Linux-audio-user February 2007

linux-audio-user@lists.linuxaudio.org

163 participants
149 discussions

[linux-audio-user] Re: Converting IEEE FLOAT WAV's to ... anything?

by Kjetil S. Matheussen

Ken Restivo: >> The only tool I've found so for that can handle IEEE FLOAT WAV's is >> oggenc. Yay Vorbis. But, the FLAC tools, even though they're from the >> same developers as Ogg Vorbis, dies horribly, with ERROR: unsupported >> compression type 3 I think IEEE floats can't be used in flac, because its a lossless compressor and that it gurantees not to alter the data. Floats are probably too hard (probably impossible?) to handle to sattisfy requirements. >> >> ---------------------------------------- >> Sample Rate : 48000 >> Frames : 14277625 >> Channels : 2 >> Format : 0x00010006 >> Sections : 1 >> Seekable : TRUE >> Duration : 00:04:57.450 >> Signal Max : 0.724668 (-2.80 dB) >> >> This is the format that jack_capture produces. It supposedly has I chose this format as default because its the most common one for audio processing. >> options to save in other bit-depths, which I tried, but they didn't >> seem to do anything. >> Works here: kjetil@ttleush ~ $ jack_capture -b 16 Recording to "jack_capture_03.wav". Press <Return> or <Ctrl-C> to stop it. Please wait while writing all data to disk. (shouldn't take long) disk thread finished connection thread finished kjetil@ttleush ~ $ sfinfo jack_capture_03.wav File Name jack_capture_03.wav File Format Microsoft RIFF WAVE Format (wave) Data Format 16-bit integer (2's complement, little endian) Audio Data 278528 bytes begins at offset 44 (2c hex) 2 channels, 69632 frames Sampling Rate 44100.00 Hz Duration 1.579 seconds >> Granted, I can open these files in Audacity or Rezound or Sweep, and with the GUI convert the file to just about anything. However, I dislike GUI's, and I'd like to be able to do this with the commandline tools. Opening 20 WAV files and click-clacking around on menu options is anathaema: the whole process wants to be a 1-liner bash script really. >> > > > OK after a bit more experimentation, I solved my own problem. > > This is for converting from 32-bit 48000 to redbook CD format: > sox something-32bit.wav -t wav -w -r 44100 something-cd.wav resample -ql > You might want to be careful with sox's resampling... You probably want to listen to the results, but sndfile-resample or resample are probably safer program to use instead. > And, for making FLAC's: > sox something-32bit.wav -w -t wav - | flac - > something.flac > jack_capture actually supports flac very conveniently: "$jack_capture -f flac" > A similar thing works for making mp3's; I just pipe it to lame instead of to flac. Et voila. > I have a script called makemp3.sh that looks like this: #!/bin/sh sndfile-convert -pcm32 $1.wav $1o.wav lame $1o.wav mv $1o.wav.mp3 $1.mp3 rm $1o.wav

19 years

[linux-audio-user] Problem starting bristol with jack

by Larry Troxler

Hi, I'm using the 64 studio 1.0 distro. with the command: startBristol -jack -b3 -midi alsa I get the following, and was wondering what I might be doing wrong. larry@64studio:~$ startBristol -jack -b3 -midi alsa You may want to make bristolengine a suid-root executable spawning midi thread Could not reschedule thread to 2 parent going into idle loop connected to :0.0 (5cc430) display is 1280 by 1024 pixels Window is w 1280, h 1024, d 24, 0 0 0 Using TrueColor display masks are ff0000 ff0000 ff0000 flags are 80000000 Returning socket 3 Opened listening control socket: 5028 ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC0D0 failed: Device or resource busy Could not open the MIDI interface hw:0,0. Error opening midi device hw:0,0, exiting midi thread Initialise the hammondB3 link to bristol: 5d15b0 starting upper manual connection hostname is localhost, bristol port is 5028 connfailed opening link to engine: -1 hostname is localhost, bristol port is 5028 connfailed cleanupBristol(-4) larry@64studio:~$ parent exiting Larry Troxler

