Linux-audio-user January 2009

linux-audio-user@lists.linuxaudio.org

165 participants
173 discussions

[LAU] The Drummer's Gigsaw: DANCE, DISCO and ROCK.

by Philippe Hezaine

Hi all, (Sorry for the cross-posting.) Here come some DANCE & DISCO patterns put into the TECHNO book: http://philippe.hezaine.free.fr/spip.php?article50 And there the first part of ROCK book: http://philippe.hezaine.free.fr/spip.php?article42 Compared with the previous releases you'll find: - For the .ly files: the bar numbers which make easier the reading. - For the midi files: You'll be able to see the instrument names printed in each track of the sequencer. Before a general upgrade of the existing work, if you have some remarks, criticisms or suggestions i will be happy to hear them. Cheers. -- Phil. Superbonus-Project (Site principal) <http://superbonus.project.free.fr> Superbonus-Project (Plate-forme d'échange): <http://philippe.hezaine.free.fr>

17 years, 1 month

[LAU] lost sound (it had been working)

by Reid Vail

Hello Group - I'm really confused about what's gone wrong with my sound. Total Silence even though it was working less than two weeks ago after an upgrade to 8.10. User error?, or a bad patch maybe...can you help? I'm running ubuntu 8.10 on a 64-bit machine _______________________________________________________________________ I have done some research and have been following the steps I found at this posting: http://ubuntuforums.org/showthread.php?t=205449&highlight=comprehensive+sou… First step is this: results appear normal. reid@linux-rv:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: V8237 [VIA 8237], device 0: VIA 8237 [VIA 8237] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 0: V8237 [VIA 8237], device 1: VIA 8237 [VIA 8237] Subdevices: 1/1 Subdevice #0: subdevice #0 reid@linux-rv:~$ _____________________________________________________________________________ Second step: reid@linux-rv:~$ lspci -v : : 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 60) Subsystem: ASUSTeK Computer Inc. Device 80b0 Flags: medium devsel, IRQ 22 I/O ports at c800 [size=256] Capabilities: <access denied> Kernel driver in use: VIA 82xx Audio Kernel modules: snd-via82xx Looks Ok, I think ______________________________________________________________________ Third step: reid@linux-rv:~$ sudo modprobe snd-via-82xx [sudo] password for reid: FATAL: Module snd_via_82xx not found. reid@linux-rv:~$ That's not good, or expected. Fourth step (sorry about the screwy formatting): reid@linux-rv:~$ lsmod|grep snd snd_via82xx 36904 3 snd_via82xx_modem 21900 0 gameport 21776 1 snd_via82xx snd_seq_dummy 11524 0 snd_ac97_codec 133080 2 snd_via82xx,snd_via82xx_modem ac97_bus 10368 1 snd_ac97_codec snd_pcm_oss 52608 0 snd_mixer_oss 25088 1 snd_pcm_oss snd_pcm 99208 4 snd_via82xx,snd_via82xx_modem,snd_ac97_codec,snd_pcm_oss snd_seq_oss 42368 0 snd_mpu401_uart 16768 1 snd_via82xx snd_seq_midi 15872 0 snd_rawmidi 34176 2 snd_mpu401_uart,snd_seq_midi snd_seq_midi_event 16768 2 snd_seq_oss,snd_seq_midi snd_seq 67168 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 34320 2 snd_pcm,snd_seq snd_seq_device 16404 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 79432 18 snd_via82xx,snd_via82xx_modem,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_mpu401_uart,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 17680 3 snd_via82xx,snd_via82xx_modem,snd_pcm soundcore 16800 1 snd So...I'm not sure what to look for here.... Any thoughts would be appreciated. Reid

17 years, 1 month

[LAU] placement.

