holmes" wrote: > Hi, > > The Sony Minidisc (remember those?) was considered a desirable item > for live recording in its time. I'm currently experimenting with a > Sony Walkman Minidisc Net MD (Walkman MZ-N707 Type-R) that I picked > up in a charity shop for ~6 pounds. Its probably a collectors item, > it was first introduced in 2002. But I'm curious if it lives up to > all the recommendations I heard at the time from live sound > recordists. > > Did some seaside and other recordings just using a basic mono > microphone. Sounds OK when I play back on headphones from the device. > I'm currently transferring to my PC. The problem with the Minidisc is > that it has too much DRM on it (which is one of the reasons it failed > as a technology). Fibre optic connection only works INTO the device. > Recordings cannot be transferred off the device in digital form and > have to be re-recorded through an analog output - but the quality > loss is negligible I think. Currently looking at "linux-minidisc" for > some possible ways around the DRM. > > Who knows what other second had devices are out there ? It's amazing > what you can find in charity shops as well ;) > > djbarney I had one of those, many years ago. I only used it as a music player, but I had to use some really buggy sony software to get any music on it, and it was in some obscure format. Not too long after the first mp3 players showed up and then the ipod hype started. That was pretty much the end of MD. I know there are some enthusiasts left, but I don't shed a tear. Regards, Philipp From jh at brainiac.com Tue Jul 30 12:30:32 2013 From: jh at brainiac.com (Joe Hartley) Date: Tue, 30 Jul 2013 08:30:32 -0400 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <51F77710.4030406@hawaii.rr.com> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> <51F77710.4030406@hawaii.rr.com> Message-ID: <20130730083032.53562d166332192f39309ec2@brainiac.com> On Mon, 29 Jul 2013 22:19:28 -1000 david wrote: > On 07/29/2013 09:28 AM, Fons Adriaensen wrote: > > > Sony became sort of evil when it turned from an (excellent) > > equipment manufacturer into a media multinational, with > > things like the CD rootkit and SACD as the result. > > The only part of Sony making money is its insurance division. The > equipment and media divisions are both running losses. "Ha ha!" - Nelson Muntz I have been thinking about other Sony gear I've had - one of the best monitors I ever had was a big Trinitron monitor on a Sun workstation. Absolutely stunning graphics back in the early 80s. It's too bad that their ham-fisted attempts at customer control (along with the standard music industry panic over losing the traditional distribution model) cause a lot of people to choose other companies to do business with. "I said, 'Ha ha!'" - Nelson Muntz -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From linux at alextone.info Tue Jul 30 15:01:04 2013 From: linux at alextone.info (Alex) Date: Tue, 30 Jul 2013 17:01:04 +0200 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130730083032.53562d166332192f39309ec2@brainiac.com> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> <51F77710.4030406@hawaii.rr.com> <20130730083032.53562d166332192f39309ec2@brainiac.com> Message-ID: <51F7D530.5080606@alextone.info> On 07/30/2013 02:30 PM, Joe Hartley wrote: > On Mon, 29 Jul 2013 22:19:28 -1000 > david wrote: > >> On 07/29/2013 09:28 AM, Fons Adriaensen wrote: >> >>> Sony became sort of evil when it turned from an (excellent) >>> equipment manufacturer into a media multinational, with >>> things like the CD rootkit and SACD as the result. >> The only part of Sony making money is its insurance division. The >> equipment and media divisions are both running losses. > "Ha ha!" > - Nelson Muntz > > I have been thinking about other Sony gear I've had - one of the best > monitors I ever had was a big Trinitron monitor on a Sun workstation. > Absolutely stunning graphics back in the early 80s. It's too bad that > their ham-fisted attempts at customer control (along with the standard > music industry panic over losing the traditional distribution model) > cause a lot of people to choose other companies to do business with. > > "I said, 'Ha ha!'" > - Nelson Muntz > Sony, the Monsanto of the media world? Alex. From wizardofgosz at gmail.com Tue Jul 30 15:22:21 2013 From: wizardofgosz at gmail.com (Ricardus Vincente) Date: Tue, 30 Jul 2013 11:22:21 -0400 