microbit http://www.wired.co.uk/news/archive/2015-09/18/bbc-micro-bit-delayed http://www.wired.co.uk/magazine/archive/2015/10/features/bbc-microbit-inside-story rtprio http://tedfelix.com/linux/linux-midi.html systemd http://itpro.nikkeibp.co.jp/article/COLUMN/20140708/569925/ inkscape http://wazly.blog87.fc2.com/blog-entry-55.html http://yaxu.org/how-to-publish-open-access-conference-proceedings/ endless https://endlessm.com/ alda https://github.com/alda-lang/alda === uname -a === aplay -l arecord -f cd -d 15 myrecording.wav === machine waiting(pausin) til calc get done can create some grooves .. then machines exchanging data via the net will be more important!! =jam clockless soundcard is what i want? http://www.eetimes.com/news/latest/showArticle.jhtml?articleID=179101800 === Watch out, that binary only drivers as the nvidia modules are probably incompatible with -rt kernels. Use the X11 nv driver instead. Even if the binary only module was compatible, it is not guaranteed that it fits into the model of the -rt kernel well and thus might destroy the low latency guarantees we get from the rest of the kernel. In a default -rt kernel all IRQ handler threads have default priorities in the range of 40-60, so making jackd run with a priority of 70 is a good choice here: jackd -R -P 70 -d alsa … But we also want the soundcard IRQ handler run at a higher prio than jack, as the soundcard irq is what ultimately drives jackd. Thus we can use the chrt utility to change this (my soundcard is on IRQ 4 in my system - make sure you check in /proc/interrupts what’s yours): chrt -f -p 82 `pidof “IRQ-4″` A priority of 82 is well above jackd. jackd also runs threads besides its main loop. Namely a watchdog at priority 10 above its main loop (80 in this example), and all clients’ process() threads will run with a priority 1 smaller than the main loop (69). So this has the audio part of the system covered. With this kind of settings i can achieve ridiculously low buffer sizes with jackd and my M-Audio delta 66 (even down to 8 samples/period though context switching overhead gets a bit costly here) :) RTC, The realtime clock every pc compatible has. This timer runs at a programmable frequency (i.e. 1024 hz) and thus enables the application to get run very often very regularly. To get RTC based timing to work nicely the rtc IRQ handler (which is # 8 on an XT-PIC system. Check /proc/interrupts to find out its IRQ) needs to get a priority above jackd and all the rest, so 98 is a good choice: chrt -f -p 98 `pidof “IRQ-8″` And if RTC is not used, this setting doesn’t do any harm either. IF you want ordinary users to be able to make use of the RTC at reasonably high frequencies, make sure that /proc/sys/dev/rtc/max-user-freq is set to i.e. 8192. You can do that by executing (as root): echo 8192 > /proc/sys/dev/rtc/max-user-freq To inspect what its current value is, use cat /proc/sys/dev/rtc/max-user-freq Summarized priority setup 99 System timer IRQ 98 RTC IRQ 95 . . Midi threads of softsynths/midi sequencers . 85 - 82 Soundcard IRQ 80 Jackd watchdog thread 70 Jackd main loop 69 Jackd client (softsynths/midi sequencers) audio process callbacks 60 . . Other IRQ handlers (disk, network, USB, GFX) . 40 http://tapas.affenbande.org/wordpress/?page_id=40 /proc/sys/dev/rtc/max-user-freq > [Jackit-devel] JACK 0.109.2 released > > This is a maintainance release of JACK. Three important issues are > fixed: > > * the old ALSA PCM port names are available, so that applications > that either hardcode or have existing patching information > containing these names will function. === Don't run alsaconf on 2.6 systems. The sounddevice is configured by udev. Try running # /etc/init.d/udev restart && udevtrigger === If mplayer ca play it, this might work: mplayer -ao pcm input.wma -ao pcm:file=output.wav > Use mplayer, that's what I usually do and it works fine: > mplayer -ao pcm:file=output.wav input.wma A little hint: You can use "fast" as an additional audio-out option in "-ao pcm:fast:file=output.wav" as I wrote in my other reply. This will dump faster than realtime, which is, well, faster. ====================================================== "preemptive kernel" and "realtime" which are different stuff. AFAIK you can try going realtime without a preemptive kernel (this is, a kernel without Ingo Molnar patches). Going preemptive: Download a vanilla kernel sources, version according to latest available Ingo Molnar patches. Apply patches, recompile, install, reboot. [1]This article explains nicely what preemption is all about. 1. http://lwn.net/Articles/146861/ If running a debianized distro things are far easier, simply download a [2]Musix kernel or a [3]64 Studio kernel (I'm right now running Debian sid with a Musix 2.6.20 kernel and everything's smooth). 2. http://linux.ilmainen.net/musix/ 3. http://archive.64studio.com/64studio/testing/pool/main/l/linux-2.6/ Talking about realtime, the newest the kernel the better although other things break. Going realtime is a matter of running a fairly recent version of libpam-modules (higher than 0.79 I think), making sure the user is a member of the audio group, putting this stanza into /etc/security/limits.conf and relogging. # limit realtime and memory locking access to users in the group audio # there is no way to say "allow locking all memory", 4G should be enough # * - rt_priority 0 * - nice 0 # @audio - rtprio 99 @audio - memlock 250000 @audio - nice -10 You should then be able to run jackd realtime. : [ ~ ] $; jackd -R -d alsa Also it should be useful enabling the high resolution timer in your /etc/sysctl.conf as this: dev.rtc.max-user-freq=1024 And then sysctl -p as root. === Just another realtime kernel from Tapani R=C3=A4ikk=C3=B6nen for Musix and = SuomiKnoppix: It has a new scheduler. To install it: sudo apt-get update sudo apt-get install linux-image-2.6.22.1-rt2 alsa-modules-2.6.22.1-rt2 Repository: # /etc/apt/sources.list Musix GNU+Linux deb ftp://musix.ourproject.org/pub/musix/deb/ ./ ==== tmpfs 1014M 0 1014M 0% /dev/shm You have to check if the tmpdir exists and if it's a tmpfs (sort of fs directly in ram). Jackd will work with its tmp on regular fs but you'll have better perf with ramfs. ------------------------------------------------------------------------- Subject: Re: [sc-users] terminated scons @ Slackintosh11.0 was Re:svn thing didn't work on UbuntuPPC7.04 > and somebody told me to > try running jack in 16Bit non-Realtime-mode > which is i don't know how to > gee ... will have to sep back and study before running any more commands I also told you to try to run jack with lesser frames/period, as i see you still try to run jack with 1024. If you find a value where jack runs without realtime mode, it is most likely that it will also run in realtime mode, but this is not always the case, so you will have to try several settings. What works for me is: jackd -dalsa -dhw:0 -r44100 -p256 -n6 -S -R where -R stands for Realtime, and what is most important -p256 stands for 256 frames instead of the default -p1024, which isn't working in my case. Imho your problem is related to jack, and not to SC. === James Warden wrote: > BUT! something in my setup makes jackd spit out an xrun every 10mns, > precisely 10mns. OK, sometimes, not 10mns but 20 or 50. But always > multiples of 10mns almost down to the msec! I have no clue how to start > tracing this down. It's not really annoying but I have no other xruns > otherwise, and that's really the beautiful part of it :) So is there a real audible xrun or just a message about possible one? Please check what sort of clock source your kernel is using. I assume you are using default clock source with jackd? What is your jack version? You can look for these kind of messages from dmesg: hpet0: at MMIO 0xfefff000 (virtual 0xf8800000), IRQs 2, 8, 31 hpet0: 3 32-bit timers, 25000000 Hz Using HPET for base-timer Using HPET for gettimeofday Using hpet for high-res timesource === James Warden wrote: > After googling around, it seems that it is related to whether you use > /dev/rtc or /dev/hpet. The jackd option "-c h" is not known by qjackctl > (I could not find it). I am trying to see if that makes a difference. I > will let jackd run for a few hours. Jack never uses /dev/rtc, but it can use memory mapped /dev/hpet. When using qjackctl, you can specify clock source parameters in the command line combobox. However, with latest kernels, good option is to try to check that kernel is having sane clock source for it's time keeping and to let jack use monotonic system clock (which is the default for reasonably new versions). > Mobo: DG965SS This should have HPET available and correctly reported by ACPI. > Distro : Ubuntu gutsy, realtime kernel 2.6.22-rt Check that you have, ACPI, APIC (local && IO), HPET and high resolution timer support enabled in kernel config. Which version of jack you were using? ==== Subject: Re: bash vs. python scripts - which one is better? >>>>> "Jeff" == Jeff D writes: > You would still have rename the file extention: > for FILE in *wav ; do lame -h -b 160 "$FILE" "`echo $FILE > |sed s/.wav/.mp3/g ` " ; done Or just use the shell itself: for FILE in *wav ; do lame -h -b 160 \"$FILE\" \"${FILE%.*}.mp3\"; done > >> for FILE in *.wav; do lame -h -b 160 "$FILE" "$FILE.mp3"; done > > Correct me if I'm wrong but wouldn't I just end up with with a > bunch of > files named blahblah.wav.mp3? Follow it with "rename .wav.mp3 .mp3 *". :) ==== Learn enough to be able to parse it and convert it to your language of choice. There are exceptions, of course, but by and large it has been my experience that any place you're required to use shell Python is also available and a far better choice. As an example here is the shell example given earlier: for FILE in *.wav; do lame -h -b 160 "$FILE" "$FILE.mp3"; done The same in Python but with far greater functionality: import os for file in os.listdir('.'): root, ext = os.path.splitext(file) if ext.lower() == 'wav': mp3 = root + '.mp3' result = os.system("lame -h -b 160 '%s' '%s'" % (file, mp3)) if result: print '%s not converted' % file Longer, yes. Easier to follow? Most certainly. Superior, no doubt. The shell example would miss WAV, Wav, wAv, etc. Secondly the only place we need to escape the variable is when we need shell to do some work, namely the call to lame. Finally we don't end up with '.wav.mp3' files all over the place. We can check the results easily and handle failures gracefully. Can all of that be done in shell? Certainly. Is it worth doing in shell? Not hardly. === Subject: Re: Conversion of .flv to .mpg format? On 8/9/07, ISHWAR RATTAN wrote: > Is there a way to go form .flv file to .mpg > under linux? > Use mencoder. A basic example: mencoder your.flv -oac copy -ovc lavc -of mpeg -lavcopts vcodec=mpeg1video -o your.mpg On 08/09/07 13:56, ISHWAR RATTAN wrote: > Is there a way to go form .flv file to .mpg > under linux? Check out clive and mpgtx in the repository. > Is there a way to go form .flv file to .mpg > under linux? > Assuming that you have mencoder, the script I use prior to converting from anyvideo.* to *.mpg prior to converting to DVD format is as follows: /usr/bin/mencoder -oac lavc -ovc lavc -of mpeg -mpegopts format=dvd -vf scale=720:576,harddup \ -srate 48000 -af lavcresample=48000 \ -lavcopts vcodec=mpeg2video:vrc_buf_size=1835:vrc_maxrate=9800:vbitrate=5000:keyint=15:aspect=4/3:\ acodec=ac3:abitrate=192 -ofps 25 -o original.flv newvideo.mpg Don't be too hard on yourself. mencoder's command line syntax is almost as confusing as sox's. ;) Subject: Re: flac and wav When ripping cassette tapes, I do: - use sox's "rec" to record a wav file of the whole side of a tape. - open the wav in `audacity' to visually find the spots that separate one song from another, writing down the second at which they occur. I generally check the timestamps I write down by comparing them to the "official" duration of each song. - run `wavsplit' passing it the timestamps I just wrote down. - rename the resulting wav files (so the name reflects the title, tracknumber, ...). - pass them through a `for' loop that compresses them with oggenc. === > I don't know of a way to do it (and I doubt it) but closing the > firefox tab should kick flash, also using the flashblock extension and > reloading the page works. One way to do it is to set up ALSA to use the Jackd plugin as an output device: http://alsa.opensrc.org/index.php/Jack_(plugin) === A frame of audio means 1 sample per channel.. E.g. if you have a 6-channel signal and you process one frame of it you have processed one sample per channel, 6 samples in total.. frames/period means frames per processing cycle (e.g. if you have 64 frames/period, then the buffersize per channel is 64 samples) periods/buffer means how many periods are there [on windows a period is more often called a "buffer"]. But in a typical double buffer scheme you have e.g. two buffers combined into a single chunk of memory.. And in alsa speak, this combined buffer is the "buffer" and the individual chunks in the buffers are called periods [kinda, the terminology is a little squishy] So one period is the first half of the buffer and the other period is the second half of the buffer.. === > below is output by lsmod: [...] > snd_cs4281 20000 0 > gameport 14632 2 snd_cs4281 > snd_rawmidi 22560 1 snd_cs4281 > snd_ac97_codec 83104 1 snd_cs4281 > snd_ac97_bus 2400 1 snd_ac97_codec > snd_pcm 68676 2 snd_cs4281,snd_ac97_codec > snd_page_alloc 9640 1 snd_pcm > snd_opl3_lib 9920 1 snd_cs4281 > snd_seq_device 7820 2 snd_rawmidi,snd_opl3_lib > snd_timer 20996 2 snd_pcm,snd_opl3_lib > snd_hwdep 8836 1 snd_opl3_lib > snd 47012 8 snd_cs4281,snd_rawmidi,snd_ac97_codec,snd_pcm,snd_opl3_lib,snd_seq_device,snd_timer,snd_hwdep [...] > soundcore 9248 2 btaudio,snd [...] Try to do (as root): modprobe snd_pcm_oss If that does not help then you should post the output of cat /dev/sndstat and the error messages that you get from