by alex stone

Ok, this might be a bit of curly question, and as i don't know if this is possible, either valid or not. The subject is placement, and pertains to orchestral recording. (My own work composed within the box with linuxsampler, from midi in RG, and recorded in Ardour.) I'd like to place my instruments as close as possible to an orchestral setup, in terms of recorded sound. That is, once i've recorded, i'd like to use convolution and other tools to 'correctly' place instruments within the overall soundscape. example: With the listener sitting 10 metres back from the stage, and facing the conductor (central) my 1st violins are on the listener's left. Those first violins occupy a portion of the overall soundscape from a point approximately 2 metres to the left of the conductor, to an outside left position, approximately 10 metres from the conductor, and with 8 desks (2 players per desk) about 4 metres deep at the section's deepest point, in the shape of a wedge, more or less. That's the pan width of the section. Now as i understand it, a metre represents approximately 3ms, so calculating the leading edge of the section across the stage as 'zero', the first violin players the furthest in depth from the front of the stage, should, in theory, (and i know this is approximate only, as i sat as a player in orchestras for some years, and understand the instinctive timing compensation that goes on) play about 12ms later than those at the front. Using the ears, and experimenting, this actually translates as about 6ms, before the sound becomes unrealistic, using layered violin samples, both small section and solo. (highly subjective i know, but i only have my own experience as a player and composer to fall back on here.) A violin has it's own unique characteristics in distribution of sound emanating from the instrument. The player sits facing the conductor, and the bulk of the overall sound goes up, at an angle, at more or less 30degrees towards the ceiling to a 'point' equivalent to almost directly over the listener's right shoulder. Naturally the listener 'hears' the direct sound most prominently, (both with ears, and the 'visual perception' he gains from listening with his eyes.) Secondly, the violin also sounds, to a lesser degree, downwards, and in varying proportions, in a reasonably 'spherical' sound creation model, with the possible exception of the sound hitting the player's body, and those in his immediate vicinity. (and other objects, like stands, sheet music, etc, all playing a part too.) I've experimented with this quite a bit, and the best result seems to come from a somewhat inadequate, but acceptable, computational model based on using, you guessed it, the orchestral experience ears. So i take one 'hall' impulse, and apply it to varying degrees, mixed with as precise a pan model as possible (and i use multiple desks to layer with,more or less, so there's a reasonably accurate depiction of a pan placed section, instead of the usual pan sample model of either shifting the section with a stereo pan, or the inadequate right channel down, left channel up method.) to make this more complicated (not by intent, i assure you), i'm attempting to add a degree of pseudo mike bleed, from my 1st violins, into the cellos sitting deeper on the stage, and in reduced amounts to the violas and second violins sitting on the other side of the digital stage. All of this is with the intent of getting as as lifelike a sound as possible from my digital orchestra. The questions: In terms of convolution, , can i 'split' a convolution impulse with some sort of software device, as to emulate the varying degrees of spherical sound from instruments as described above? So, one impulse (I use Jconv by default, as it does a great job, far better than most gui bloated offerings in the commercial world) that can be, by way of sends, and returns, be 'split' or manipulated not only in terms of length of impulse, but fed as 'panned' so as to put more impulse 'up', less impulse 'down' and just a twitch of impulse 'forward' of the player, with near enough to none on the sound going back into the player. I've written this rather clumsily, but i hope some of you experts may understand what i'm trying to achieve here. Can the impulse be split down it's middle, separating left from right, aurally speaking, and if this is possible, can i split the impulse into 'wedges' emulating that sphere i wrote of, more or less? if there's a way to do this, then i'm all ears, as my mike bleed experiments suffer from a 'generic' impulse per section affecting everything to the same degree, including the instruments bled in. I should note here, this is not about gain, but a wedge of impulse, cut out of the overall chunk, that represents a 'window' or pan section of the whole. This might seem somewhat pedantic in recording terms, but i'm trying to build a model, per 'hall' and create half a dozen templates to represent different size ensembles in different halls. So my small string section hall template might be a small church or performance room, like a library, or dancing hall. I would then only need to be creative with where i put my 'human' soloists, as emulators of slightly different interpretations of where note start and finish, velocity variation, etc....., and not have to reconstruct a convoluted model each time, for......my particular orchestra. (tongue firmly in cheek here.) Any help would be appreciated. I'm open to suggestions of placement convolution models, etc, but would prefer to use Jconv, as it works all day every day, and would be a constant across all templates. I suppose an analogy for the chunk of impulse idea would be to stretch a ribbon across a stage, and cut a metre out of the middle. That metre would be the bit i'd use, as a portion of the whole, in an aural soundscape, to manipulate, or place, instruments, to a finer degree, in the attempt to create a more realistic '3d' effect for the listener. That metre along with other cut out sections of the impulse soundscape could help me introduce a more....'human' element to a layered instrument section. I still rue the day orchestral sample devs decided to record sections at a time. This would have been so much easier if they'd recorded a number of desks for each section instead. Alex. p.s. I'm having a modest degree of success using Ardour sends, as a mike bleed template. Lot of work, and a lot of sends, but it's slowly coming together, and has reached the stage where i no longer think it's impossible.....

17 years, 1 month

[LAU] [ANN] lv2dynparam1-2

by Nedko Arnaudov

lv2dynparam is LV2 extension for dynamic parameters. The extension consists of a header describing the extension interface and libraries, one for plugins and one for hosts, to expose functionality in more usable, from programmer point of view, interface. Changes since version 1: * host library: API is refactored, the new API is NOT compatible with the version 1 API * host library: support for dynparam automation * host library: support for dynparam parameter save/restore Project homepage: http://home.gna.org/lv2dynparam/ Get tarball from here: https://gna.org/files/?group=lv2dynparam -- Nedko Arnaudov <GnuPG KeyID: DE1716B0>

17 years, 1 month

[LAU] Jconv

by alex stone

Got a bit of a challenge, that is sure to be user error somewhere. I'm building Jconv from source, and get the following error when trying to 'make': impdata.cc: In member function 'int Impdata::sf_open_read(const char*)': impdata.cc:264: error: 'SFC_WAVEX_GET_AMBISONIC' was not declared in this scope impdata.cc:264: error: 'SF_AMBISONIC_B_FORMAT' was not declared in this scope impdata.cc: In member function 'int Impdata::sf_open_write(const char*)': impdata.cc:306: error: 'SFC_WAVEX_SET_AMBISONIC' was not declared in this scope impdata.cc:306: error: 'SF_AMBISONIC_B_FORMAT' was not declared in this scope make: *** [impdata.o] Error 1 Any clues? Alex.