19 years

RE: [linux-audio-user] Problem starting bristol with jack

by David Baron

On Thursday 08 February 2007, linux-audio-user-request(a)music.columbia.edu wrote: > Have you tried with just the following? > > startBristol -jack -b3 > > It looks like the engine has failed to start due to the midi selection (the > default midi options are the ALSA seq so I would advise not changing that). > This causes the GUI to fail as well since the engine is not available. What > version of bristol is this? I am surprised by the error message regarding > 'suid root' as I thought this was dropped in the -69 release on sourceforge > - it should now rely on PAM RT permissions or be started as root, so the > message is superfluous and you only need suid permissions if you want to > use bristol dedicated to the audio device with RT scheduling. Just tried this. Here is the result: >startBristol -jack -b3 You may want to make bristolengine a suid-root executable spawning midi thread Could not reschedule thread to 2 parent going into idle loop flags are 8a000000 midi sequencer Returning socket 5 Opened listening control socket: 5028 Client ID = 129 Queue ID = 0 Device name did not parse, defaults 128.0 Cannot subscribe port 0 from client 128: Operation not permitted Error opening midi device /dev/midi, exiting midi thread connected to :0.0 (80f90a0) display is 1280 by 1024 pixels Window is w 1280, h 1024, d 16, 0 0 0 Using DirectColor display masks are f800 f800 f800 Initialise the hammondB3 link to bristol: 8101220 starting upper manual connection hostname is localhost, bristol port is 5028 Connected to the bristol control socket: 5 bristolengine already active 80d4ea0 80000000 0 parent exiting return - no data in buffer cleanupBristol(0 Doing it sudo skips the suid message but the rest is the same!

