Linux-audio-user July 2005

linux-audio-user@lists.linuxaudio.org

191 participants
230 discussions

[linux-audio-user] Mix Minus with AM-Pitchshift; a study in Linux Audio usability

by R Parker

Hi, I've begun working alot more with LADSPA plugins and have found it very easy to achieve satisfying results. www.iaxs.net/~rparker/legaladvice.ogg Despite the relative ease with which my partners Bill, Dana and I were able to achieve our final product, legaladvice.ogg, the real acid test is whether or not anyone thinks the final product is effective. Give it a listen and let me know what you think. A JK Audio RemoteMix3, phone dialer with audio i/o, is used to build a mix minus. I imagine you can google for that. We run a Neumann M 147 tube mic into an Avalon Vt 737sp which inserts into a channel on the Tascam DM-24. We send that channel direct (it's not assigned to the DM-24 stereo buss) to Ardour where Barry's Satan Maximizer and the Steve Harris AM-Pitchshift are inserted pre-fader. The effected return from Ardour routes back to the RemoteMix3 via an Aux bus (DAC/analog send) from a new DM-24 channel that is not assigned to stereo buss. That pitchshifted/maximized Aux send replaces my telephone and is mixed into a conversation with another caller in the RemoteMix3. The RemoteMix3 analog output routes to the DM-24 in a channel that is assigned to the stereo bus where we finally get to monitor it. A wireless lav would have enabled us to wonder around the studio while monitoring phone conversations on the Alesis M1 Active monitors. Definitely a bonus for the multitasking professional that doesn't want to be restricted to a single task because of some stupid wire. ron __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 5 months

[linux-audio-user] [LAM] A piece with singing...

by James Stone

Uh oh.. I think this could be quite embarrassing! A piece I recorded today, with me singing (slightly off-key at times, I notice now). The production is awful: the singing was recorded thru an ultra-cheap computer microphone, the organ improvisation is terrible (by me) etc. etc. Still, the software was very cool: Seq24 to sequence the piano (Trachtman's non-free ($5) Steinway soundfont),running in qsynth.. all recorded into Ardour. The hammond organ was from the FluidR3 soundfont, and played live (like you would never have guessed!). Also no mastering but I am not sure it is worth it when the quality is so bad to start with. Anyway, I would be interested to know what others think.... http://www.jmstone.dsl.pipex.com/rs232-shiningsun.ogg James

19 years, 5 months

[linux-audio-user] My first track made with Linux

by Cesare Marilungo

http://www.cesaremarilungo.com/download/music/thedifference.ogg It's still unfinished. Drums are just a placeholder. Tools/instruments: - Muse (for sequencing and for recording audio tracks) - Zynaddsubfx - Some ladspa effects - tr808 soundfont. If you like it there are four albums and some sparse tracks of mine to download for free, here: http://www.cesaremarilungo.com/music/listen_to_my_music/ Mostly made with macs. In mp3 (sorry). Now I've totally switched to linux. I just still need to find a good guitar amp simulator. c.

19 years, 5 months

[linux-audio-user] ams + list dead?

by Atte André Jensen

Hi I've stated using alsa modular synth, but there's been no traffic on the list for a couple of days except my single post. Also It seems the latest version of ams is almost one year old. Is this project dead? Or it's user base small? Anybody here using it, for example? If so, maybe someone here can supply me with a working example of the analogue driver. It doesn't seem to work in any predictable way... -- Peace, love & harmony Atte http://www.atte.dk