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <51F7D530.5080606@alextone.info> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> <51F77710.4030406@hawaii.rr.com> <20130730083032.53562d166332192f39309ec2@brainiac.com> <51F7D530.5080606@alextone.info> Message-ID: <51F7DA2D.6060108@gmail.com> On 07/30/2013 11:01 AM, Alex wrote: >> I have been thinking about other Sony gear I've had - one of the best >> monitors I ever had was a big Trinitron monitor on a Sun workstation. >> Absolutely stunning graphics back in the early 80s. It's too bad that >> their ham-fisted attempts at customer control (along with the standard >> music industry panic over losing the traditional distribution model) >> cause a lot of people to choose other companies to do business with. >> >> "I said, 'Ha ha!'" >> - Nelson Muntz >> > Sony, the Monsanto of the media world? I had two of their 17" Trinitron monitors back in the day, and they were stunning to look at. One of my favorite outboard reverbs was the Sony DPS-R7. Sounded amazing. Was my go-to reverb in a studio I worked at in the late 90s. I see there is one for sale on Ebay... if I just had some spare cash! Their 3324 and 3348 digital decks were pretty cool back in the day. The Sony Oxford was an amazing large format digital desk, and even the smaller DMX-R100 was pretty cool. Anyway, at one time Sony was cool and had an amazing Pro Audio heritage. Now they make a gaming console. Welcome to the new world! Rich... From kvutter at frii.com Wed Jul 31 03:30:50 2013 From: kvutter at frii.com (Kevin Utter) Date: Tue, 30 Jul 2013 21:30:50 -0600 Subject: [LAU] Jack.Plumbing Message-ID: Hi all! I wanted to use Jack.Plumbing with some connect-exclusive rules for when certain apps were up. But it seems that connect-exclusive rules can create a loop effect where connections are made and remade unnecessarially. Is something wrong with my Jack.plumbing, or am I not understanding something? It seems to be a problem even if I don't have other rules that would go against the exclusive ones, so that just using an exclusive rule at all results in this looping. Those of you using Jack.Plumbing, are you constantly re-writing the rules, or can't you use exclusive rules to change the setup when other apps run without manually removing all other rules? Thanks much. Kevin From gnome at hawaii.rr.com Wed Jul 31 06:11:53 2013 From: gnome at hawaii.rr.com (david) Date: Tue, 30 Jul 2013 20:11:53 -1000 Subject: [LAU] Looking for advice on a field recording device In-Reply-To: <20130730083032.53562d166332192f39309ec2@brainiac.com> References: <20130726152621.5a4b65f8@eeyore.mozart.uni-klu.ac.at> <37625f97eba1e2d108adc5b461c8a3d2.squirrel@djbarney.org> <20130729143104.782197f50b601f55708c219e@brainiac.com> <20130729192838.GA27373@linuxaudio.org> <51F77710.4030406@hawaii.rr.com> <20130730083032.53562d166332192f39309ec2@brainiac.com> Message-ID: <51F8AAA9.605@hawaii.rr.com> On 07/30/2013 02:30 AM, Joe Hartley wrote: > On Mon, 29 Jul 2013 22:19:28 -1000 > david wrote: > >> On 07/29/2013 09:28 AM, Fons Adriaensen wrote: >> >>> Sony became sort of evil when it turned from an (excellent) >>> equipment manufacturer into a media multinational, with >>> things like the CD rootkit and SACD as the result. >> >> The only part of Sony making money is its insurance division. The >> equipment and media divisions are both running losses. > > "Ha ha!" > - Nelson Muntz > > I have been thinking about other Sony gear I've had - one of the best > monitors I ever had was a big Trinitron monitor on a Sun workstation. > Absolutely stunning graphics back in the early 80s. It's too bad that > their ham-fisted attempts at customer control (along with the standard > music industry panic over losing the traditional distribution model) > cause a lot of people to choose other companies to do business with. > > "I said, 'Ha ha!'" > - Nelson Muntz I have a couple pieces of Sony consumer stuff around here - a small boombox with decent sound quality and stereo line in, and an alarm clock. They've outlasted many other such non-Sony stuff I've had over the years. -- David gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com http://clanjones.org/david/ http://dancing-treefrog.deviantart.com/ From julien at mail.upb.de Wed Jul 31 07:48:01 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 09:48:01 +0200 (CEST) Subject: [LAU] [New music]: Some eurodance - Be Frank Message-ID: Hello everyone! Here's your dance delivery. It's totally free and I don't even want a tip. :-) Before we get to the download links, two things, that are