running "amixer". === $alsaplayer -o jack === > I am trying to > get a good overall picture of what can be done here and anyone with more > experience than me might be able to give me some insight here. Finally, > are there any strong audio applications that run in Linux that were not > included in Ubuntu studio, or does it come with all the best one already? Rosegarden is only included in an outdated version - this rogram has made quite great progress since the version in ubuntustudio. Lmms is not included, phasex dito. Some other goodies are missed like slag, aldrin or traverso. === Subject: Re: [LAU] unable to set scheduling priority with rt kernel On 10/30/07, Lars Luthman wrote: > > > With JACK output? > > From the Csound manual: > > "DO NOT use "--sched" if you are using JACK for audio output. JACK > controls scheduling for the audio applications connected to it, and also > tries to run at the highest possible priority. If the "--sched" flag is > used, Csound and JACK will be competing rather than cooperating, > resulting in extremely poor performance." Hold up. From the exact same page of the Csound manual: "An example of buffer settings for low latency on a fast system: jackd -d alsa -P -r 48000 -p 64 -n 4 -zt & csound -+rtaudio=jack -b 64 -B 256 [...] with real time scheduling (as root): jackd -R -P 90 -d alsa -P -r 48000 -p 64 -n 2 -zt & csound --sched=80,90,10 -d -+rtaudio=jack -b 64 -B 192 [...]" So on the same page where it says DO NOT use --sched with jack, it says to use --sched=N with jack. If you look in the Csound manual under the commandline flags section, you will see that --sched and --sched=N are NOT the same flag. Thanks though for drawing my attention to it. === Subject: Re: [LAU] jackd @ startup > i feel like a fool > whenever i do > jackd -R -dalsa -S > at the beginning of everyday's routine.. Maybe you should get qjackctl and set it to auto-start jackd > but i also wonder if there is an issue that i'd better get aware of > befor going this way > maybe flash things will loose sound when in firefox? That's what I find, I think. I only start jack when I'm recording (I have qjackctl bound to a hotkey in fluxbox). If I'm just using web or listening to mp3s jack isn't running. If firefox is playing sounds (or has played some in the past) then I have to quit it before starting jack. === Subject: Re: [SoX-users] Get the length of a .wav-file... How? > > i want to mix two files, but i do not the legth of the first-/above-file. > > how can i find out the length of a file? > If you're using sox, you probably have libsndfile installed. With libsndfile there is a utility called sndfile-info. From a shell, just execute the command sndfile-info file.wav This will print some information including the file duration. === Subject: [LAU] [ANN] Aqualung 0.9beta9 released The Aqualung development team is pleased to announce the latest release of the Aqualung music player. Aqualung is an advanced music player originally targeted at the GNU/Linux operating system, today also running on FreeBSD, OpenBSD and Microsoft Windows. It plays audio CDs, internet radio streams and podcasts as well as soundfiles in just about any audio format and has the feature of inserting no gaps between adjacent tracks. * Support for podcasts. Aqualung can subscribe to RSS and Atom audio podcasts, and automatically download and add new files to the Music Store. Optional limits for the age, size and number of downloaded files can be set. === Subject: Re: [LAU] [jack] not running (permissions)[solved] > JACK more or less requires exclusive access to the audio interface > hardware. If you want to run an application that does not interface > with JACK (and these days, most music-creation related applications > do), you do have to stop JACK. That's not entirely true. If you install alsa-plugins you can configure alsa to default to the jack plugin. See the instructions at http://alsa.opensrc.org/index.php/Jack_%28plugin%29 . This lets me run prboom through JACK even though prboom is not JACK-aware: the ALSA configuration silently reroutes prboom through JACK. === Subject: Re: [LAU] Saving youtube audio and video Adam Sampson: > thomas fisher writes: > >> Tommi promising looking application, but it appears to be not open >> source and a limited trial time period teaser. > > It's really not very hard to save videos from YouTube, and no > proprietary software is required -- you just need to build the right > URL, and there are various scripts to do that: > > http://www.arrakis.es/~rggi3/youtube-dl/ > http://offog.org/darcs/misccode/youtubeget > > (I wrote the second, but I wouldn't have bothered had I known the > first existed...) > > The resulting files can be played with VLC or MPlayer. Here's my script to reliably stream youtube videos: (it depends on yotube-dl) #!/bin/sh rm -f /tmp/ai echo $1 >/tmp/link xterm -e youtube-dl `cat /tmp/link` -o /tmp/ai & while [ ! -s /tmp/ai ] ; do echo "File not available yet, sleeping 2 seconds" sleep 2; done echo "Caching 10 seconds before starting to play" sleep 3; echo 7 sleep 3; echo 4 sleep 3; echo 1 sleep 1; echo 0 echo echo "In case mplayer stops abruptly, run:" echo "mplayer -delay 0.3 /tmp/ai" echo "...to manually restart the player." xterm -e mplayer -delay 0.3 /tmp/ai === Subject: [LAU] [ANN] jack-keyboard 2.2 Version 2.2 of jack-keyboard can be downloaded from http://pin.if.uz.zgora.pl/~trasz/jack-keyboard/jack-keyboard-2.2.tar.gz jack-keyboard is a virtual MIDI keyboard - a program that allows you to send JACK MIDI events (play ;-) using your PC keyboard. It's somewhat similar to vkeybd, except it uses JACK MIDI instead of ALSA, and the keyboard mapping is much better - it uses the same layout as trackers (like Impulse Tracker) did, so you have two and half octaves under your fingers. === Subject: [LAU] [ANN] new version of ssg (Simple Sine Generator) New version of Simple Sine Generator is available. It now requires lv2core. Simple Sine Generator is very simple instrument/generator plugin with midi in and audio out ports. It expected to be useful for testing LV2 hosts and as base for writing your own plugins. It is written in plain C. http://nedko.arnaudov.name/soft/ssg/ http://nedko.arnaudov.name/soft/ssg/ssg-20080109.tar.bz2 http://nedko.arnaudov.name/soft/ssg/ssg-20080109.tar.bz2.sig === Subject: [LAU] [ANN] miniloop-0.0 Released miniloop is a simple live looping program. It can load a number of stereo audio loops of equal length from the disk and loop them in sync with each other, sending each loop to a different pair of JACK audio outputs. These outputs are intended to be subsequently fed into an external software mixer, such as Ardour. For live performance, you will want to control the mixer using a MIDI control surface. miniloop is similar in intent to Stephen Sinclair's LoopDub. I actually created miniloop to explore some design ideas that I had while working on LoopDub. Given that, it is appropriate to provide a comparison between the two programs. The most important difference is that LoopDub uses a built-in mixer, while miniloop uses an external mixer. This means that miniloop is more flexible, but requires a more complex software setup. Another important difference is the user interface, which is radically different, and, I hope, somewhat easier to use. Finally, LoopDub has many features that miniloop lacks; miniloop is currently quite small (~500 SLOC) and quite feature-poor, and I intend to keep it that way. Project homepage here: http://code.google.com/p/miniloop/ Download here: http://code.google.com/p/miniloop/downloads/detail?name=miniloop-0.0.zip === Subject: Re: [LAU] simple DJ program > Still exploring the Eee possibilities, is there a simple mp3 mixing > program out there ? No even doing real time stretching but lightweight > with two playlist and an assignable USB crossfader would be nice. > > Is there such a thing ? http://mixxx.sourceforge.net > http://www.mixxx.org/ (soon) > http://mixxx.sf.net (now) > i don't know if it's still being developed, but in its heyday, http://terminatorx.org/ was quite ok. === Subject: Re: [LAU] music apps scripts Studio 32 wrote: > This is a *.sh script I have now: > > #!/bin/sh > qjackctl & > sleep 2 > lash_panel & > sleep 10 > jack-rack & > ardour2 [session] & > sleep 10 > hydrogen & > exit This is one of the main things that's driving me nuts with Linux applications at the moment- why oh why oh why do so few programs handle resources properly. The main issue with practically everything is that some resource (hardware, daemon, network connection, etc) must be available when the application starts, and remain available until the application exits. Why can't these apps: 1. WAIT for the resource to be ready when starting (especially in the case of jack applications- don't die if jack's not around- it's probably being started right now- "you can't rush these things" "we only let you use it when it's ready"). 2. Release the resource when it's not being actively used. 3. Support the resource becoming unavailable (primary example: I mount a filesystem over the network or wireless. If I want to disconnect from the network or go out of range, I have to stop all applications using the mount and unmount it otherwise everything dies horribly (and I still have to unmount and remount when the network connection comes back)). 4. Handle suspend-to-RAM sanely. I have to reload the module for my miniPCI wireless card, reload the module for my PCMCIA bluetooth card (though that rarely works- it just can't remain in the machine across a suspend cycle or it 'forgets' it's hardware address and ignores me until the next reboot), mute and unmute all channels on my soundcard and reload jack for some stupid reason. The most irritating thing was when I used to have to reload the module for my soundcard as well, which was plain stupid... I need to unload and reload the soundcard module to get sound again. But I can't unload the module because I have audio applications running (xmms, jack, or worse the flash plugin for firefox), and they've all locked the ****ing device, even if they aren't actually playing to it at the time. So I would have to kill all audio applications, unload the module, reload the module, and then reload the applications again. If you aren't playing to the damn card then release the device dammit! Or better yet, ALSA, don't kill off the device node if I want to remove the module. Thank goodness that I don't have to do that anymore. === J M Needham wrote: > LASH might be what you're looking for. > > http://www.nongnu.org/lash/ And it might not be, in which case it's bash to the rescue. Here's the shell script I use to launch my sequencer, QJackCtl, Jack-Rack, JOST, and QSynth: xdosemu -input \r big.bat & qjackctl & sleep 7 jack-rack /home/dlphilp/racks/favoriteplate.rack & qsynth & jost & The big.bat is a DOS batch file that runs in DOSemu to launch my sequencer. QJackCtl and QSynth can be configured for autostart and autoconnection, plus the PatchBay in QJackCtl can recall all connections. See the help options for each program to see what can done at the command prompt. Btw, the sleep command (for seven seconds) is necessary to keep the other apps from loading until JACK has been started. I named the file sequencer.scr and added it to my Fluxbox music apps menu. I click on the menu entry, everything happens. It's not LASH, but it'll do until all my favorite Linux audio apps have been LASHed. Also btw: I think all the apps that the original poster wants to run *can* be controlled by LASH. === Subject: Re: [LAU] where did soxmix go? Are you directly dependent on soxmix or just on some commandline tool to mix? If the latter is the case, you could try ecasound. It's scriptable as well. === Subject: Re: [LAU] [ANN] Minicomputer 1.0 softwaresynthesizer released On Fri, 2008-02-22 at 23:44 +0100, Malte Steiner wrote: > > try starting the editor > like this > > ./miniEditor -bg grey -fg black > > this should bring you closer to my vision (pun intended) Great, that's a lot easier to read. The synth sounds great BTW! Thanks for sharing. A couple more UI suggestions: - Even with the correct colours, I still think the typeface is too small - it would be good to have a 'test note' button for quickly auditioning sounds. This would save either hooking up a keyboard or running and connecting vkeybd. === Subject: Re: [LAU] Ardour Session Exchange - experience? > I've been having a blast recording with Ardour on my 64 Studio v2 setup. I'd > like to collaborate with a friend who lives in another country. > > I'm very interested in Ardour Session Exchange, but I can't find much > documentation on it, or much in the way of user experiences. Can anyone point > me in the right direction for a newbie? Does anyone have any other tips of > sharing Ardour sessions remotely? as they say, sex is currently under development. its not really usable with ardour 2.0 at this time, but there is a little work going on. http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user depending on what you are trying to do, this may help: http://www.linuxbs.org/git?p=ardourpack.git;a=tree also, these mail archives may have some help for you: http://lists.lau-cb.peterlutek.com/pipermail/lau-cb-users-lau-cb.peterlutek.com/ === Subject: Re: [LAU] RT error? Quoting Josh Lawrence : > I get this a lot with various applications, I've been meaning to ask > about it. Does it mean that something is wrong with my RT kernel? > This specific error came from Renoise, but again, I've seen it with > other applications as well. > > cannot lock down memory for RT thread (Cannot allocate memory) Realtime applications need instant access to their memory. For that to be possible, the application must "lock" it's memory so that the operating system does not swap it. However, locking is "dangerous" in the sense that a program could lock all available memory and thus take down the whole system. This is why Linux has a system of setting the maximum amount of locked memory. This is controlled via /etc/security/limits.conf memlock attribute. See http://irc.esben-stien.name/mediawiki/index.php/POSIX_1.e_Realtime_Capability_with_PAM for a quick reference on how to set it up. The reason you are getting that message is probably due to a too low memlock setting. === Subject: Re: [LAU] How to find USB sound card IRQ? lsusb (as root) Subject: Re: [LAU] How to find USB sound card IRQ? 20: 3849997 3850202 IO-APIC-fasteoi uhci_hcd:usb3 and, Bus 003 Device 008: ID 0a92:0091 EGO SYStems, Inc it looks like its IRQ20. === Subject: Re: [LAU] CPU Priorities at limits.conf and this might help in understanding the qjackctl patchbay somehow, http://www.rncbc.org/drupal/node/19#comment-37 Subject: Re: [LAU] CPU Priorities at limits.conf http://alsa-project.org/main/index.php/Low_latency_howto#PAM === Subject: Re: replace RealPlayer On Thu, 13 Mar 2008 08:01:06 pm Peter Garrett wrote: > mplayer -playlist > http://abc.net.au/adelaide/onair/891stream.ram mplayer > -playlist http://abc.net.au/streaming/newsradio.ram > > Both play here from a terminal. Mplayer can play .rm files even > without w32codecs, using ffmpeg. === Subject: Re: [LAU] recommendations for usb sound card? I think my current standard parameters for jack are something like that: /usr/bin/jackd -R -P70 -p128 -dalsa -dhw:1 -r48000 -p1024 -n3 -Xseq Subject: Re: [LAU] recommendations for usb sound card? In Non-advanced mode plays fine: bill:~$ aplay -Dhw:1,0 Hells_intro.wav Playing WAVE 'Hells_intro.