19 years, 5 months

[linux-audio-user] Sound Blaster Live External USB does not work with jack

by Peter Butler

I'm having problems using jack with the Sound Blaster Live 24 Bit External USB sound card. I can use non-jack ALSA applications (e.g. xmms) with the card with no problems. This machine has the following sound cards: - Onboard Intel (intel8x0) hw:0 - MOTU USB MIDI Interface (snd-usb-audio) - Sound Blaster Live 24 Bit External (snd-usb-audio) hw:1 Starting jack like this: jackd -v -d alsa -d hw:1 Results in the following error message: registered builtin port type 32 bit float mono audio loading driver .. new client: alsa_pcm, id = 1 type 1 @ 0x8056860 fd = -1 creating alsa driver ... hw:1|hw:1|1024|2|48000|0|0|nomon|swmeter|-|32bit control device hw:1 configuring for 48000Hz, period = 1024 frames, buffer = 2 periods Couldn't open hw:1 for 32bit samples trying 24bit instead Couldn't open hw:1 for 32bit samples trying 24bit instead new buffer size 1024 registered port alsa_pcm:capture_1, offset = 4096 registered port alsa_pcm:capture_2, offset = 8192 registered port alsa_pcm:playback_1, offset = 0 registered port alsa_pcm:playback_2, offset = 0 registered port alsa_pcm:playback_3, offset = 0 registered port alsa_pcm:playback_4, offset = 0 registered port alsa_pcm:playback_5, offset = 0 registered port alsa_pcm:playback_6, offset = 0 ++ jack_rechain_graph(): client alsa_pcm: internal client, execution_order=0. -- jack_rechain_graph() could not start playback (Broken pipe) DRIVER NT: could not start driver cannot start driver When I start jack using hw:0 everything works well. I can use the MIDI controller to play qsynth (and other instruments) through the onboard Intel sound. However the sound quality and noise from this card is terrible, so I want to use the external USB card. I am running ALSA driver version 1.0.9rc4a on Ubuntu Linux (kernel 2.6.10-5-686). Jack version is 0.99.0. I compiled the ALSA driver and libraries from source. Other (possibly relevant) information is below. If anyone can help me out with this I would be very grateful. Thanks in advance Peter Butler # cat /dev/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.9rc4a emulation code) Kernel: Linux laptop 2.6.10-5-686 #1 Fri Jun 24 17:33:34 UTC 2005 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: Intel 82801CA-ICH3 with AD1881A at 0x1c00, irq 11 Creative Technology SB Live! 24-bit External at usb-0000:00:1d.1-1.3, full spee MOTU Fastlane at usb-0000:00:1d.1-1.4, full speed Audio devices: NOT ENABLED IN CONFIG Synth devices: NOT ENABLED IN CONFIG Midi devices: 2: Fastlane Timers: 7: system timer Mixers: NOT ENABLED IN CONFIG # cat /etc/modprobe.d/sound options snd device_mode=0666 alias char-major-116 snd alias char-major-14 soundcore # On-board Intel Sound Card alias snd-card-0 snd-intel8x0 options snd-intel8x0 index=0 install snd-intel8x0 /sbin/modprobe --ignore-install snd-intel8x0 && \ /usr/sbin/alsactl restore >/dev/null 2>&1 || : remove snd-intel8x0 { /usr/sbin/alsactl store >/dev/null 2>&1 || : ; }; \ /sbin/modprobe -r --ignore-remove snd-intel8x0 alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss # USB Sound Blaster Live Sound Card alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 alias sound-slot-1 snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss options snd cards_limit=2 # cat /etc/asound.conf pcm.intel { type hw card 0 } pcm.dmix_intel { type dmix ipc_key 1024 ## needs to be a power of 2 slave { pcm "intel" period_time 0 period_size 1024 buffer_size 8192 rate 48000 ## not necessary } } pcm.dsnoop_intel { type dsnoop ipc_key 2048 slave { pcm "intel" } } pcm.asym_intel { type asym playback.pcm "dmix_intel" capture.pcm "dsnoop_intel" } pcm.pasym_intel { type plug slave.pcm "asym_intel" } pcm.dsp_intel { type plug slave.pcm "asym_intel" } ctl.dsp_intel { type hw card 0 } pcm.sblive { type hw card 1 } pcm.dmix_sblive { type dmix ipc_key 1024 slave { pcm "sblive" period_time 0 period_size 4096 buffer_size 32768 rate 48000 } } pcm.dsnoop_sblive { type dsnoop ipc_key 2048 slave { pcm "sblive" } } pcm.asym_sblive { type asym playback.pcm "dmix_sblive" capture.pcm "dsnoop_sblive" } pcm.pasym_sblive { type plug slave.pcm "asym_sblive" } pcm.dsp_sblive { type plug slave.pcm "asym_sblive" } ctl.dsp_sblive { type hw card 1 } pcm.!default { type plug slave.pcm "asym_sblive" } ctl.!default { type hw card 1 }

19 years, 5 months

[linux-audio-user] [ANN] Musix Live CD

by Lisandro Damián Nicanor Pérez Meyer

Musix [1] it's a Live CD, DemuDi like, specially prepared for the spanish speaking people. It has been arranged by a member of Lau-es. So, if there's anyone here who speaks spanish (or wants to give it a try), it can be downloaded in [2] and [3] [1] http://www.musix.org.ar/ [2] https://200.32.4.29/utiles/torrent [3] https://e.ututo.org.ar/utiles/torrent/ Cheers, Damian.- Anyone who says he can see through women is missing a lot. -- Groucho Marx

19 years, 5 months

[linux-audio-user] newbie with muse, ams and om

by Atte André Jensen

Hi I'm trying to sequence a few tracks in muse using ams and/or om. I started two instances of ams with "ams -j" and loaded a different patch in each. I also started om and om_gtk, and loaded two different patches. Audio out from the synths are routed to the alsa_pcm input. Now, I have the following problems: 1) If I select the two different om patches output is distorted when *both* patches are sounding. If one is ams an the other om sound is also distorted, but if I route it through jamin the sound is ok. Routing both om synths through jamin is still distorting... Shouldn't it be possible to just route multible synths the the alsa_pcm or should they be mixed somewhere? 2) I only see one output named "ams" in the O-Port in the track list. In the "Midi Port/Soft Synth" dialog how ever, I see 7 grayed out occurences of the name "ams". Am I doing something wrong or isn't it possible to run multible instances of ams from midi? I didn't include technical information about my system or software version, since I suspect that I'm simply doing something wrong... -- .''`. Peace, love & harmony : :' : Atte `. `' `- http://www.atte.dk

19 years, 5 months

[linux-audio-user] New Sound Subsystem Needed For Qt4/KDE4

by Mark Constable

Apparently aRts could be removed from KDE4 CVS HEAD RSN. As far as I know there is no decision on an official replacement yet. I have no say in the matter but I would like to see something like qjackctl ported to KDE4 and usable from within kcontrol, basically, Jack embedded into the system from the ground up. My general question, to those who know far more than me, is just what would be the ideal sound subsystem for KDE in 2006 seeing there is a clean slate and an opportunity to get it right from the get go ? --markc

19 years, 5 months

[linux-audio-user] Noise reduction plugins

by Iain Duncan

Are there any decent linux noise reduction plugins? Something to just clean up some old recordings with a lot of tape noise? Thanks Iain

19 years, 5 months

[linux-audio-user] Managing instruments

by rheise＠it.uts.edu.au

What I'm dreaming of is configuring my sequencer to output to just one alsa sequencer port. On that port is one "logical" bank of all the "logical" instruments I use (just instrument names, only). The piece of software listening on this port is to associate each instrument with a synthesiser, and most importantly, can be reconfigured by selecting a different profile if I unplug my laptop from an external synth and am forced to use a software synth temporarily. I am a roaming midi user. Are there any solutions already to this problem? What do you guys do to manage your instruments? Ryan

19 years, 5 months

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Linux-audio-user July 2005