important to me. Alexandre Ratchov: your Midish is a fantastic piece of software! With some of my own "user defined commands" it was a piece of cake to do this! So, if you haven't heard of Midish, take a look and don't forget to RTFM, because it's very well-written and complete! Second: Joel, you know, tat Nama is cool, but the new Hotkeys mode really rocks! And here are the links: http://juliencoder.de/nama/be_frank.ogg http://juliencoder.de/nama/be_frank.mp3 Or as ever from the website: http://juliencoder.de/nama/music.html This piece wouldn't have been possible without a MIDI sequencer with quantisation, cut/copy&paste and all the other tools of trade such programs bring into the equation. There were no software synthesizers used in this song, next time perhaps. :-) The main piece of equipment is a Roland JV-1080, so it's authentic at least. :-) I guess with such music, there are only two ways about it: love it or leave it. :-) Which one is it for you? Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From gabbe.nord at gmail.com Wed Jul 31 08:18:54 2013 From: gabbe.nord at gmail.com (Gabbe Nord) Date: Wed, 31 Jul 2013 10:18:54 +0200 Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: Haha, maan that brought me back to my childhood! It reminds me so much of the first song I remember really really liking, which was "Get Freaky" with Music Instructor. Pretty different to what I've heard from you earlier, and while it's not something I'd listen to normally, I certainly do appreciate it. There's something about the combination of nostalgia and a pretty smooth beat that makes me feel care free. Really cool Julien, keep it coming! /Gabriel On Wed, Jul 31, 2013 at 9:48 AM, Julien Claassen wrote: > Hello everyone! > Here's your dance delivery. It's totally free and I don't even want a > tip. :-) > Before we get to the download links, two things, that are important to > me. > Alexandre Ratchov: your Midish is a fantastic piece of software! With > some of my own "user defined commands" it was a piece of cake to do this! > So, if you haven't heard of Midish, take a look and don't forget to RTFM, > because it's very well-written and complete! > Second: Joel, you know, tat Nama is cool, but the new Hotkeys mode > really rocks! > And here are the links: > http://juliencoder.de/nama/be_**frank.ogg > http://juliencoder.de/nama/be_**frank.mp3 > Or as ever from the website: > http://juliencoder.de/nama/**music.html > This piece wouldn't have been possible without a MIDI sequencer with > quantisation, cut/copy&paste and all the other tools of trade such programs > bring into the equation. There were no software synthesizers used in this > song, next time perhaps. :-) The main piece of equipment is a Roland > JV-1080, so it's authentic at least. :-) > I guess with such music, there are only two ways about it: love it or > leave it. :-) Which one is it for you? > Warm regards > Julien > > ------------------------------**---------- > http://juliencoder.de/nama/**music.html > ______________________________**_________________ > 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 julien at mail.upb.de Wed Jul 31 09:58:51 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 11:58:51 +0200 (CEST) Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: Hello Gabriel! Thanks a lot for these kind words. If it takes you back there, I reached my goal. :-) If be freaky is one of your childhood tunes, then you must obviously be a little younger than me. Well: the young to the power and the top! :-) Not dancingly yours Julien ---------------------------------------- http://juliencoder.de/nama/music.html From allcoms at gmail.com Wed Jul 31 10:17:21 2013 From: allcoms at gmail.com (Dan MacDonald) Date: Wed, 31 Jul 2013 11:17:21 +0100 Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: Is it a Technotronic vs MARRS revival collaboration or LA's / NAMA's most prolific user having some fun? Not really the sort of music I'd listen to out of choice any more if I'm honest but I'm always interested to hear what you've been up to Julien. You never know what to expect with your recordings and you seem to be able to knock out a good tune in any genre. Not tried MIDISH myself yet, will have to add that to my overly long list of stuff I should check out. Thanks for sharing Julien! On Wed, Jul 31, 2013 at 8:48 AM, Julien Claassen wrote: > Hello everyone! > Here's your dance delivery. It's totally free and I don't even want a > tip. :-) > Before we get to the download