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo In advanced mode: bill:~$ aplay -Dhw:1,0 Hells_intro.wav Playing WAVE 'Hells_intro.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:900: Sample format non available In advanced mode jackd says: bill:~$ jackd -R -P70 -p128 -dalsa -dhw:1 -r48000 -p1024 -n3 jackd 0.103.0 Copyright 2001-2005 Paul Davis and others. Subject: Re: [LAU] recommendations for usb sound card? Bill Allen wrote: > $ aplay -Dhw:1,0 Hells_intro.wav > Playing WAVE 'Hells_intro.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > aplay: set_params:900: Sample format non available Using the "hw" device prevents ALSA from doing any automatic sample format conversions. === Subject: [LAU] [ANN] jack-keyboard 2.3 Version 2.3 of jack-keyboard can be downloaded from: http://pin.if.uz.zgora.pl/~trasz/jack-keyboard/jack-keyboard-2.3.tar.gz List of changes is here: http://pin.if.uz.zgora.pl/~trasz/jack-keyboard/changelog.html jack-keyboard is a virtual MIDI keyboard - a program that allows you to send JACK MIDI events (play ;-) using your PC keyboard. It's somewhat similar to vkeybd, except it uses JACK MIDI instead of ALSA, and the keyboard mapping is much better - it uses the same layout as trackers (like Impulse Tracker) did, so you have two and half octaves under your fingers. Subject: Re: [LAU] [ANN] jack-keyboard 2.3 > >> tell me please what the most simple connection with jack keyboard to g= et > >> sound. > > > > Either a2jmidid or "-X seq" option to jackd. > > Can you give a screenshot of the jackconnections in either jack connect > or patchage? Hello if you click on the "+" befor the sytem (Input Ports) you can see all = alsa_midi ports you have. They all called "midi_playback_* " Now you need t= o = find out hwo is hwo. For example, if you start fluidsynth/qsynth and the MI= DI = connect widget from qjackctl is open, you can see pop up a new port. That i= s = fluidsynth. You can simply rename it with a right click on it and rename. = Next time you start fluidsynth it will use the new name. Same work for ever= y = app you use with no direkt jack_midi_port Connectios work as usual under alsa. === Subject: Re: [LAU] Realtime Kernel What's the output of uname -a ?? Subject: Re: [LAU] Realtime Kernel asmok@edubuntu:~$ uname -a Linux edubuntu 2.6.22-14-rt #1 SMP PREEMPT RT Tue Feb 12 09:57:10 UTC 2008 i686 GNU/Linux asmok@edubuntu:~$ Here is wiki page about real time kernel in Ubuntu 7.10: https://wiki.ubuntu.com/RealTime/Gutsy More deeper: http://rt.wiki.kernel.org/index.php/Main_Page Every music oriented linux distro has one: http://linux-sound.org/distro.html Subject: Re: [LAU] Realtime Kernel It is not possible to use ANY kernel with NO latencies. EVERY kernel has latencies. The question is how low do you want the latencies to be? If you can exist with 50mS or higher you might get away with a standard kernel. If you want to run with 1.2mS latency then you will absolutely have to have a real-time enabled kernel. xruns are just an indication of the system not meeting the latency requirement you set in Jack, i.e. - a by product of how you answered the question in the first paragraph. Subject: Re: [LAU] Realtime Kernel > So what does a realtime kernel change ? Makes my latency 1 ms ? A typical situation with non -rt systems is that thay can work very well 99.9% of the time, but not when you are doing some specific things, e.g. - having heavy network traffic, - doing frequent screen updates, changing workspaces, etc., - using some hardware devices such as WAN or IR interfaces. An -rt kernel will be more robust in those cases, and also gives you the tools to optimize your system by assigning interrupt handling priorities. The result is that you can have lower latencies. Subject: Re: [LAU] Realtime Kernel http://rt.wiki.kernel.org/index.php/Main_Page There is really no reason that I know of that you cannot build this kernel yourself. It's not horribly difficult once you've done it a few times. It can be quite daunting when you're up to bat for the first time though. the trick is understanding what driver modules you are using and making sure all of those are included in the configuration of your new kernel. Once you get past that step it's just a matter of adding a new option in the grub config file and trying it out. From the RTWiki these are the commands to download and patch a kernel: # wget ftp://ftp.kernel.org/pub/linux/kernel/v2.6/linux-2.6.23.1.tar.bz2 # wget http://www.kernel.org/pub/linux/kernel/projects/rt/patch-2.6.23.1-rt= 11.bz2 # tar xfj linux-2.6.23.1.tar.bz2 # cd linux-2.6.23.1 # bzcat2 ../patch-2.6.23.1-rt11.bz2 | patch -p1 The above is for 2.6.23. If you were going to do it you might as well go for 2.6.24. Changing the kernel itself should not cause any changes in the stability of your applications. They really exist at a higher level and just talk to the kernel underneath it. If a new minor revision kernel upgrade breaks an application I think it would generally be considered a regression in the kernel and would get fixed. I have kernels ranging back to 2.6.14. they all work. Subject: Re: [LAU] Realtime Kernel Even the standard kernels have 'realtime support' in the sense that you can have realtime threads. The difference with the -rt kernels is that the latter will allow you to push the limits, by making some possibly long kernel code paths pre-emptable, and by handling interrupts in a more flexible way. But today's 'standard' kernels can perform quite well unless you're unlucky or you want very low latencies. > I changed my /etc/security/limits.conf file as: > @audio - rtprio 95 > @audio - memlock 512000 > @audio - nice -19 This has no direct relation to the -rt features of the kernel, but it allows normal users (in the 'audio' group) to create realtime threads and to lock memory. Without this you would have to be root to do this. Subject: Re: [LAU] Realtime Kernel Looking at the name of the kernel is not a good idea;). zcat /proc/config.gz | grep CONFIG_PREEMPT_RT Also, take a look here: http://irc.esben-stien.name/mediawiki/index.php/Setting_Up_Real_Time_Operation_on_GNU/Linux_Systems === Subject: [LAU] Nted (lilypond export) I asked a few months ago for a music typesetter with lilypond export function. I've downloaded the new nted today >>> http://vsr.informatik.tu-chemnitz.de/staff/jan/nted/nted.xhtml And guess what? New! a export to lilypond function :) I'm starting to like this program! === Subject: [LAU] music page updated http://linux-sound.org/ardour-music.html you can try with rubberband: http://breakfastquay.com/rubberband/ === Subject: Re: how to save streaming real media > > Is there any way to save the streaming web casts (say, the ones > from UC Berkeley) on my Ubuntu PC? I am behind an http proxy. > I tried saving them with mplayer, but have been unsuccessful > so far. That's odd MPlayer normally works. Remember that sometimes the RealMedia file on the web is actually a playlist so try -playlist file.rm Also don't forget -dumpstream -dumpfile "myfile" You may need to tell MPlayer to use the HTTP proxy, make sure the environmental variable http_proxy is set correctly. I have had problems streaming RealMedia using the version of MPlayer shipped with 6.06LTS, installing the latest version (and applying the security patches!) from the MPlayer website fixed it. Are you getting any particular error message when streaming using MPlayer? Have you tried verbose mode? Subject: Re: how to save streaming real media I do not know if this is what you are looking for but there is an extension for Firefox, DownloadHelper, that allows you to save almost any video. Probably not what you are looking for but just tossing it out from the noob suggestion box. Subject: Re: how to save streaming real media This first part is youtube & flash oriented, and very simplified as I posted it on the Mozilla SeaMonkey group sometime back. Info on rm streams follows the flash info. For Ubuntu/Debian users (and probably most linux) here is a simple trick for keeping the video on your hard drive: [Simplified for new Ubuntu users w/Gnome] After you've played the video (example: http://www.youtube.com/watch?v=B-kKKXB2_ZI&NR=1 - that's a clip of Mohammad Ali) and with the _video screen still open_, go to Places|Home Folder - click the tiny arrow next to 'File System' and then go to the 'tmp' folder. If you close the flash - the tmp file will go away, so don't close the window until you're finished copying (below). You will see the file that was downloaded to play the video. Just right-click to copy and then paste to a folder in your home directory and then rename it to something that makes sense. You can add a .flv or .swf extension after the filename if you wish so that people with Windows systems can understand what the heck it is if you send it to them, but it's not necessary if you only use it on your system as linux will automatically recognize it as a Flash video. To play the video from your hard drive in it's new location, just double-click it and MoviePlayer will automatically come up and start playing the video. I realize that does not work with a RM stream, but UC Berkeley also publish using Flash: http://www.youtube.com/ucberkeley If you're lucky the course might be in flash. Otherwise, to capture an rm use mplayer (sorry - I know you already have mplayer, but I'm adding how to get mplay for others that might not know how, so bear with me): http://www.mplayerhq.hu/DOCS/HTML/en/streaming.html#streaming-save $ sudo apt-get install mplayer mozilla-mplayer mplayer-fonts Also see: https://help.ubuntu.com/community/MPlayer https://help.ubuntu.com/community/RestrictedFormats You'll need to know the rm url, so the first thing is to save the metafile. Open that file in a text editor to see the actual rm url. Then from a terminal window: mplayer -dumpstream -dumpfile So, for example, to capture the tomography lecture: I determine that the actual rm URL is: I do this by right clicking the tomography rm php (the one on the left with the eye icon) and save that to file (stream.php) to my disk. I then open the file with the text editor and can see & copy the actual rm url link. And to capture I drop everything after the .rm?, so: $ mplayer rtsp://169.229.131.16:554//bibs/s2006/group1/ee225b/20060118.rm -dumpstream -dumpfile tomography.rm [above is all one line] and mplayer goes out, starts the stream, and dumps to the file tomography.rm. Be sure to wait for the entire stream to finish, otherwise you'll be left with a partial file. Note: the above is a very large stream - 138.5MB, so unless you're _really_ interested in tomography don't bother :-) Also, the file will have a bunch of color bar/tone junk at the front, but once past that the entire lecture is there & plays in both RealPlayer and Mplayer just fine. Also note that this url uses port 554, so you'll need to open that on your proxy/firewall in order to actually get the example stream this way. Subject: Re: how to save streaming real media Also there is a Firefox add on named Video DownloadHelper. You can download it from here: https://addons.mozilla.org/en-US/firefox/addon/3006 Subject: Downloading a youtube video and extracting only the audio I succesfully downloaded a youtube video in .flv format after lots of googling. However following directions to extract the audio has not worked for me with no obvious reasons to me why. I checked the video I download with VLC. It does have audio. Here's what I've tried to extract the audio. On other places on the net, I saw the same two commands working for others. Here's the video. http://www.youtube.com/watch?v=Ejb8QOyjz-E& In order to get the video.flv, I did some tricky stuff with wget I found from google that worked! I just wished extracting the audio was as easy. Here's how I did it. http://www.go2linux.org/wget-to-download-youtube-videos Attempt #1 [chris@localhost tmp]$ mplayer -dumpaudio video.flv MPlayer 1.0rc2-4.1.2 (C) 2000-2007 MPlayer Team CPU: AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ (Family: 15, Model: 43, Stepping: 1) CPUflags: MMX: 1 MMX2: 1 3DNow: 1 3DNow2: 1 SSE: 1 SSE2: 1 Compiled with runtime CPU detection. mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing video.flv. libavformat file format detected. [lavf] Video stream found, -vid 0 [lavf] Audio stream found, -aid 1 VIDEO: [FLV1] 320x240 0bpp 25.000 fps 0.0 kbps ( 0.0 kbyte/s) Core dumped ;) Exiting... (End of file) [chris@localhost tmp]$ Attempt #2 [root@localhost tmp]# ffmpeg -i video.flv -f mp3 -vn -acodec copy carsounds2.mp3 FFmpeg version SVN-r10703, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --prefix=/usr --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags=-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic --enable-liba52 --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libogg --enable-libtheora --enable-libvorbis --enable-libxvid --enable-libx264 --enable-pp --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-opts --disable-strip libavutil version: 49.5.0 libavcodec version: 51.45.0 libavformat version: 51.14.0 built on Oct 18 2007 03:18:27, gcc: 4.1.2 20070925 (Red Hat 4.1.2-31) Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 25.00 (25/1) Input #0, flv, from 'video.flv': Duration: 00:09:58.7, start: 0.000000, bitrate: 56 kb/s Stream #0.0: Video: flv, yuv420p, 320x240, 25.00 fps(r) Stream #0.1: Audio: mp3, 22050 Hz, mono, 56 kb/s Output #0, mp3, to 'carsounds2.mp3': Stream #0.0: Audio: libmp3lame, 22050 Hz, mono, 56 kb/s Stream mapping: Stream #0.1 -> #0.0 Press [q] to stop encoding size= 4188kB time=598.8 bitrate= 57.3kbits/s video:0kB audio:4188kB global headers:0kB muxing overhead 0.000746% [root@localhost tmp]# Subject: Re: Downloading a youtube video and extracting only the audio Make sure you have installed ffmpeg and then read the man page. Depending on your requirements, the command will look something like this: ffmpeg -i filename.flv -ab 192 -ar 44100 filename.mp3 Subject: Re: Downloading a youtube video and extracting only the audio mplayer file.flv -ao pcm:file=audio.wav Subject: Re: Downloading a youtube video and extracting only the audio After you've played the video (example: http://www.youtube.com/watch?v=B-kKKXB2_ZI&NR=1 - that's a clip of Mohammad Ali) and with the _video screen still open_, go to Places|Home Folder - then 'File System' and then go to the 'tmp' folder. If you close the flash - the tmp file will go away, so don't close the window until you're finished copying (below). You will see the file that was downloaded to play the video. Just right-click to copy and then paste to a folder in your home directory and then rename it to something that makes sense. You can add a .flv or .swf extension after the filename if you wish so that people with Windows systems can understand what the heck it is if you send it to them, but it's not necessary if you only use it on your system as linux will automatically recognize it as a Flash video. Subject: Re: Downloading a youtube video and extracting only the audio Try an FLV produced by Download Helper, then bring it into avidemux or audacity to take the audio. Download Helper - https://addons.mozilla.org/en-US/firefox/addon/3006 avidemux/audacity can be found in the ubuntu repositories (Add/Remove or sudo apt-get install ) === Subject: Re: Jack rt problems in hardy Here is my limits.conf : @audio - rtprio 99 @audio - nice -15 @audio - memlock unlimited =20 To achieve a good latency (10 ms), I launch jackd with the command nice -n-15 before. Here is the command in qjackctl (server path) : nice -n-15 /usr/bin/jackd To achieve an ultra low latency (< 2.67 ms), I have to stop the Network Manager... Subject: Re: Jack rt problems in hardy Cory K. wrote: > Can someone try adding *nohz=off* to their grub line? > > Example: > > title Ubuntu 8.04, kernel 2.6.24-16-generic > root (hd0,1) > kernel /boot/vmlinuz-2.6.24-16-generic root=UUID=d01c8f74-2cb0-4f2e-940e-8428084b03f3 ro quiet splash *nohz=off* > initrd /boot/initrd.img-2.6.24-16-generic > quiet > Sorry *nohz=off* should be: nohz=off. Subject: Re: Jack rt problems in hardy > rob free wrote: > = >> I've added nohz=3Doff to my rt grub line and run jack with nice - 15 and >> I dont seem to have any xruns with a latency of 8ms. so many thanks =EF= =BB=BFfor >> your help Fabien and Cory ! >> Robin >> = >> = > > I'd like others to try this and report. If this is the problem, we can > get a kernel fix in soon. > = I can confirm this. However it also breaks my acpi support... Sleep mode = no longer works. === Subject: Re: Ripping a DVD It depends if you opt for an application with GUI frontend or console. I prefer mencoder, but it is not the best choise for everyone. Read this http://www.linux.com/feature/128105 Subject: Re: Ripping a DVD Personally I find dvd::rip very powerful, and after a small learning curve not that hard to use. Subject: Re: Ripping a DVD > First: > What is the easiest program to use Thoggen, hands down. > Second: > Want to rip in a lossless format > > Third: > Want to setup screen size to 1280 x 800 or close to that. Thoggen just rips to ogg, no choice there. You do have choice about the resolution, though. I wonder what you hope to gain by ripping to 1280x800, though, since AFAICT the DVD native resolution is just 576p === Subject: Re: No sound open a terminal, enter lspci and copy and paste the output into an email. === Subject: Re: [LAU] Voice to MIDI waon is a Wave-to-Notes transcriber, it read wav file and write standard MIDI file format 0. You will find it here: http://www.kichiki.com/WAON/waon.html Subject: Re: [LAU] Voice to MIDI > one part of the clam software can do that. it is called voice2midi. Subject: Re: [LAU] Voice to MIDI Denemo has a "enter notes through mic" option Subject: Re: [LAU] Voice to MIDI and also rakarrack have a wav to midi converter. It work near realtime. http://rakarrack.sourceforge.net/ === Subject: Re: [LAU] rubberband and batch processing] I often just use find for batch processing, eg: $ find . -name "*wav" -exec rubberband === Subject: Re: [LAU] Microtuneable Software / Piano > you can use Scala to generate > tuning files to use with Timidity and other synthesizers. It takes some > time to get used to it but it works very well. Scala is truly awesome software. IIRC, ZynAddSubFX supports Scala intonation files. === Subject: Re: Media Players - Hardy > I am looking for a reliable music player. Right now, all the > programs (Amarok, RhythmBox, Banshee) seem to work imperfectly on > this platform. Amarok and banshee freeze all too frequently and > RhythmBox often just simply doesn't play. Most likely you have some issues with your low-level sound configuraton, things like you describe used to happen to me when I had some flash animation in firefox locking sound card. If your player freezes or do not play, you can try sudo lsof /dev/snd/* to see which (if any) processess are using your sound devices. In the long term, getting properly configured pulseaudio is most likely the best solution. ===== Subject: Re: Media Players - Hardy Most likely you have some issues with your low-level sound configuraton, things like you describe used to happen to me when I had some flash animation in firefox locking sound card. *I am finding that this is exactly when my audio player no longer works. It seems that whichever process (flash or media player) is open first will prevent subsequent audio from playing.* If your player freezes or do not play, you can try sudo lsof /dev/snd/* *I tried this but not entirely sure which exactly is the offending process or even how I close it.* COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME pulseaudi 5867 sam 17w CHR 116,0 11509 /dev/snd/controlC0 pulseaudi 5867 sam 24u CHR 116,0 11509 /dev/snd/controlC0 mixer_app 6084 sam 20r CHR 116,0 11509 /dev/snd/controlC0 firefox 6175 sam mem CHR 116,16 11467 /dev/snd/pcmC0D0p firefox 6175 sam 49r CHR 116,33 11289 /dev/snd/timer firefox 6175 sam 52u CHR 116,16 11467 /dev/snd/pcmC0D0p firefox 6175 sam 53u CHR 116,0 11509 /dev/snd/controlC0 Any help would be appreciate here as it is a pretty annoying problem. === Subject: Re: audio from mp4 To just get the audio from one of those files I would be inclined to use mplayer with the -dumpaudio option, which can take a description as to the type of audio you want, and conversions can be done on the fly, or you can reencode the dumped audio with an mp3 encoder (or oggenc for ogg). You could for instance specify -ao pcm for wav format: You might need to play with the command line options as well. But try something like this: mplayer -ao pcm:file=foo.wav test.avi Then you can add -vc null -vo null if you don't want to wait for the video to play while you dump the audio. === Subject: Re: How to list directory size? I use $du -h which tells you how many gigabytes you have. Subject: Re: How to list directory size? try also 'du -h -s' and 'du -h -s *' Subject: Re: How to list directory size? du -sh dir_name replace dir_name with the name of your directory :) You may want to look at other options in the manpage. === Subject: Re: [LAU] Editing sound via shell Ecasound. sox will do basic trim / effects / resample / stretching as well, with a shorter command line. It doesn't do effects routing, which can be an advantage if you don't want to deal with that complexity. ecasound is kind of a middle ground with greater flexibility and greater complexity, then you have things like nyquist, clm, and csound, which are meant to be command-line invokable for non-realtime audio processing, but are full fledged programming languages rather than command line tools (ie. not only can you apply effects, but write them from scratch if so desired, and the effects have a much larger number of parameters, etc. Subject: Re: [LAU] Editing sound via shell Ecasound can do a lot of wat you need. Sox may be nice for a few things. For resampling I'd stick to the sndfile-resample tool. For the more elobarte things csound might do the trick. You can also script it. For stretching and pitch-shifting there's also soundtouch/soundstretch. HTH. Kindest regards Julien === Subject: Re: Playback of MIDI files > Hi all, > What do you use for playback of MIDI files? Any good sound fonts or > program you would recommend? Many thanks, > Christian Timidity is a command line midi player. It uses the freepats sound font. Both are in the Ubuntu repositories. http://phorum.sf2midi.com/index.php/topic,65.0.html is another sound font. === mplayer -ao jack multitrack.wav === Subject: Re: [LAU] digidesign mbox 2 micro and Asus eee some things to check: Enshure the sound is not muted with alsamixer or alsamixer -c [card-number (find out which with: aplay -l )] (TAB more options, ESC quit, M un/mute) what's the output of: lsmod | egrep "^snd_usb|^snd_pcm|^soundcore" have you selected the right output in qjackctl === What does 'file HDViD.Eng.XviD' give you? === Subject: Re: [pure:dyne] sc emacs newbie @audio - rtprio 99 @audio - nice -10 @audio - memlock 512000 === zenwalk seem to be able to set priority for each apps? === > I am looking for some audio editing app (or maybe some plugin effect) > that would do the same as SoundForge's Pitch Bend. > > What this effect does is that it presents you with a X-Y graph > representing the time (X) and the pitch (Y). This way, you can draw a > line that represents the bending of the pitch, time-wise. > Snd has an interface for this, located in either gtk-effects.scm (gtk) or new-effects.scm (motif). You'll find it under Effects->"frequency effects"->"Src timevar". The simplest thing might be to install snd-ls: http://www.notam02.no/arkiv/src/snd/ > There is something similar in ReZound, but it does not sound very nice. > I think that it doesn't resample the audio, only shrinks/stretches the > samples it already has, thus outputting a very grainy and not-fat-at-all > low end. > Snd does this a way which sounds very good, using sinc resampling. === jackd, despite the d in its name, is not designed to be a daemon. its not only not designed to be a daemon, it doesn't work very well as one either. === Re: [LAU] recording stream to mp3 On Fri, Jun 15, 2007 at 12:16:47PM +0200, Atte Andr? Jensen wrote: > Hi > > I'd like to be able to record a stream (in jack) directly to mp3 (or > ogg). A great thing would be if I could split the recording into > separate files while recording, for instance by pressing a key-combo > between tracks. > > Does such a thing exist? > I use jack_capture and dump to disk, then oggenc to make oggs. Maybe it'd work with a pipe or a unix fifo? === Re: [LAU] jack problem getting out of hand # Processor type and features # CONFIG_NO_HZ=y CONFIG_HIGH_RES_TIMERS=y than try setting acpi like: # # ACPI (Advanced Configuration and Power Interface) Support # CONFIG_ACPI=y # CONFIG_ACPI_SLEEP is not set # CONFIG_ACPI_PROCFS is not set # CONFIG_ACPI_AC is not set # CONFIG_ACPI_BATTERY is not set # CONFIG_ACPI_BUTTON is not set # CONFIG_ACPI_FAN is not set # CONFIG_ACPI_DOCK is not set # CONFIG_ACPI_PROCESSOR is not set # CONFIG_ACPI_ASUS is not set # CONFIG_ACPI_IBM is not set # CONFIG_ACPI_TOSHIBA is not set # CONFIG_ACPI_CUSTOM_DSDT is not set switching on the acpi in general, but switching off the other acpi stuff like fan, button etc. resolved some issues here. read here: http://rt.wiki.kernel.org/index.php/RT_PREEMPT_HOWTO "NOTE: Since rt patch 2.6.18-rt6 you will probably have to activate ACPI option to activate high resolution timer. Since the TSC timer on PC platforms, as used in the previous versions, are now marked as unsuitable for hrt mode due to many lacks of functionalities and reliabilties, you will need i.E. pm_timer as provided by ACPI to use as clock source. To activate the pm_timer, you can just activate the ACPI_SUPPORT in menuconfig and deactivate all other sub modules like "fan", "processor" or "button". " --- Thanks for the tip, I'll have to recompile my kernel again. But still I don't think this is the cause for Julien's problem. I was having very similar symptoms after installing the latest jack from svn last week. At the same time a new version of gstreamer0.10-plugins-bad got into Debian testing, with the new jackaudiosink. The gstreamer update came a bit bad time, because now I don't know if the frequent zombification, time-outing and watchdog barking is due to some new jack bug