Linux-audio-user November 2005

linux-audio-user@lists.linuxaudio.org

147 participants
156 discussions

by Dave Phillips

Greetings: I've uploaded a new song: http://linux-sound.org/keepmesatisfied.ogg And I've revised and updated my "Music Made With Ardour" page: http://linux-sound.org/ardour-music.html Lyrics and license for the new tune are on the Ardour music page, if anyone's interested. Enjoy! Best, dp

20 years

[linux-audio-user] Even more music

by james＠dis-dot-dat.net

http://blog.dis-dot-dat.net/2005/11/more-music.html And now I have to get some work done. James -- "I'd crawl over an acre of 'Visual This++' and 'Integrated Development That' to get to gcc, Emacs, and gdb. Thank you." (By Vance Petree, Virginia Power)

20 years

[linux-audio-user] [ANN] jack_convolve-0.0.13/libconvolve-0.0.6

by Florian Schmidt

Hi, this is a new release of jack_convolve/libconvolve, the simple commandline based, convolution engine for JACK. New features are - LASH support: have jack_convolve be restored with the exact same commandline parameters when you restore your LASH session. Comment out two lines in the makefile if you don't have LASH installed. - Pseudo multichannel support (--pseudo-multi): This creates an input port for each responsefile channel, instead of only one input per response. While technically i.e. a stereo response file is a response of a mono impulse recorded with stereo equipment it many cases it might still sound good to convolve i.e. a stereo sum's left and right channel with the left and right channel of a response file individually :) And what sounds good is The Right Thing To Do. - Output mixing now optional (--mix-outputs): previously all outputs from the same channel number of different responses (if multiple responses were loaded) have been mixed together to save on some IFFT's. This is now optional. You can load several responses with varying channel counts, if your heart desires. Note that in mix-outputs mode all response files still need to have the same channel count. libconvolve had to be changed somewhat for these features, thus the upgrade. Note that dssi_convolve will not link against this libconvolve. I consider dssi_convolve broken atm anyways (still missing gui, etc). Will take another while to get that fixed up. Grab them at: http://www.affenbande.org/~tapas/jack_convolve/ Bug reports welcome, especially with odd combinations of response files with different channel counts :) Flo -- Palimm Palimm! http://tapas.affenbande.org

20 years

[linux-audio-user] Hello !!

by lilli chiffon

Hi everybody, I'm a new one on this list, excuse-me for my english it's not my natural languange (I've got many red line under this word) This message is a test and a hello. see you. P'tit Louis

20 years

Re: [linux-audio-user] [ANN] jack_capture v0.0.1

by Kjetil S. Matheussen

tim hall: > On Friday 25 November 2005 19:26, Kjetil S. Matheussen was like: > > In addition, as you say, timemachine use this (and I > > really mean it) stupid w64 fileformat by default, and I very seldom > > remember to use the -f flag to override that. Instead of doing all that, > > now I can just write "jack_capture", and it starts recording immmediately. > > I use a custom menu entry. Listen, jack_capture is not a competitor for timemachine. Its a program that fills a hole that timemachine previously filled the best. Timemachine is, as Steve Harris says, a simulation of: "a minidisc recorder in my studio running in a mode where when you pressed record it wrote the last 10 seconds of audio to the disk and then caught up to realtime and kept recording." --

20 years

Re: [linux-audio-user] [ANN] jack_capture v0.0.1

by Kjetil S. Matheussen

james: > On Fri, 25 Nov, 2005 at 12:34AM -0800, Kjetil S. Matheussen spake thus: > > > > http://www.notam02.no/arkiv/src/ > > > > I normally use timemachine for recording like this - how does this > differ? timemachine is a different kind of program. Its a GUI thing that is ment to be running all the time. This is just a quick tool to get the sound going into your loudspeakers as quickly as possible into disk. It happens quite a lot when I play with PD or jamin, for example. What happened previously, when I had something cool going, was that I had to start timemachine (which name does not start with jack, so I allways use a lot of time remembering its name), connect the correct jack ports into it in qjackctl, and press the recording button. In addition, as you say, timemachine use this (and I really mean it) stupid w64 fileformat by default, and I very seldom remember to use the -f flag to override that. Instead of doing all that, now I can just write "jack_capture", and it starts recording immmediately. --

20 years

[linux-audio-user] Text to MP3

by Terrence

Hello, all, Does anyone know if there's a way to convert text files to MP3s in Linux? I'd like to be able to take some of my text files and make audio out of them so I can listen to them while walking. Thanks, Terrence