links, two things, that are important to > me. > Alexandre Ratchov: your Midish is a fantastic piece of software! With > some of my own "user defined commands" it was a piece of cake to do this! > So, if you haven't heard of Midish, take a look and don't forget to RTFM, > because it's very well-written and complete! > Second: Joel, you know, tat Nama is cool, but the new Hotkeys mode > really rocks! > And here are the links: > http://juliencoder.de/nama/be_**frank.ogg > http://juliencoder.de/nama/be_**frank.mp3 > Or as ever from the website: > http://juliencoder.de/nama/**music.html > This piece wouldn't have been possible without a MIDI sequencer with > quantisation, cut/copy&paste and all the other tools of trade such programs > bring into the equation. There were no software synthesizers used in this > song, next time perhaps. :-) The main piece of equipment is a Roland > JV-1080, so it's authentic at least. :-) > I guess with such music, there are only two ways about it: love it or > leave it. :-) Which one is it for you? > Warm regards > Julien > > ------------------------------**---------- > http://juliencoder.de/nama/**music.html > ______________________________**_________________ > 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 julien at mail.upb.de Wed Jul 31 10:25:52 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 12:25:52 +0200 (CEST) Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: Hello Dan! Haha! No it's just me having the time of my life! :-) It was tremendous fun. Just to experience that way of working, recording a few snipets, using loops, stacking tracks and creating "interest" by adding or removing bits of the arrangement. I think now, taht this is done, I might move on to my slightly dubstep influenced dream track. :-) Warmly yours Julien ---------------------------------------- http://juliencoder.de/nama/music.html From hollunder at lavabit.com Wed Jul 31 11:23:56 2013 From: hollunder at lavabit.com (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Wed, 31 Jul 2013 13:23:56 +0200 Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: <20130731132356.44fa1358@eeyore.mozart.uni-klu.ac.at> On Wed, 31 Jul 2013 09:48:01 +0200 (CEST) Julien Claassen wrote: > Hello everyone! > Here's your dance delivery. It's totally free and I don't even > want a tip. :-) > Before we get to the download links, two things, that are > important to me. Alexandre Ratchov: your Midish is a fantastic piece > of software! With some of my own "user defined commands" it was a > piece of cake to do this! So, if you haven't heard of Midish, take a > look and don't forget to RTFM, because it's very well-written and > complete! Second: Joel, you know, tat Nama is cool, but the new > Hotkeys mode really rocks! > And here are the links: > http://juliencoder.de/nama/be_frank.ogg > http://juliencoder.de/nama/be_frank.mp3 > Or as ever from the website: > http://juliencoder.de/nama/music.html > This piece wouldn't have been possible without a MIDI sequencer > with quantisation, cut/copy&paste and all the other tools of trade > such programs bring into the equation. There were no software > synthesizers used in this song, next time perhaps. :-) The main piece > of equipment is a Roland JV-1080, so it's authentic at least. :-) > I guess with such music, there are only two ways about it: love it > or leave it. :-) Which one is it for you? > Warm regards > Julien Cool track, thanks Julien. It reminds me of some 90s music I like, your song is just a bit busier. I like it :) Regards, Philipp From alf at mellomrommet.no Wed Jul 31 12:56:55 2013 From: alf at mellomrommet.no (Alf Haakon Lund) Date: Wed, 31 Jul 2013 14:56:55 +0200 Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: <51F90997.4030801@mellomrommet.no> On 31. juli 2013 09:48, Julien Claassen wrote: > Hello everyone! > Here's your dance delivery. It's totally free and I don't even want a > tip. :-) > Before we get to the download links, two things, that are important > to me. > Alexandre Ratchov: your Midish is a fantastic piece of software! With > some of my own "user defined commands" it was a piece of cake to do > this! So, if you haven't heard of Midish, take a look and don't forget > to RTFM, because it's very well-written and complete! > Second: Joel, you know, tat Nama is cool, but the new Hotkeys mode > really rocks! > And here are the links: > http://juliencoder.de/nama/be_frank.ogg > http://juliencoder.de/nama/be_frank.mp3 > Or as ever from the website: > http://juliencoder.de/nama/music.html > This piece wouldn't have been possible without a MIDI sequencer with > quantisation, cut/copy&paste and all the other tools of trade such > programs bring into the equation. There were no software synthesizers > used in this song, next time perhaps. :-) The main piece of equipment is > a Roland JV-1080, so it's authentic at least. :-) > I guess with such music, there are only two ways about it: love it or > leave it. :-) Which one is it for you? > Warm regards > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html Well, It sounds like the 90's are back, albeit with a touch of the 80's too. When this kind of music was in fashion I really couldn't stand it, but those extra years must have softened me a bit. Now, there's a touch of nostalgia... As for the track I get the feeling it was made more for the fun of it than this actually being the "new you". It's catchy and well put together, but with too little progression and dynamics to make me listen multiple times. Thanks for sharing! Alf From julien at mail.upb.de Wed Jul 31 13:11:24 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 15:11:24 +0200 (CEST) Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: <51F90997.4030801@mellomrommet.no> References: <51F90997.4030801@mellomrommet.no> Message-ID: Hello Alf! No this isn't the new me.It's an old me peaking out and having a day on the beach. :-) But it was also an outing for my new and partly old acquaintances being once more with me. I had to go a while without a few of my sounds and others were completely new here in my synth gathering. :-) Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From cave.dnb2m97pp at aliceadsl.fr Wed Jul 31 14:52:31 2013 From: cave.dnb2m97pp at aliceadsl.fr (Nigel Henry) Date: Wed, 31 Jul 2013 16:52:31 +0200 Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: References: Message-ID: <201307311652.31221.cave.dnb2m97pp@aliceadsl.fr> On Wednesday 31 July 2013 09:48, Julien Claassen wrote: Hi Julien, Now that really is my kind of music, great stuff. Now I'll have to get the decks back out and get mixing again. Nigel. > Hello everyone! > Here's your dance delivery. It's totally free and I don't even want a > tip. > > :-) > > Before we get to the download links, two things, that are important to > me. Alexandre Ratchov: your Midish is a fantastic piece of software! With > some of my own "user defined commands" it was a piece of cake to do this! > So, if you haven't heard of Midish, take a look and don't forget to RTFM, > because it's very well-written and complete! > Second: Joel, you know, tat Nama is cool, but the new Hotkeys mode > really rocks! > And here are the links: > http://juliencoder.de/nama/be_frank.ogg > http://juliencoder.de/nama/be_frank.mp3 > Or as ever from the website: > http://juliencoder.de/nama/music.html > This piece wouldn't have been possible without a MIDI sequencer with > quantisation, cut/copy&paste and all the other tools of trade such programs > bring into the equation. There were no software synthesizers used in this > song, next time perhaps. :-) The main piece of equipment is a Roland > JV-1080, so it's authentic at least. :-) > I guess with such music, there are only two ways about it: love it or > leave it. :-) Which one is it for you? > Warm regards > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From julien at mail.upb.de Wed Jul 31 14:57:31 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 16:57:31 +0200 (CEST) Subject: [LAU] [New music]: Some eurodance - Be Frank In-Reply-To: <201307311652.31221.cave.dnb2m97pp@aliceadsl.fr> References: <201307311652.31221.cave.dnb2m97pp@aliceadsl.fr> Message-ID: Hello you two! Ha, so that was the secret. Making catchy, flashy electronic music. :-) It also shows me - and hopefully others-, that there is a good text-based way even for that genre. and hopefully it might attract a few more to Nama and Midish. Advertisingly yours Julien ---------------------------------------- http://juliencoder.de/nama/music.html From axeldenstore at gmail.com Wed Jul 31 15:58:01 2013 From: axeldenstore at gmail.com (alexander) Date: Wed, 31 Jul 2013 18:58:01 +0300 Subject: [LAU] two new tracks Message-ID: <51F93409.1000904@gmail.com> They are actually not that new, It's some 95% finished stuff I found when I cleaned my ardour sessions folder and figured I might just finish and release it.. The first track is simply based around a drum improv track I did when I got my new drums last summer.. Second