or the gstreamer applications constantly entering and leaving the processing graph, triggering some old misfeature (it seems that the time-outs can happen at random times, but I'm not sure, because mostly they happen on graph changes and on my system the graph can change unexpectedly due to a cron job). Anyway, my solution was to add a --timeout parameter (with value 4000), and now jackd has been running happily for a day without the watchdog coming and killing everyone in sight. Pulseaudio and Ingen has been mysteriously zombified only once, as opposed to every 15 minutes, and everything else has been working fine, although there haven't been much stress on the system during the last day. A related question: what does this error message exactly mean? subgraph starting at timed out (subgraph_wait_fd=, status = 0, state = Running) Every client is only connected directly to the sound card playback ports, so the subgraph shouldn't be particularly long. Does the error mean that the said client didn't return from its processing function fast enough or what? After that message jack used to stop delivering audio and then after a few seconds the watchdog would come and eat everyone. I still have got that error twice (the Pulseaudio and Ingen zombifications), but the whole jackd hasn't died anymore. --- === rd drive speed has absolutely nothing to do with audio xruns - disk and audio IO are done in different threads. Try increasing periods per buffer to 3 - lots of HDA intel hardware cannot do 2. Markus Herhoffer wrote: >>> This chipset is known to have some problems, try running jackd with > number of >>> periods = 3. >>> >>> If that doesn't fix it, you might also want to try out the module option >>> position_fix=1 for the snd-hda-intel module. > > I've set both parameters and now it works without xruns at all. But what > does "position_fix=1" actually do? Setting position_fix=1 and jack periods=3 also cleaned up more problems running on my Intel chipset based laptop with: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Just quickly looking at the driver source, position_fix appears to select one 2 or 3 different methods of managing buffers (or something like that). --- > If you run "hdparm -t /dev/'hd or sd1partion" tell you ? --- >> This chipset is known to have some problems, try running jackd with number of >> periods = 3. >> >> If that doesn't fix it, you might also want to try out the module option >> position_fix=1 for the snd-hda-intel module. I've set both parameters and now it works without xruns at all. But what does "position_fix=1" actually do? ===== Re: Configuring ALSA to record audio Re: [linux-audio-user]Extracting/Ripping tracks from DVD-Audio discs To: "A list for linux audio users" On Mon, 2007-03-19 at 19:47 +1100, D. Sen wrote: > Is there a way to configure ALSA (via .asoundrc or such) such that > instead of playing the sound through hardware - the raw pcm data is > actually stored in files? So /dev/dsp0 would actually be linked to a > file...etc. pcm.!default { type plug slave.pcm "asym" } pcm.asym { type asym playback.pcm "save" capture.pcm "dmixin" } pcm.dsp0 { type plug slave.pcm "asym" } pcm.dsp { type plug slave.pcm "asym" } pcm.save { type file slave.pcm "dmixout" file "/var/tmp/record.raw" } pcm.dmixout { type dmix ipc_key 12345 ipc_key_add_uid yes slave { pcm hw channels 2 period_size 1024 buffer_size 32768 rate 44100 } bindings { 0 0 1 1 } } ctl.mixer0 { type hw card 0 } pcm.dmixin { type dsnoop ipc_key 98765 ipc_key_add_uid yes slave { pcm hw channels 2 period_size 1024 buffer_size 32768 rate 44100 } bindings { 0 0 1 1 } } === Subject: Re: [linux-audio-user] jack things please On 12/9/06, we are wrote: > > > #!/bin/bash > > > > if [ -e /proc/asound/$HDSP ]; then > > qjackctl --preset=$HDSP > > else > > quackctl --preset=$ICH > > > > fi > > Yep, > thats exactly what i need. problem solved! > > thanks. > > tom. > To make it clear for users who have other types of cards, a generic syntax could be : #!/bin/bash if [ -e /proc/asound/$MY_REMOVABLE_CARD ]; then qjackctl --preset=$MY_REMOVABLE_CARD else quackctl --preset=$MY_DEFAULT_INTERNAL_CARD fi This is just what I needed to run jackd as a start up daemon... === Subject: Re: [linux-audio-user] qjackctl patchbay - individual ports of a client? > It seems that the qjackctl patchbay only works with clients as a whole - > you cannot selectively connect individual ports within a client. I'd > just like to verify that this is really true. No, it can connect individual ports, but the required incantations are not very intuitive. I always forget the procedure so I never use it. > If the connections in the "Connections" dialog were saved as a > configuration file, that would help, but I think that this is not the case. There is command line app that provides this, called jsnap. Just use this together with qjackctl instead of the patchbay. === Subject: Re: [linux-audio-user] MOre realtime! You don't need the realtime-lsm module any more. That has been superceded by rtlimits. 5ms sounds pretty good to me.. depends what card you are using. I get 20ms (512) for full duplex with SBLive! with no xruns, which is fine for my purposes. If you really want to push the envelope with realtime, you need to install the realtime kernel patches, see: http://www.osadl.org/projects-realtime-kernel.0.html and http://people.redhat.com/mingo/realtime-preempt/ but it is possible that your sound card might not be able to do this, unless it is a proper audiophile quality one.. === Subject: [linux-audio-user] jackd forcing me to choose input or output exclusively On OS X on the MacBook, which has two input sources (mic and line in) and one output source. $ jackd -d coreaudio jackd 0.102.20 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with POSIX SHM support. loading driver .. Default input and output devices are not the same !! Cannot open default device Cannot open the coreaudio driver cannot load driver module coreaudio no message buffer overruns 'jackd -d coreaudio -C' works fine, as does 'jackd -d coreaudio -P'. Fine as in it does what you expect, which is give you either capture or playback, but not both. The -l output is (snipped): Device name = 'Built-in Microphone', internal_name = 'AppleHDAEngineInput:1' (to be used as -d parameter) Device name = 'Built-in Input', internal_name = 'AppleHDAEngineInput:2' (to be used as -d parameter) Device name = 'Built-in Output', internal_name = 'AppleHDAEngineOutput:0' (to be used as -d parameter) I can select which capture device to use using the -d parameter. So if I do "jackd -d coreaudio -D -d 'AppleHDAEngineInput:1'" that doesn't crash, but gives me only capture and no playback. I've tried a few other variations on the theme, including things like "jackd -d coreaudio -C -d 'AppleHDAEngineInput:1' -P -d 'AppleHDAEngineOutput:0'" but to no avail. Am I missing something obvious? Has anyone else had this trouble? Subject: Re: [linux-audio-user] jackd forcing me to choose input or output exclusively On Wed, 2006-12-20 at 13:06 -0700, Hans Fugal wrote: > "jackd -d coreaudio -C -d 'AppleHDAEngineInput:1' -P -d > 'AppleHDAEngineOutput:0'" > but to no avail. > Maybe "jackd -d coreaudio -C 'AppleHDAEngineInput:1' -P 'AppleHDAEngineOutput:0'"? That's the ALSA syntax for using different devices... Subject: Re: [linux-audio-user] jackd forcing me to choose input or output exclusively On Wednesday 20 December 2006 21:06, Hans Fugal wrote: > On OS X on the MacBook, which has two input sources (mic and line in) > and one output source. > Am I missing something obvious? Has anyone else had this trouble? > I believe this has to do with the new way in which the MacBook handles its output and input. You have to create an "Aggregate Device", combining the input and output module, in the Audio/MIDI setup panel. Then you should be able to start JACK using this Aggregate Device, and have both input and output. I had to do the same to get SuperCollider to use both input and output. === Subject: Re: [linux-audio-user] Real-time kernel On Thu, 2006-12-21 at 17:50 -0500, Chuckk Hubbard wrote: > alsamixergui loads fine, runs fine, but no sound. > During boot, I get the message: > hda_codec: invalid dep_range_val 0:7fff > many times, and I'm told this is the problem. I can't seem to find > any documentation on what this means or how to combat it, and the > usual channels have no answers. > > As Paul pointed out, Intel HDA stuff is a mess. It's not the fault of Linux but of the hardware which was designed to be as cheap as possible. My advice to you is to get a good soundcard. --- Subject: Re: [linux-audio-user] Real-time kernel On Thu, 2006-12-21 at 16:47 -0800, Ken Restivo wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > That's bleak. > > I bought a Mac Mini to do audio on. It's got the dreaded hda_intel chip. > Actually that's the problem - there's no single "dreaded hda_intel chip", because the design gives the vendor of your laptop/mobo so much flexibility to screw things up. Each laptop/mobo with this chip has to be debugged separately, based on the codec and how broken your BIOS is. In general the cheaper your machine the more issues you will have because the bargain hardware vendors are more likely to skimp on QA and just ship it as soon as it boots Windows. === Subject: [linux-audio-user] a beat of rhytmic noise http://konstruktiv.org/_media/c0ff/music/beat27.ogg sw: jackd, timemachine, hdspmixer hw: hdsp, a6, mpx550 mandatory /dev/hands + /dev/ears. === Subject: [linux-audio-user] [Announce] IanniX 0.639b online IanniX 0.639b is now online with new examples : http://sourceforge.net/project/showfiles.php?group_id=174402 Any feedback, comment, bug report, etc, would be greatly appreciated and can be posted to the following mailing-list: https://lists.sourceforge.net/lists/listinfo/iannix-discussion === Subject: [linux-audio-user] ANN: Freewheeling Live Looper 0.5.5 Release (Mac OS X / Linux) ( FreeWheeling is an audio tool for live looping. It provides a highly configurable, fluid user interface for instrumentalists to capture loops in real-time. FreeWheeling runs under Linux (PC/Mac) and Mac OS X (PPC/Intel). ) http://freewheeling.sourceforge.net/welcome.shtml Freewheeling v0.5.5 is here. What's new? A host of stuff: - MIDI sync transmit for syncing to Ableton Live / other sequencers - Select, trigger, and fade multiple loops all at once - New configuration with more MIDI controls - Continuously variable feedback for overdubs- EchoPlex style - Browse patches, scenes, and loops via MIDI program change knobs - Save over existing scenes - Toggle-able sync panel for reduced screen clutter - Better stability === Subject: Re: [linux-audio-user] debian / ubuntu I've been running Debian "testing" that has then been dist-upgraded to the latest "unstable" and running a few scripts and a realtime capable kernel called "slh" found at http://www.sidux.com (a project to build on the latest "sid") Subject: Re: [linux-audio-user] Re: debian/ubuntu Regarding the audio-question: the main thing is the Kernel and the sytem setup. I use a third-Party-Kernel in Ubuntu Edgy Eft from a french team called ttoine: http://forum.ttoine.net/viewtopic.php?id=38 Works OK - not extreme hi performance but very good results with realworld-usage-scenarios. The most apps I installed from official universe repos, some (cinelerra) from third-party repos and some like Audacity with dpkg. === Subject: Re: [linux-audio-user] Which distribution? (Used to be DeMuDi) 64 Studio also does a 32bit version: http://www.64studio.com/wiki/FrequentlyAskedQuestions Most of the work done on Demudi has flown back to Debian. "Choose Debian, it's the system that's here to stay!" The only thing missing is a low latency kernel, but you could use the i386 one from 64 Studio: http://archive.64studio.com/pool/main/l/linux-2.6/ Subject: Re: [linux-audio-user] Which distribution? (Used to be DeMuDi) Yes, I would also recommend testing. Testing is very stable, though not by name. === Subject: [linux-audio-user] SuperCollider works great on Linux, audio results Searching the archive, there has been very little mention of SuperCollider on this list in the past year. I have been learning SuperCollider since the summer and think it's super. I wanted to share with you all what I have made: a tool called SCVamp for improvising music with a GUI programmed in SuperCollider. http://the3rd2nd.com/SCVamp/ You might like to listen to some examples. Here are two zips of mp3s. The first set contains nine mp3s (about 45 minutes of music), and the second has two mp3s (10 minutes of music); each set uses different synth sounds and scales. All of the tracks were improvised with this SuperCollider program and recorded in Ardour; they were also very quickly processed with Jamin. http://the3rd2nd.com/mp3/SCVamp_ensemble_a_mp3s.zip http://the3rd2nd.com/mp3/SCVamp_ensemble_b_mp3s.zip More information about the audio is here: http://the3rd2nd.com/SCVamp/about-scvamp/audio-examples/ I would love to hear your feedback, criticism, suggestions, and advice. Renick === Subject: [linux-audio-user] random sample generator I'm having great fun using fastbreeder (and also a bit amsynth) for surprising me with random sounds. Are there any other tools outthere that'll do similar things, esp weird, rhythmic grooves? Subject: Re: [linux-audio-user] random sample generator not quite the same, but check out bbcut: http://www.cus.cam.ac.uk/~nc272/bbcut2.html "Algorithmic Composition Methods for Breakbeat Science" http://www.cus.cam.ac.uk/~nc272/papers/pdfs/acmethodsforbbsci.pdf === Subject: Re: [linux-audio-user] Tao Synth >Where, pray tell, is the new website? > Took some digging, but here 'tis : http://taopm.sourceforge.net/ === Subject: [linux-audio-user] Re: SuperCollider works great on Linux, audio results > In past experience I've run SC3 with the scsynth and sclang binaries. > Now that I'm using an AMD64 machine that's all changed. The