17 years, 1 month

[LAU] Some fun for drummers, and those who like them

by Ken Restivo

A drum-driven show last night, big time: http://www.archive.org/details/AudiobrailleLive2009-01-16 Super fun. Not too much Linux going on there, just my usual Fluidsynth, AZR3-JACK, and AMS stuff, with a little WhySynth and NekoString too. The drummers were driving the bus all night, and, IMHO, they did a fine job of it. Example here: http://www.archive.org/download/AudiobrailleLive2009-01-16/audiobraille-200… I'm starting to get quicker at editing these shows in Ardour, so I can post them the morning after. -ken

17 years, 1 month

Re: [LAU] How to do an online album release?

by Emanuel Rumpf

2009/1/19 Tapani Sysimetsä <tapani.sysimetsa(a)rokki.net>: > > Here's some quick notes on Equal Dreams: > Thank you Tapani. I've extended the page with the info you provided. The wiki is public. In case you'd like to add or change anything press the edit button. - Emanuel

17 years, 1 month

[LAU] QJackMMC: New Qt based JACK application released!

by Alex Montgomery

Hello all, We're pleased to announce the release of a new version of jackctlmmc and a new Qt based graphical version called QJackMMC. The main page including download links and documentation is here: http://jackctlmmc.sourceforge.net/ In brief, QJackMMC is a Qt based program that can connect to a device or program that emits Midi Machine Code (MMC) and allow it to drive JACK transport, which in turn can control other programs. JackCtlMMC is a slightly simpler command-line version of QJackMMC. You might need such a tool if you have hard-disk recorders (HDRs) or other external MIDI compliant devices that are capable of sending out MMC to keep other devices in sync. You might have a multi-track recorder and you want to be able to start, stop, or fast-forward JACK-based programs such as Rosegarden, Hydrogen, and Ardour. Enjoy, -- Alex

17 years, 1 month

[LAU] was loomer VST - BPM-Delay, Bounty and stuff

by Hartmut Noack

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good news everyone... how could I be blind enough to ignore the fantastic, cool and promising-a-great-future-for-free-FX-plugins CALF-Collection?? The Calf Vintage Delay completely qualifies as a perfect fullfillment for all the wishes I issued here regarding a simple-to-work-with, good sounding and configurable delay with BPM-parameter. Now I'm only searching for a way to send the promised bounty to Krzysztof Foltman and Thor Harald Johansen. They have no donation-button on their site and I am not sure about sourceforges system so I left a message at sf... If anybody from the Calf-people is tracking this list, please let me know :-) best regs HZN -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.9 (GNU/Linux) Comment: Using GnuPG with SUSE - http://enigmail.mozdev.org iEYEARECAAYFAklzwEMACgkQ1Aecwva1SWNLTgCfZryTlSDde11he4+toWxV90s4 WPwAnAn418QjdGxGsnQdE0SSHVUmESjP =gAq2 -----END PGP SIGNATURE-----

17 years, 1 month

Re: [LAU] [PD] : PD sync with JACK (linux)

by Rich E

Sorry for the double post, but this question is sort of directly in between the two mailling lists, pd and linux audio users.... Anyone been able to get seq24 to start from pd? I've tried numerous times to use seq24 for sequencing harmonies and whatnot, but it is always a pain getting it to start at the same time that I want my things to start in pd. In seq24, the settings are 'jack transport' and 'live mode' enabled. I then try to start the jack transport from the pd external or in qjackctl and seq24 doesn't do anything. On Sat, Sep 29, 2007 at 3:57 PM, Tim Blechmann <tim(a)klingt.org> wrote: > i once write a small external to query jacktransport ... maybe it is > useful for your purposes? > > it is in the pd cvs under externals/tb/jacktransport > > best, tim > > On Sat, 2007-09-29 at 13:20 +0200, Mysth-R wrote: >> Hi, >> I am new on this list. I tried to find some informations about syncing >> PD and Jack transport/BPM in the archive list but I am not >> satisfied :D >> I am making an Audio sequencer with PD, and I would like to use it >> with other linux softwares (seq24, hydrogens, freewheeling,...). >> So I need to synchronise PD and Jack with MIDI or OSC; >> I saw on the archive list some module like midirealtimein but it >> doesn't seems to work for me. Only receive zeros. >> Today I am trying to use jack.tools, to synchronise with OSC >> ( jack.osc). I can send /start /stop messages, but I don't understand >> how I can send transport informations, bpm ... >> >> Can you help me / give me some informations ? >> Thank you for your help >> cheers, >> >> Erwan >> >> -- >> {^_^} Mysth-R {^_^} >> >> http://myspace.com/mysthr >> http://myspace.com/aideauditive >> _______________________________________________ >> PD-list(a)iem.at mailing list >> UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list > -- > tim(a)klingt.org ICQ: 96771783 > http://tim.klingt.org > > The aim of education is the knowledge, not of facts, but of values > William S. Burroughs > > _______________________________________________ > PD-list(a)iem.at mailing list > UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list > >

17 years, 1 month

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Linux-audio-user January 2009