one is a slow, ambient, Gybe inspired piece that I don't even remember when I started, ancient stuff... http://vortexofinfiniteirony.bandcamp.com/ I think I'll post later stuff under this new name instead of sharpattack, I want a name that at least used to give zero hits on google.. (It's alot easier to stalk yourself then, ie feed your ego) I made the cover with mypaint, a really satisfying experience. Because usually I really suck at painting, I mean REEALLY suck.. but I'm very happy how the cover turned out. Didn't take all that long either, maby abit over an hour or so... From julien at mail.upb.de Wed Jul 31 19:09:50 2013 From: julien at mail.upb.de (Julien Claassen) Date: Wed, 31 Jul 2013 21:09:50 +0200 (CEST) Subject: [LAU] two new tracks In-Reply-To: <51F93409.1000904@gmail.com> References: <51F93409.1000904@gmail.com> Message-ID: Hello Alexander! Before I start spreading my opinions through this aether, there's only one point: the two songs I have, appear to be the same. Is that meant to be? Perhaps different mixes? I haven't compared them that minutely yet. It's not, what I would listen to all day or regularly buy at the shop, but that only makes it more interesting to the habitual ear. :-) the first part reminded me of a crime play from Sweden. OK, it's German translation. but the slightly menacing, dark atmosphere. Slow, oppressive and somewhat mysterious. Aided by the radio sample/effect. The dragging rhythm really helps this impression. I would have liked your drumkit even more, if you had dampened the snare, until it was dead as a door nail. :-) That's my bad 70s influence, not to be confused with my good 70s influence. :-) the second song has a much lighter tone. Still dusky or better night-timely. Especially playing it with the rimshot at the beginning, opens it up. You have the feeling, that you can breathe more easily. My association for this piece would be more towards Raymond Chandler than Marie Jungsted. - Ah, stupid me, I just had the revelation, that I have both pieces only in one file. :-) And I thought it was a two part piece. :-) It' wasn't a glitch in either aether or radio, but in my brain. :-) And mix and processing of both tunes is nicely done. Unobtrusive, but not colourless. The mix itself sounds warm and open, it is alive, able to breathe and convey, that there was a human being playing living instruments. Were they all real, acoustic instruments? To my ear they sound like it. If so, did you play them all yourself? Again if so: double chapot! For two very impressive - or expressive - pieces and good playing skills. Certainly those two pieces swept me away in their atmosphere much more than the trace of events. I also like the mix better, when comparing those two or three. They do set a much darker scene than Surfing the Waves of Creation and they moe farther away from everyday jazz harmonies. At least I think so. Not, that I'm a jazz expert... These two have personality. Thanks a lot for sharing and keep us posted with any other odd projects, that you just happen to find and finish. :-) Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html From lists at quirq.net Wed Jul 31 20:11:29 2013 From: lists at quirq.net (Q) Date: Wed, 31 Jul 2013 21:11:29 +0100 Subject: [LAU] two new tracks In-Reply-To: <51F93409.1000904@gmail.com> References: <51F93409.1000904@gmail.com> Message-ID: <51F96F71.8090008@quirq.net> On 31/07/13 16:58, alexander wrote: > They are actually not that new, It's some 95% finished stuff I found > when I cleaned my ardour sessions folder and figured I might just finish > and release it.. > > The first track is simply based around a drum improv track I did when I > got my new drums last summer.. > Second one is a slow, ambient, Gybe inspired piece that I don't even > remember when I started, ancient stuff... > > > http://vortexofinfiniteirony.bandcamp.com/ > > I think I'll post later stuff under this new name instead of > sharpattack, I want a name that at least used to give zero hits on > google.. (It's alot easier to stalk yourself then, ie feed your ego) > > I made the cover with mypaint, a really satisfying experience. Because > usually I really suck at painting, I mean REEALLY suck.. but I'm very > happy how the cover turned out. Didn't take all that long either, maby > abit over an hour or so... Just listening to Glitch. Cool. Great late night stuff, not that it currently is here, but nevermind :-) It reminds me of the bits of Univers Zero I've heard, early