sclang > interface will not work with x86_64, and I'm currently unwilling to set > up a chroot for it. Sorry, Dave. I have no personal experience with x86_64, but in the sc-users archive, Stephan Kersten does say: "you need to build sclang in a 32 bit chroot, some info on how to do that on debian is here: https://alioth.debian.org/docman/view.php/30192/21/debian-amd64-howto.html http://wiki.debian.org/DebianAMD64 scsynth needs to be built separately for 64 bit, passing LANG=0 to scons. sclang can be run directly from the 64 bit environment: http://elists.resynthesize.com/sc-users/2005/11/1542178/" http://www.nabble.com/compile-problem-amd64-tf1376023.html#a3693833 > 1. What GUI are you using for SC3 ? I am using SwingOSC. I chose it for the amount of documentation (relative to SCUM or Puxeddu's GTK bindings) as well as, with the GUI class, being able to run a good deal of stuff done by Mac SuperCollider users. Hanns Rutz has been developing it pretty actively over the fall. He's been very responsive to any questions/problems I've had. He's also looking for developers to help the project move along. http://www.sciss.de/swingOSC/ > 2. What version of SC3 are you using ? I built it from sources I downloaded in late July. > 3. Will SCVamp work on AMD64 ? I have not yet heard from anyone who has tried it in that environment yet, but I'd love for you to be the first :) > 4. Does SCVamp replace sclang ? Not at all. In fact, it requires it at the moment. > 5. Are there any other replacements for sclang ? There are replacements for sclang, such as Rohan Drape's stuff which I mentioned before: http://slavepianos.org/rd/r/ as well as Rutz's jcollider: http://www.sciss.de/jcollider/ I haven't heard of anyone using Psycollider on Linux (an implementation in Python, which is currently used to do SuperCollider stuff in Windows with SwingOSC as the GUI). Unfortunately, the only one of these that might work as an alternative to run my code is Psycollider. The rest would require me to rewrite (which I am somewhat considering, using rsc). Thanks for asking; let me know if I can help/encourage you more to get SCVamp running on your machine: Renick Bell === Subject: [linux-audio-user] [ANN] Mammut V0.57 Homepage: http://www.notam02.no/arkiv/doc/mammut/ Screenshot: http://www.notam02.no/arkiv/doc/mammut/mammut.png Changes 0.22 -> 0.57 --------------------- -Ported from gtk1 to juce. -Various bells and whistles -Binaries for windows and mac -Various other things === Subject: Re: [linux-audio-user] dual-core and jack question in jackd, the JACK threads of every client will be serialized. this means that audio processing will happen on one CPU at a time (not necessarily the same one). as you said, however, this hardly means that the other CPU(s) are idle, they just don't get used for audio processing. note that jackdmp can only use more than 1 CPU if there are parallel data flows in the audio connections. for example, this will not use multiple cores: freqtweak -> sooperlooper -> timemachine neither will ardour -> jamin zynaddsubfx ----+ | +------> ardour | bristol --------+ audio data processing is rarely as inherently parallel as, say, graphics. Subject: Re: [linux-audio-user] dual-core and jack question The problem is that jackd apparently only uses one CPU, so the other CPU is basically running "everything else" that isn't jack-related or RT. Good, but not great. jackdmp seems really promising, is apparently is multithreaded for use on dual-core machines, but I haven't been able to use it yet because its freebob support has only been recently added. It probably won't be shaken out until someone can give the developers a machine with the same CPU as I have, and I don't have that kind of money sitting around. That said, after shaking stuff out, and lots of great help from the developers, I've figured out where my limits are (no more than one softsynth at a time, no more than 3-4 LADSPA plugins at a time), and I am happily making music on a Mac Mini Intel Core Duo 1.66Ghz with 2GB RAM. === Subject: text sound file (was [LAU] (no subject)) On Wed, Jul 25, 2007 at 10:02:21AM -0700, Kevin Cosgrove wrote: > > Is there a program which will convert a sound file into a text > (.txt) or comma separated value (.csv) file? what's about the sox .dat format? sox audio.wav audio.dat Gives something like ; Sample Rate 44100 0 -0.1524353 2.2675737e-05 -0.19134521 4.5351474e-05 -0.2215271 6.8027211e-05 -0.21386719 9.0702948e-05 -0.18746948 0.00011337868 -0.17770386 === Subject: Re: [LAU] LADPSA or DSSI granular synthesis A band member has asked me to reproduce this lead synth sound in Linux. He originally recorded it using Maelstrom. http://www.restivo.org/misc/unreasonable.ogg === Subject: Re: [LAU] Freewheeling?? You can remove a sample from a control by holding down the space bar while activating that control. F7 Saves the current loops to disk. (I think) I also think that F9 sets freewheeling to autosave loops Printscreen writes the current recording to disk On Sun, 29 Jul 2007 03:14:51 pm Brian King wrote: > I've tried freewheeling and it works well to record samples and loop > them but how do I delete samples or save samples?? and are there options > on the interface...seems to work a little strangely on the interface > front?? _______________________________________________ === Subject: Re: [LAU] JACK sample browser? "mplayer -ao jack " works good here... === Subject: Re: [LAU] Ripping audio from a video cd On Tuesday 28 Aug 2007 23:24:13 Ken Restivo wrote: > mplayer vcd://track/device -ao pcm:file=3Dripped.wav -vo null > oggenc ripped.wav === Subject: Re: Parsing Real Media .ram files > The problem I am having with my attempts is that I always get > apparently "invisible" characters at the end of the URL, which stop > mplayer from accessing the URL correctly. For example: > > $ cat classicfm.ram > rtsp://202.6.74.108/broadcast/classicfm.rm > rtsp://202.6.74.108/broadcast/classicfm2.rm > > mplayer $(head -n 1 ~/classicfm.ram) > > [snipped output from mplayer] > > STREAM_LIVE555, URL: rtsp://202.6.74.108/broadcast/classicfm.rm > Stream not seekable! > file format detected. > Failed to get a SDP description from URL > "rtsp://202.6.74.108/broadcast/classicfm.rm%0D": cannot handle DESCRIBE > response: RTSP/1.0 404 Not Found > > The puzzle to me is - where is the "%0D" at the end coming from, and > how do I get rid of it? > > I have tried, for example, adding "cut -d '%' -f 1 " : > > mplayer $(head -n 1 ~/classicfm.ram | cut -d '%' -f 1) > > but this doesn't make any difference... The "%0D" is the mplayer's display of the invisible ASCII character "Carriage Return" or "\r" for C programmers. For several internet protocols it is quite usual to use "\r\n" as line delimiter, while for Unix only "\n" (= ASCII character for "Line Feed") is used. There is no "%" character in your string. Therefore you can't use it as delimiter for cut. Try this instead: mplayer $(head -1 ~/classicfm.ram|cut -f1 -d$(echo -ne '\r')) === Subject: Re: qjackctl realtime question what says: #uname -a ? and: #cat /etc/groups|grep audio ? If you got the rt-kernel running and if you are in group audio, everything should be at its best... === Subject: [LAU] [ANN] Traverso 0.42.0 About Traverso: Traverso is a GPL licensed, cross platform program for recording and mixing music, speech, and sounds on the computer Important changes in this release: * Read and write support for Ogg Vorbis, WavPack, FLAC and MP3 * On the fly sample rate conversion * A simple yet effective Project backup and restore system * Better use of available hard disk bandwidth * Encoding formats for recording added: W64 and WavPack * 3 new themes added, notably the medium-contrast and ubuntu theme * Fixed a number of bugs, added new ones and various improvements all over the place * User Manual translated into German and Portuguese Source tarball and installers for Mac OS X and Windows are available on the download page [1] Distribution packages are being created, with some (openSUSE, Gentoo) allready available for installation. We welcome any feedback in the forums [2], user mailing list [3] or internet relay chat, channel #traverso Enjoy! The Traverso team. [1] http://traverso-daw.org/ [2] http://traverso-daw.org/forum/ [3] http://savannah.nongnu.org/mail/?group=traverso === Subject: Re: [LAU] music apps scripts LASH might be what you're looking for. http://www.nongnu.org/lash/ > I'm using Debian testing with fluxbox. > > I often wants to start several programs to make music... For example: jack + jack-rack or jack + jack-rack + ardour2 + hydrogen. > > How can I make a script to start several programs in a particular order, with the right jack/alsa/midi settings? > > Are there scripts available? === Subject: Re: [LAU] Virtual (midi) keyboard > I want to play a bit with rosegarden and maybe linuxsampler. Is there a > good virtual keyboard on linux or a other usefull tool for it? http://www.alsa-project.org/~tiwai/alsa.html#vkeybd === Subject: Re: [LAU] Virtual (midi) keyboard > I want to play a bit with rosegarden and maybe linuxsampler. Is there a > good virtual keyboard on linux or a other usefull tool for it? > Maybe jack-keyboard is what you are looking for ? http://pin.if.uz.zgora.pl/~trasz/jack-keyboard/ It is jack-midi enabled. Subject: Re: [LAU] Virtual (midi) keyboard > > jack-keyboard-2.2# make install > install -o 0 -g 0 -m 0755 jack-keyboard /usr/local/bin/jack-keyboard > install: cannot create regular file `/usr/local/bin/jack-keyboard': No > such file or directory > make: *** [install] Error 1 > you must do that as root. Subject: Re: [LAU] Virtual (midi) keyboard > I seems not able to build it on debian lenny/ sid... you can find vkeybd in lenny/sid Subject: Re: [LAU] Virtual (midi) keyboard I think that the install command in the Makefile needs the -d option to create any directories/parent directories that don't already exist. Just run mkdir /usr/local/bin as root if that directory doesn't already exist....correct me if I'm wrong !! Linux-audio-user@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user === Subject: [LAU] [ANN] Snd-Rt homepage update, Snd-ls V0.9.8.13 and jack_capture V0.9.7 Snd-Rt ====== Snd-rt is a realtime music programming environment running inside the Snd sound editor. The homepage has been updated with links, papers and many examples on how to create GUI's and how to access Faust. The old fancy design is still kept unchanged. http://www.notam02.no/arkiv/doc/snd-rt/ Snd-ls V0.9.8.13 ================ Snd-ls is a distribution of Bill Schottstaedt's sound editor SND. (http://ccrma.stanford.edu/software/snd/) Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Snd-ls also serves as base code for the San-Dysth softsynth (http://www.notam02.no/~kjetism/sandysth/) and the Snd-rt music programming language (http://www.notam02.no/arkiv/doc/snd-rt) http://www.notam02.no/arkiv/src/snd/ Changes 0.9.8.5 -> 0.9.8.13 --------------------------- -Updated Snd-Rt. (unguote-splicing, gui) -Updated Snd-Rt. (gui, dynamic scoping, oscil* etc.) -Updated Snd-Rt. Faust support. -Fixed rt timer. Previously it only ran 3 hours before giving negative numbers. Now it should run approx. 13 million years. (not tested) -Removed access to removed functions in jack. -Updated eval-c.scm. jack_capture ============ jack_capture is a program for recording soundfiles with jack. Its default operation is to capture whatever sound is going out to your speakers into a file. (But it can do a number of other operations as well...) http://www.notam02.no/arkiv/src/ 0.9.6 -> 0.9.7: *Don't segfault if the opening of a subsequent wav soundfile fails. (ie. if the 4GB limitation is reached on the old file and jack_capture continues writing on a new file.) === Subject: Re: [LAU] Normalize from command line? Take ecanormalize, part of the ecasound package. It normalises files with intact dynamics. What it actually does is using ecasound functions. First it analysis the file, by how much can it be amplified without clipping, than it applies the amplification. Subject: Re: [LAU] Normalize from command line? this is a script I call 0db ----------------------------------------------------------------------------------------------------------- #!/bin/sh if [ $# -eq 1 ]; then { o=`echo $1 | sed -e 's/\([^\.]*\)\(\.\)\(.*\)/\1_normalized.\3/'` }; else { o=$2 }; fi sox -v $(sox $1 -n stat 2>&1 | grep Volume | sed -e 's/[^0-9]*\(.