20 years

Re: [linux-audio-user] Maudio audiophile USB

by Ronny

Hi Paul and All, > if its like aplay, use the -v argument to get a verbose listing of the > h/w config that the device ends up with. Unfortunately this device is Big Endian so I need to use plughw also using a play: $ aplay -D plughw:2 /usr/share/sounds/shutdown1.wav -v Playing WAVE '/usr/share/sounds/shutdown1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Linear conversion PCM (S16_BE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Hardware PCM card 2 'Audiophile USB (tm)' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 > then try to replicate that using the hw:N device with JACK (not the > plughw device). JACK can use plughw devices, but it is intended to > work better with a hw device. >> Problems rise when I try using Jack. >> even if I managed to get it starting with jackd -dalsa -dplughw:2,0 >> -r44100 -p1024 -n4 -Pplughw:2,0 -S -o2 > this is a very very odd way to start JACK. the -Pplughw:2,0 seems > redundant. You're right, I'll try to start with a better setup using the parameters above, first block, and using plughw because BE, jackd -v -dalsa -dplughw:2 -r44100 -p4096 -n4 -S the problem is that I get the card playing few milliseconds of audio continuously even without using any player connected to Jack. A sort of tic -tac - tic -tac .... if I reduce p to 1024 then it seems to work fine but the quality is still lousy. See below part of the output, any suggestion is most welcome. $ jackd -v -dalsa -dplughw:2 -r44100 -p4096 -n4 -S getting driver descriptor from /usr/lib/jack/jack_dummy.so getting driver descriptor from /usr/lib/jack/jack_alsa.so getting driver descriptor from /usr/lib/jack/jack_oss.so jackd 0.100.1 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 System V SHM support. server `default' registered loading driver .. apparent rate = 44100 creating alsa driver ... plughw:2|plughw:2|4096|4|44100|0|0|nomon| swmeter|-|16bit control device hw:2 configuring for 44100Hz, period = 4096 frames, buffer = 4 periods You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer. Usually the name of the hardware device that corresponds to the first soun nperiods = 4 for capture registered builtin port type 32 bit float mono audio new client: alsa_pcm, id = 1 type 1 @ 0x8746a78 fd = -1 You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer. Usually the name of the hardware device that corresponds to the first soun nperiods = 4 for playback new buffer size 4096 registered port alsa_pcm:capture_1, offset = 16384 registered port alsa_pcm:capture_2, offset = 32768 registered port alsa_pcm:playback_1, offset = 0 registered port alsa_pcm:playback_2, offset = 0 ++ jack_rechain_graph(): client alsa_pcm: internal client, execution_order=0. -- jack_rechain_graph() 4167 waiting for signals late driver wakeup: nframes to process = 8192. load = 0.2853 max usecs: 530.000, spare = 92349.000 ..... Thank You in advance Ronny

20 years

[linux-audio-user] Hello List

by Sean Edwards

Greetings! My name is Sean Edwards. I have used Linux on my desktop at home since 1997. My computer career started in 1993, and I have done System Admin work since earning a Novell Certification in 1995. In 2002, my career became Linux focused on a full-time basis. I have done both contract and full-time work in Chicago, Dublin Ireland, and now Omaha. I learned to play guitar 30 years ago, and took a break from it for 10 years when I started my family. Before the kids came along, I used to do sequencing on a 486 DOS computer, connected to external MIDI equipment. For years, the only Linux audio I used was for ripping and creating CD's. Last year, I discovered the Planet CCRMA project, and was a lurker on that list for several months. This Autumn, I decided to get back into computer music and I find what is available for Linux to be overwhelming! Here is my equipment list: Yamaha TX81Z (2) Alesis MIDI Data Disk Alesis HR 16 Drum Machine Yamaha MJC8 MIDI Junction Controller Alesis 1622 Mixer A.R.T. MultiVerb Behringer V-Amp Pro Behringer UltraTwin The computer is an AMD K7 500Mhz, 768MB RAM, VooDoo 3D AGP Video Card, and a 20GB hard drive. My previous employer provided me with a PowerBook G4 with 1GB RAM, and Garage Band could not play more than 3 synthesized voices simultaneously. My 6 year old desktop does more music with Linux than Garage Band on a newer PowerBook! The software I mostly use: Qjackctl Rosegarden4 Qsynth Hydrogen Ecasound -=Sean Edwards=- __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

20 years

[linux-audio-user] Alsa support for ZOOM G2.u1 and generic usb audio devices in general

by Cesare

Hi, I've just bought myself (€130, in Italy) a Zoom 2G.u1 guitar effect pedal (http://www.zoom.co.jp/english/products/g21u/index.php). It is a very good sounding effect if you're on a budget, like me, and with my strat plugged in it sounds like this: http://www.cesaremarilungo.com/download/music/Hope_IN_PROGRESS_20051125.mp3 What makes it interesting is that it has a usb interface so it works like an audio interface. You can listen to the mix from the DAW from its headphone jack along with the direct output of the effected guitar (no additional latency). The sampling is done at 24bit/96kHz. I bought it to complete my transition to linux and I would really like to use it with jackd. In windows and macos x it works without any driver (it is seen as a generic usb audio device), and in fact through USB it is limited to 16bit/44.1kHz. The question is: how can I get it to work with alsa, and therefore jackd? I compiled my kernel (2.6.13) without the usb-audio support, as I read somewhere that it can interfere with alsa usb implementation. And I guess that recompiling the kernel with it wouldn't allow me to use it with jackd anyway. Thanks in advance. c.

20 years

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Linux-audio-user November 2005