Anekdoten, Goblin and Morte Macabre (half of Anekdoten plus others doing covers of Goblin and other horror soundtrack music). The radio brought to mind the end of MM's cover of Fabio Frizzi's Apoteosi Del Mistero from one of the Living Dead films, with its snippet of (I think Russian) radio picked up by mistake on someone's amp. The mix is quite nice, but a bit muddy for my taste tending towards being a bit dull in places. There were one or two points where I think the drums were overpowering some of the other instruments just a little. Ocean In The Radio is not the sort of thing I'd normally listen to, a bit too ambient and lacking in structure for my liking. But it sounded good and very well executed. The mix was great. Thanks for sharing Q From axeldenstore at gmail.com Wed Jul 31 21:01:29 2013 From: axeldenstore at gmail.com (alexander) Date: Thu, 01 Aug 2013 00:01:29 +0300 Subject: [LAU] two new tracks In-Reply-To: References: <51F93409.1000904@gmail.com> Message-ID: <51F97B29.2050109@gmail.com> On 31/07/13 22:09, Julien Claassen wrote: > Hello Alexander! > Before I start spreading my opinions through this aether, there's > only one point: the two songs I have, appear to be the same. Is that > meant to be? Perhaps different mixes? I haven't compared them that > minutely yet. > It's not, what I would listen to all day or regularly buy at the > shop, but that only makes it more interesting to the habitual ear. :-) > the first part reminded me of a crime play from Sweden. OK, it's > German translation. but the slightly menacing, dark atmosphere. Slow, > oppressive and somewhat mysterious. Aided by the radio sample/effect. > The dragging rhythm really helps this impression. I would have liked > your drumkit even more, if you had dampened the snare, until it was > dead as a door nail. :-) That's my bad 70s influence, not to be > confused with my good 70s influence. :-) > the second song has a much lighter tone. Still dusky or better > night-timely. Especially playing it with the rimshot at the beginning, > opens it up. You have the feeling, that you can breathe more easily. > My association for this piece would be more towards Raymond Chandler > than Marie Jungsted. - Ah, stupid me, I just had the revelation, that > I have both pieces only in one file. :-) And I thought it was a two > part piece. :-) It' wasn't a glitch in either aether or radio, but in > my brain. :-) > And mix and processing of both tunes is nicely done. Unobtrusive, > but not colourless. The mix itself sounds warm and open, it is alive, > able to breathe and convey, that there was a human being playing > living instruments. Were they all real, acoustic instruments? To my > ear they sound like it. If so, did you play them all yourself? Again > if so: double chapot! For two very impressive - or expressive - pieces > and good playing skills. Certainly those two pieces swept me away in > their atmosphere much more than the trace of events. I also like the > mix better, when comparing those two or three. They do set a much > darker scene than Surfing the Waves of Creation and they moe farther > away from everyday jazz harmonies. At least I think so. Not, that I'm > a jazz expert... These two have personality. > Thanks a lot for sharing and keep us posted with any other odd > projects, that you just happen to find and finish. :-) > Warm regards > Julien > > ---------------------------------------- > http://juliencoder.de/nama/music.html > Oh sorry Julien, I mixed up the download links, I decided to change the name of one of the songs ;P dunno why there's a complete file of the other 'name' tho.. Here they are again, if anyone else don't/can't/won't use flash/java for bandcamp. https://dl.dropboxusercontent.com/u/16547648/Glitch%20In%20The%20Aether.flac https://dl.dropboxusercontent.com/u/16547648/Ocean%20In%20The%20Radio.flac And yes as allways every instrumment is "real" and played, except some of the effects.. I use yoshimi for the distorted sound(second half of the song) in 'ocean in the radio' and for the "glitchy" sound in glitch.. the rest of the effects are various cymbal effects, bowls and stuff.. For instance; tapping a cymbal softly then moving a mic so it gets within a centimeter of the cymbal produces some interesting sounds.. The radio is a manual seek AM reviever that I played around with.. The ringy sound in the beginning and first half of 'Ocean' is a minute long cymbal effect track that I loaded into lv2-ir convolution plug and ran the piano through it..