*\)/\1/') $1 $o ------------------------------------------------------------------------------------------------------------- If you drag file.wav to its icon, it will create file_normalized.wav, if you call it on the command line with one argument, it has the same behaviour. If you call from the command line like so: '0db file.wav new.wav' it will create new.wav, as you would expect. Subject: Re: [LAU] Normalize from command line? The normalize program does that for me. Name : normalize Relocations: (not relocatable) Version : 0.7.6 Vendor: Mandriva Release : 7mdk Build Date: Sat 01 Oct 2005 06:07:54 AM PDT Install Date: Tue 13 Feb 2007 05:10:05 PM PST Build Host: n5.mandriva.com Group : Sound Source RPM: normalize-0.7.6-7mdk.src.rpm Size : 148355 License: GPL Signature : DSA/SHA1, Wed 20 Sep 2006 11:34:12 AM PDT, Key ID 445935f878d019f5 Packager : Couriousous URL : http://www.cs.columbia.edu/~cvaill/normalize/ Summary : A tool for adjusting the volume of wave files Description : normalize is an overly complicated tool for adjusting the volume of wave files to a standard volume level. This is useful for things like creating mp3 mixes, where different recording levels on different albums can cause the volume to vary greatly from song to song. Subject: Re: [LAU] Normalize from command line? This is probably not the best way to do it, but it's worked for me: for i in `ls *wav`; do FOO=`sox $i -e stat -v 2>&1` BN=`basename $i .wav` echo "norming $i with factor $FOO ..." sox -v $FOO $i -t wav ${BN}.norm.wav done === Subject: Re: [LAU] Decent and attractive audio player I think someone else here sent it, however: http://en.wikipedia.org/wiki/Comparison_of_media_players Subject: Re: [LAU] Decent and attractive audio player > i was looking for. Describes all the problems i'm having with other > players and professes to have a solution. It's also written in my fav > programming language, python so if it doesn't work like i want i can > make it ;) > > Has anyone tried it? > http://www.sacredchao.net/quodlibet > > Looks like a good tag editing lib too. > An update, i've tried quod libet and despite the very strange name, I really, really like it. Seems to be just what i was looking for. Clean, functional, stable and works well. And FAST. Builds the collection in a few minutes where as amarok is at 20% done after a quite a few hours. All round, happy with it. Haven't really pushed it to much yet though. === Subject: Re: How will Pulse Audio change things for me? holotone@gmail.com kirjoitti: > My question is, how will Pulse Audio, in actual practice, change things for > me as a Linux-based musician? > > I think - nothing. It just expand networked audio possibilities. LTSP5 use it on the Ubuntu's thin client system. http://developer.novell.com/wiki/index.php/Feisty/HOWTO:_PulseAudio http://wiki.ltsp.org/twiki/bin/view/Ltsp/Sound#Which_sound_server_should_I_choo Thin client on the Hardy Beta x86_64: http://www.arkki.info/howto/Wiki/LTSP5_64/Beta_64/Sound_01.png Here is same machine (LTSP5 Server) working as Studio: http://www.arkki.info/howto/Wiki/LTSP5_64/Beta_64/QSynth_Jackd_01.png === Subject: Re: [LAU] Scripts for basic audio processing On Mon, 2008-06-09 at 10:27 -0400, Christopher Stamper wrote: > I need a way to automate some basic audio processing. >=20 > For example, I have a wave audio file of about 80-90 minutes that > needs to be written to a CD in audio-cd format as quickly as possible. > It's speech, and needs to be available immediately. The file should be > split into tracks before writing, though. And, depending on the > length, it may also need to be either trimmed or time-'compressed' to > fit on an regular cdr. >=20 > I'd like to be able to do this with something like audacity, since it > may need to run on windows. But if something exists good enough for > Ubuntu Studio, I may run a VM(Windows is required as host-OS). >=20 > So is there a way to do it easily? It would be excellent if the output > could be a cd-ready iso, but individual wav-files would be fine to. Load it in audacity, do the editing you want and split into tracks, then "Export multiple" to get all tracks as separate audio files in a directory. There should be tools that burn a directory of audio files as tracks on an audio CD on all platforms. Subject: Re: [LAU] Audio editor for editing a 90-minute-long file? > On Wed, 11 Jun 2008 13:56:17 -0700 > Ken Restivo wrote: > > > Can someone recommend an audio editor for editing a 90-minute-long WAV file? > > mhwaveedit ? > > http://gna.org/projects/mhwaveedit/ > or > http://www.gnomefiles.org/app.php/mhWaveEdit oops, wrong account. sweep is worth a look too of course: http://www.metadecks.org/software/sweep/ but it also loads entire files into memory and doesn't support jack (yet). if you are short on memory you'll likely see similar problems :( === Subject: Re: scripting fun > One more refinement I can see - grep piped into awk is redundant most of > the time. > > The command pipeline: > grep DROP | awk '{print $4}' > can be replaced with > awk '/DROP/ {print $4}' > > The second one: > grep -i slurp .../access.log | awk '{print $1}' > is a little trickier since awk does not have a non-verbose > case-insensitive switch, so you have to do one of these: > awk '/[Ss][Ll][Uu][Rr][Pp]/ {print $1}' .../access.log > awk '/slurp/ {print $1}' IGNORECASE=1 .../access.log > === If you would upgrading to the latest Debian you could have a looke her: http://sidux.com/index.php?module=pnWikka&tag=englishAudiostudio sidux is not a audio distributions, but you can simpel fit it to your need's it based on debian/sid and the latest rt-kernel is always aviable. === Subject: Re: [LAU] Audio editor for editing a 90-minute-long file? It is very easy to do that in Snd. Snd also handles large files appropriately (by reading from harddisk while playing and using cached peak files). Using snd-ls, you can just mark the area you want in a new file, and right click, select copy to new or save selection. === Subject: Re: [LAU] How to listen to songs on myspace without proprietary software? you can run 64bit firefox and use nspluginwrapper to run the 32bit flash http://gwenole.beauchesne.info//en/projects/nspluginwrapper You also have to set the 32bit emulation in the kernel. If you have gentoo, here's some instructions. If not, it'll give you the way to do it. http://gentoo-wiki.com/HOWTO_Flash_9_and_Firefox === Subject: Re: [LAU] Playing a wav file from stdin in JACK? > I'm looking for a way to play a wav file from STDIN that'll work with JACK. In general, WAV files are not designed for reading from things like pipes. The problem is that they can contain metadata *after* the audio data. There are however file formats that are specifically design to be allowed to be read from pipes, like AU and CAF. Both of these formats have all metadata before the audio data and do not allow trailing metadata. Both AU and CAF formats support a wide range of audio data formats including 16, 24 and 32 bit integer PCM, 32 bit floats, 64 bit floats, Alaw, ulaw etc. > mplayer -ao jack (or just mplayer without JACK) won't handle WAV files on stdin, for some reason. > > Cannot seek backward in linear streams! > Seek failed Mplayer is probably being a little conservative. libsndfile does allow reading from pipes as long as the file is not too broken. > I need to do this because I'm playing wavpack'ed files: I need to do > "wvunpack -o - filename.wv | SOMETHING - ". I do *not* want to have to > unpack them to disk just to play them. Does wvunpack support output as AU of CAF? If wvunpack only does WAV output, you have two solutions: - Write SOMETHING to use libsndfile which does read from a pipe. - Hack wavpack support into libsndfile :-). === Subject: Re: [LAU] Playing a wav file from stdin in JACK? Interestingly, wavpack can write *and read* wav files from a pipe. I'll try inserting sndfile-convert into the pipeline and see if there's a way to make mplayer happy that way. Subject: Re: [LAU] Playing a wav file from stdin in JACK? > Interestingly, wavpack can write *and read* wav files from a pipe. Its doable for the vast majority of structurally correct files. If the file is malformed (which is way more common than you might think), its can be vastly more difficult. > I'll try inserting sndfile-convert into the pipeline and see if > there's a way to make mplayer happy that way. Isn't there are libsndfile player with a Jack backend? I thought there was one in the JACK sources. Subject: Re: [LAU] Playing a wav file from stdin in JACK? Did you try ecasound? I think I have never seen a wavpack file but perhaps something like: ecasound -i yourwavpackedfile.wv -o jack_alsa Subject: Re: [LAU] realtime kernel Julien Claassen kirjoitti: > If I compile a realtime kernel, with Ingo's patches, do I need to set the timer to 1000Hz. if you use Ubuntu 7.10, you don't have to compile anything, you just install "linux-rt": https://wiki.ubuntu.com/RealTime/Gutsy I believe that any popular distro has their own rt-kernel for installing. I like it that way - you just install rt-kernel, jackd, qsynth and rosegarden and start playing... But if you really want make your own kernel you might want to read this too: http://rt.wiki.kernel.org/index.php/Main_Page === Subject: [jsyn] planets These days I came up with a new Java-JSyn-application - "Planeten" - for audiovisual simulation of elliptic planet movements according to the laws of Newton in a gravitation field. Different modi of graphical representation, multiple access to sounddesign, possibilities for defining and saving individual settings, manual. I want to announce this to the JSyn-list in hope of critical comments about usability and aspects of further improvement. http://www.wolfgang-fulda.de/harmonik/planeten/Planeten.zip === Subject: Re: transcode FLV (YouTube flash) to MP4 (ISO MPEG v4) > > How to I convert a FLV video to MP4 (for iPod and PSP). I downloaded > flv2mpeg4 from getdeb.net, but I doesn't support audio. > > regards > Svev The iPod has some very specific MP4 requirements. I usually use mencoder and do it like this: # encode video mencoder in.avi -ovc x264 -x264encopts crf=25:vbv_maxrate=768:vbv_bufsize=244:nocabac:level_idc=13:subq=6:partitions=all:me=umh:frameref=5:subq=6:trellis=2:direct_pred=auto -vf scale=320:-2,expand=320:240,crop=320:240,harddup -oac pcm -srate 8000 -af channels=1 -o out.avi -ofps 24000/1001 # encode audio faac --tns --mpeg-vers 4 -o out.m4a # extract raw video with MP4Box MP4Box -aviraw video out.avi # combine audio and video with MP4Box MP4Box -new -fps 23.976 -add out_video.h264 -add out.m4a out.mp4 But I like doing things the hard way, so there may already be scripts that do all this automatically. Subject: Re: transcode FLV (YouTube flash) to MP4 (ISO MPEG v4) Very easy & nice GUI for ffmpeg is WinFF: http://winff.org/ It's not in the repositories, but should be shortly: https://bugs.launchpad.net/ubuntu/+bug/172804 In the interim, just download the deb and install. I can't recall if it also pulls in ffmpeg, but if not you can install that via: sudo apt-get install ffmpeg === Subject: Re: [Jack-Devel] Known issues with ubuntu/kubuntu 64 bit >> One is that Hardy Heron uses kernel 2.6.24, and there's = >> a known regression in realtime audio performance with that release, = >> compared to 2.6.23 and earlier kernels. > What about 2.6.22 ( 2.6.22-14-rt) ? It should be better; we still use 2.6.21-rt in our stable branch, on the = 'if it ain't broke don't fix it' principle. >> Another possible reason is that the realtime kernel package in Ubuntu = >> universe is not quite so aggressively configured as the one we use in = >> 64 Studio. > = > Could you be more verbose here ? I'm referring in particular to this kernel config setting: Preemption Mode (Complete Preemption (Real-Time)) I'm not sure if Ubuntu's RT kernel uses that setting. > Do you have any advice to give = > regarding the Ubuntu RT tuning so it can be as aggressive as 64 Studio ? There's a useful page here: http://devel.goto10.org/puredyne/wiki/KernelAndSystemOptimization We've been talking to the pure:dyne team about having a new = linuxaudio.org mailing list for distro building and tuning issues, = because at the moment we're all working to solve the same problems, and = so there's a lot of duplicated effort. === Subject: Re: JACK almost working Figuring out exactly what Jack does was a stumbling block for me, too. I'll give you some real-world explanations which make it easier to understand... Audacity: A multitrack recording program. ZynAddSubFx: A synth interface goodie. ie: A keyboard :) Hydrogen: A drum machine. Jack allows me to connect them like this: Guitar -> Rakarrack -> Audacity Track 1 ZynAddSubFx -> Audacity Track 2 Hydrogen -> Audacity Track 3 The result being that I can play guitar while my mate plays keyboard (on my PC's keyboard) and hydrogen plays back a drum track we put together to make a song that's being recorded by Audacity. To achieve the above, you start up Jack-Control, then the rest of the above programs and set up these connections: 1. Connect system output into Audacity's audio track 1 2. Connect ZynAddSubFX's output into Audacity's audio track 2 3. Connect Hydrogen's output into Audacity's audio track 3 4. Connect Audacity's MASTER track's output to the system input Step 4 lets you hear everything. You can, of course, connect a real MIDI device and then set up a connection between that device and ZynAddSubFx, allowing to play a 'real' keyboard, while still accessing the neat sounds Zyn provides. Before you understand the above, you should follow the instructions in the HOWTO mentioned by Joe and be sure you can get some sound from your speakers using ZynAddSubFx. The rest becomes a lot easier to understand at that point. === Subject: Re: [Jack-Devel] command-line jack file player for multichannel? > > > >>i'm looking for a command-line audio player that will play .wav (or > >>maybe ogg vorbis) files to jack. > > > >>it should support an arbitrary number of channels (currently i need 24) > >>and should have a sane way of congfiguring what jack ports to connect to > >>(autoconnect to the first N physical ports won't do the job). > > > >All channels in one file, or separate ones ? > = > as long as the channels are synced precisely, both are fine. Have you checked if ecasound can meet your needs? (/me pretty sure it will). === Subject: Re: What happened to RT kernel development? > I'm on Intrepid, and linux-rt seems to have stopped at 2.6.24, whereas the regular kernel is up to 2.6.26-4. > Are we supposed to compile our own? Unfortunately a few things have thrown things into a bit of disarray re the rt kernel. Firstly, the kernel package structure/separation has massively changed in intrepid. The Canonical Kernel team do not wish to support any kernels other than official kernels for desktop/server for i386/amd64. Secondly, as far as I am aware, there is no RT patch for 2.6.26 yet. We are keeping our eye on things pretty closely, as we want an rt kernel as much as everybody else, so please bare with us. === Subject: Pen Drive Linux All In One Pendrivelinux 2008 Posted: 30 Jul 2008 12:28 PM CDT http://feeds.feedburner.com/~r/PenDriveLinux/~3/350693616/ This ALL In One version of Pendrivelinux 2008 can be run from a USB thumbdrive either natively (as a stand alone Operating System at boot), or within Windows using qemu emulation. Moreover, the user can save installed software, changes, bookmarks, email, contacts and more automatically via a persistent loop image and then restore those changes [...] === One resource you could try is the Linux Documentation Project. It offers tutorials, guides, and How-To's on Linux commands. You can find it at: http://tldp.org/ These other websites also offer linux tutorials: http://www.linux-tutorial.info/ http://lowfatlinux.com/ http://www.ee.surrey.ac.uk/Teaching/Unix/ http://www.linux.org/lessons/beginner/toc.html http://www.fsid.cvut.cz/cz/U201/LINUX.HTML http://www.mediacollege.com/linux/command-tutorial/ You can also do a good search for "linux tutorials", "linux command tutorials", or "linux commands". I've found this site useful: http://www.ss64.com/bash/index.html === Linux Terminal takes alot of time to learn. It helps alot if you use it constantly. One feature that terminals have over GUIs that make them very useful is persistence. Terminal persistence gives you a terminal you can use where ever you go and it'll be as if you never left your computer at home. This is why I recommend new linux users interested in the Terminal learn to use screen. http://www.amitu.com/blog/2004/12/screen-howto.html Here is only one of many useful tutorials on using screen. Feel free to mail this list with any questions you have about screen. Subject: Re: Terminal > > from my experience, the best way to learn about a command is to use man > followed by the command you want to know about. for example man ls > or man top. A bunch of useful information us already included right in > the distro. The 'opposite' is apropos - for those days when you just can't remember the name of the command to ask 'man' about - it searches the descriptions of commands so you can often find the word you want with it. Also, 'man bash' is incredibly huge & repetitive - and has _tons_ of meaty info. Learn by looking at "shell scripts" - plenty to be found in /etc/ & elsewhere but don't alter them (yet) - use 'less filename' to view & see how others have done stuff. === Subject: Re: [LAT] Hello / kernels El Martes, 5 de Agosto de 2008 12:11, Daniel James escribi=F3: | Hi Marcos, | | > We are now testing: | > | > linux-image-2.6.26-libre - Linux kernel binary image for version | > 2.6.26-libre | > | > but it's not yet realtime patched. | Now, it's realtime http://linux.ilmainen.net/musix/temp/realtime/ | I'd be interested to know how you get on with that kernel. We are testing the *libre kernels. There are no problems with them, = but, because Musix is a Live CD-DVD distro, we need aufs and we could = not build the kernel using the realtime patches+aufs, for now. Alexandre and Tapani are trying it, maybe Free could help, I dont = know, maybe we all could work together. This mailing list should be good? http://lists.linuxaudio.org/listinfo/linux-audio-tuning We are not using squashfs for now, because knoppix does not use it = (but in the short future, that's possible). = | Perhaps = | we can arrive at a shared config or even binaries that all | Debian-derived multimedia distros can use. Well: that's possible, we use Debian ETCH for now, so all the kernel = headers will work fine on ETCH. But we will move on the near future, = I think. | I know of four based on = | Debian - Musix, 64 Studio, Ubuntu Studio and Pure:Dyne, but there | may be others. Artistix...? | Plus there are the users of regular Debian who need = | an RT kernel. | Yes, there are some kernels from Musix that they/you can use # Musix GNU+Linux (stable+backports) /etc/apt/sources.list #deb ftp://musix.ourproject.org/pub/musix/deb/ ./ Also, Gilberto Borges included a function on the musixcontrol panel to = make a realtime enabled Debian distro from any debian based distro. ftp://musix.ourproject.org/pub/musix/deb/painelmusix-1.7_i386.deb | Are you doing anything with Debian Lenny yet? = No: just SID: testing SID to see what happens :), but we are not = focused on SID for now, we are just happy with ETCH, backporting some = packages (ardour and so), so it's not necessary to upgrade to lenny = or sid for now | Our experience is = | that it's not really stable enough to recommend to users, so our | main concern is to solve multimedia package bugs. There is no | Ardour in Lenny, for instance. | I did not knew that! that's odd! | > We at Argentina are introducing FOSS Musix GNU+Linux on music | > schools; for now, the experience is great! | | I recently wrote an article about using GNU/Linux in a school | music club, and because of that I was asked to help with | music-related information for http://www.schoolforge.net/ | === Subject: Re: [LAT] A guide to optimize JACK? > What JACK configuration parameters do I have to vary (and in what way) to > achieve that? > (by "that" I mean: minimum latency, zero noise and zero xruns). read `man jackd` - you want to set "-p -n " -n: usually 2 or 3 - mostly depends on your sound-card -p: 32,64,128,... 4096 "the period" defines buffer-size and latency There's no "best" latency settings. For recording you probably want to use a small period size for low latency monitoring. But when you master, mix or edit: raise it up to >=1024 (many postprocessing effects are very CPU intense, and you do not need low latency to listen). Most versions of JACK allow the period to be changed during runtime: `man jack_bufsize` Latency and Xruns are related: The higher the latency the less chance of xruns. To get a 'zero-xrun' system you want to run jackd with /realtime-priority/ on a realtime-patched linux kernel and lock the it's pages into memory (no swap): 1. get or compile a realtime kernel 2. set up IRQs 3. permit audio users to use realtime capabilities 4. start jackd as user with realtime capabilities For (1) I suggest to get a binary for your gnu/Linux distribution. 64studio, musix, ubuntu-studio, gentoo-pro-audio-overlay, CCRMA.. (2) 64studio offers a debian package rtirq for a great script from Rui: http://www.linuxaudio.org/mailarchive/lau/2006/6/26/127914 There's probably other packages as well. (3) that's the /etc/security/limits.conf part http://tapas.affenbande.org/wordpress/?page_id=73 (4) /usr/bin/jackd -R -P 70 -T -d alsa -n 3 -r 48000 -p 128 -d hw:1,0 or use the "Realtime" checkbox in qjackctl's Setup dialog. zero noise: N/A - check your cables ;) > I'm looking for something like a "standard procedure"... use qjackctl. >...but sometimes the audio crackles, has xruns, can't get it to work > even with high latency, etc... and I don't have a clue about what to > vary to fix that. aah, although this sound can be described as /noisy/, x-runs are no *noise* ;) and are not only correlated to latency. They're normal: Unix is //fair// to all users and sometimes does not see a reason why it should keep playing your sound or even keep it in memory UNLESS you tell it to do so. (steps 1-4 above) There's a lot of background-jobs on a default gnu/Linux; fi. `updatedb` may get in your way.. === Subject: Re: sending to an xterm > > I want to open an xterm, send it a command and have it execute it. You can't do that. xterm doesn't execute shell commands passed on stdin... It can, however, execute one passed on the command line. Instead of just running xterm, you can run "xterm -e 'cmd foo bar'" where cmd is the program to run and foo and bar are its arguments. The problem is that as soon as the program exits, xterm will exit also. Subject: Re: sending to an xterm xterm -hold -e "whatever" Subject: Re: sending to an xterm This appears to be a solution; xterm -e "ls; bash" http://www.linuxforums.org/forum/misc/115239-getting-prompt-after-xterm-e-command.html Subject: RE: sending to an xterm since I do not have access to xterm, here is the interactive session for spawning bash(another session if you will), sending ls command to it, and retrieving the results. things to note are: 1. after spawning expect for the prompt, timeout, and eof #which ever happens first 2. return value is the action that matched 3. if prompt matched, the 'before' has the results 4. even the command 'ls' with '\r\n' will be in the results. ------------actual session-------------------------- madared@WNJ10002LBITAZV:/c/Edwin/Projects/expect $ python Python 2.5.1 (r251:54863, May 18 2007, 16:56:43) [GCC 3.4.4 (cygming special, gdc 0.12, using dmd 0.125)] on cygwin Type "help", "copyright", "credits" or "license" for more information. >>> import pexpect >>> c = pexpect.spawn('/bin/bash') >>> c.expect([pexpect.TIMEOUT, pexpect.EOF, '\$ ']) 2 >>> c.before '\x1b[?1034hmadared@WNJ10002LBITAZV:/c/Edwin/Projects/expect\r\n' >>> c.after '$ ' >>> c.sendline('ls') 3 >>> c.expect([pexpect.TIMEOUT, pexpect.EOF, '\$ ']) 2 >>> c.before 'ls\r\nautoexpect.exp\tscript.exp\r\nmadared@WNJ10002LBITAZV:/c/Edwin/Projects/expect\r\n' >>> c.after '$ ' >>> >>> exit() madared@WNJ10002LBITAZV:/c/Edwin/Projects/expect $ === Subject: Re: transcode FLV (YouTube flash) to MP4 (ISO MPEG v4) >> >>> I can only get the flv to save as MP4 video only - no sound. I have the same result using VLC. Has a bug been reported for this issue? I found the following vlc forum post thread with (what seems to be) the same issue: http://forum.videolan.org/viewtopic.php?f=4&t=42456&start=0&st=0&sk=t&sd=a No matter what type of transcoding I try on the audio the output has no audio. My workaround, I have tried ffmpeg which has worked and given me an FFmpeg MPEG-4 encoded video. Put the flv in your home directory and using the terminal issue the following command: $ ffmpeg -i your_flv_video.flv output_name.avi === Subject: Re: Soundcard not working Would you post the output of the following commands please. cat /proc/asound/cards cat /proc/asound/version lsmod | grep snd lspci -v (just the bits for the soundcard/s will do here) Open a terminal, and as user type, alsamixer . Which should show you the settings for the default card (card0). there may be something muted, or a slider just needs to be pushed up. To unmute controls that are muted, use the M key, which will toggle mute/unmute. If you want to see alsamixer for card1, type, alsamixer -c 1, and that should show controls for c-media one. The ESC key will exit alsamixer. See, man alsamixer for other usefull stuff. === Subject: Re: [LAU] Patching linux 2.6.26.3 > I'm trying to patch the linux kernel version 2.6.26.3 with the rt-patch. But > there's only a patch for 2.6.26. It's not working very well. Because some > changes were already taken into the kernel. Which command should I use to do > the patch properly. Or should I stop patching all together? Or is there another > experimental place to get newer rt-patches. I had this all bookmarked a couple of weeks ago, but deleted them when I got my system working :( The easiest way to get your kernel patched correctly is to use ketchup - not the regular ketchup, but one that understands the RT patches. Here's the RT-aware ketchup: Have a read of the RT wiki: RT patches: I think you are right - there doesn't appear to be a patch against 2.6.26.3 yet; I would advise that you get ketchup to build you 2.6.23-rt1 to start with. You would probably benefit from joining the linux-rt-users mailing list (I did!). Details on at vger.kernel.org: Subject: Re: [LAU] Patching linux 2.6.26.3 Am Donnerstag, den 21.08.2008, 13:02 +0200 schrieb Julien Claassen: > OK, the problem solidified thus: ketchup can't find the public key. what can I > do about this. I have gpgv and gpgkeys debian packages installed. what else is > there? did you import the current kernel key? gpg --keyserver wwwkeys.pgp.net --recv-keys 0x517D0F0E === Subject: Re: [LAU] [LAD] Ubuntu Studio 8.10, -rt and 2.6.27 > moreover, by all accounts, 2.6.26 is totally useless for most > audio/MIDI/music work, so no distro or overlay that is concerned with > that class of user should be considering 2.6.26 either. > This is the reason for either skipping a release or releasing with a -generic kernel (which I use for everything other than audio work) for now with a 2.6.27 kernel in a PPA (https://launchpad.net/ubuntu/+ppas) when upstream can base against .27. Subject: Re: [LAU] [LAD] Ubuntu Studio 8.10, -rt and 2.6.27 is not that .26 is that bad for audio, but *it is* seriously broken for midi. afaict that relates to the -rt only. vanilla might be ok. it has been suggested that the preemptive-BKL changes made after .26-rc2 are the suspect subject. good news are that .27 will have this fixed or at least goes a few steps into an acceptable solution regarding -rt === Subject: Re: [LAU] [Hardware] Small linux box / sampler? You could have a look at the motherboards and cases here: http://www.mini-itx.com/ If you know a bit of programming, you might be able to hack this case: http://www.mini-itx.com/store/?c=3#m200-lcd === Subject: [LAU] realtime kernel on ubuntu 8.04 I installed "linux-rt" which seems to have provided some kind of realtime kernel: atte@vestbjerg:~$ uname -r 2.6.24-19-rt === Subject: Re: [LAU] asoundrc question > Is it possible to create a fake mixer in .asoundrc in order to fool Java > into thinking my Delta 66 has an on-board mixer ? > An example would be most helpful. :) This will make the first soundcards-mixer the default: ctl.default { type hw card 0 } This makes the second soundcards pcm-device the default: pcm.default { type hw card 1 } So you get an alsa default device which puts the sound to the second hardwa= re=20 while using the first hardwares mixer for the apps that "need" a mixer. Of course you can change the numbers. Might even point to a mixer of a virt= ual=20 soundcard afaik. === Subject: loops for ffmpeg CLI in python I am a begineer in Python. Actually I am encoding video files with differen= t bitrates using ffmpeg CLI. I wanted to ask you that how can I make loops = so that I can vary the bitrates in the CLI of ffmpeg?? I want to bulid a loop for the command below where i just want to vary the = -vb parameter i.e. 10M.=20 =20 ffmpeg.exe -i sample.avi -y -f h264 -vb 10M -r 25 encoded_1000_0-05.h264 =20 I have tried to make simple loops but it doest work:( ===