19 years

[linux-audio-user] jackd disables client connections

by mechko＠gmx.de

Hej, I have a problem with connecting to a running jackd, clients are not able to connect. They are definitly not running as another user as jackd itself. I even tried (only for testing) to run qjackctl as root, with no success either. I can start jackd by a shell and by qjackctl, but after starting no client is able to connect. Even giving the jackd-instance a name by the -n option, clients are not able to connect while specifying that name. There are also no other clients running that could have the same client-id as my test-clients (qjackctl itself, meterbridge, adour2). OS: Ubuntu 6.10 (Edgy), Kernel 2.6.17-10-386 Hardware: M-Audio Delta 44 jackd: 0.102.20 qjackctl: 0.2.20 I had no problems with jackd before upgrading ubuntu from 6.06 to 6.10. Adour points to a problem with ulimit and locked memory while starting, so I give the contents of limits.conf, too (this is not the cause for my troubles, I believe): mechko@almeida:~$ cat /etc/security/limits.conf | grep -v "#" @audio - rtprio 99 @audio - memlock 250000 @audio - nice -10 The message-window after pressing the Start-Button in qjackctl (verbose mode on). There should be an error-message that gives a reason why clients are not able to connect, but I can find none. //snap///////////// 12:05:39.529 Patchbay deactivated. 12:05:39.602 Statistics reset. 12:05:39.973 MIDI connection graph change. 12:05:40.011 MIDI connection change. QInputContext: no input method context available QInputContext: no input method context available 12:05:47.289 Startup script... 12:05:47.289 artsshell -q terminate can't create mcop directory Creating link /home/mechko/.kde/socket-almeida. 12:05:47.559 Startup script terminated with exit status=256. 12:05:47.561 JACK is starting... 12:05:47.563 jackd -v -R -t2000 -dalsa -dhw:0 -r44100 -p128 -n2 -S 12:05:47.570 JACK was started with PID=25103 (0x620f). getting driver descriptor from /usr/local/lib/jack/jack_alsa.so getting driver descriptor from /usr/local/lib/jack/jack_dummy.so getting driver descriptor from /usr/local/lib/jack/jack_oss.so 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 System V SHM support. server `default' registered loading driver .. apparent rate = 44100 creating alsa driver ... hw:0|hw:0|128|2|44100|0|0|nomon|swmeter|-|16bit control device hw:0 configuring for 44100Hz, period = 128 frames, buffer = 2 periods ALSA: final selected sample format for capture: 32bit little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit little-endian ALSA: use 2 periods for playback 25103 waiting for signals registered builtin port type 32 bit float mono audio registered builtin port type 8 bit raw midi clock source = system clock via gettimeofday new client: alsa_pcm, id = 1 type 1 @ 0x805a738 fd = -1 new buffer size 128 registered port alsa_pcm:capture_1, offset = 512 registered port alsa_pcm:capture_2, offset = 1024 registered port alsa_pcm:capture_3, offset = 1536 registered port alsa_pcm:capture_4, offset = 2048 registered port alsa_pcm:capture_5, offset = 2560 registered port alsa_pcm:capture_6, offset = 3072 registered port alsa_pcm:capture_7, offset = 3584 registered port alsa_pcm:capture_8, offset = 4096 registered port alsa_pcm:capture_9, offset = 4608 registered port alsa_pcm:capture_10, offset = 5120 registered port alsa_pcm:capture_11, offset = 5632 registered port alsa_pcm:capture_12, offset = 6144 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 registered port alsa_pcm:playback_7, offset = 0 registered port alsa_pcm:playback_8, offset = 0 registered port alsa_pcm:playback_9, offset = 0 registered port alsa_pcm:playback_10, offset = 0 ++ jack_rechain_graph(): client alsa_pcm: internal client, execution_order=0. -- jack_rechain_graph() load = 0.0861 max usecs: 5.000, spare = 2897.000 12:05:49.611 Could not connect to JACK server as client. Please check the messages window for more info. load = 0.3877 max usecs: 20.000, spare = 2882.000 load = 0.2800 max usecs: 5.000, spare = 2897.000 load = 0.2261 max usecs: 5.000, spare = 2897.000 load = 0.2337 max usecs: 7.000, spare = 2895.000 load = 0.2030 max usecs: 5.000, spare = 2897.000 load = 0.1876 max usecs: 5.000, spare = 2897.000 load = 0.1627 max usecs: 4.000, spare = 2898.000 load = 0.3054 max usecs: 13.000, spare = 2889.000 load = 0.3077 max usecs: 9.000, spare = 2893.000 12:05:58.362 Could not connect to JACK server as client. Please check the messages window for more info. load = 1.0153 max usecs: 50.000, spare = 2852.000 load = 0.6627 max usecs: 9.000, spare = 2893.000 load = 0.6587 max usecs: 19.000, spare = 2883.000 load = 0.4327 max usecs: 6.000, spare = 2896.000 load = 0.4231 max usecs: 12.000, spare = 2890.000 load = 0.5045 max usecs: 17.000, spare = 2885.000 load = 0.5968 max usecs: 20.000, spare = 2882.000 load = 0.4362 max usecs: 8.000, spare = 2894.000 load = 0.3215 max usecs: 6.000, spare = 2896.000 load = 0.4709 max usecs: 18.000, spare = 2884.000 load = 0.5456 max usecs: 18.000, spare = 2884.000 load = 0.3934 max usecs: 7.000, spare = 2895.000 load = 0.4551 max usecs: 15.000, spare = 2887.000 load = 0.5377 max usecs: 18.000, spare = 2884.000 12:06:12.597 Could not connect to JACK server as client. Please check the messages window for more info. load = 0.3722 max usecs: 6.000, spare = 2896.000 [and so on] \\snip\\\\\\\\\\\\\\\\\\\\\ jackd is running: mechko@almeida:~$ ps aux | grep jackd mechko 25185 0.5 2.6 20552 20592 ? SLsl 12:07 0:00 jackd -v -R -t2000 -dalsa -dhw:0 -r44100 -p128 -n2 -S meterbridge fails: mechko@almeida:~$ meterbridge -t vu alsa_pcm:capture_1 alsa_pcm:capture_2 jack server not running? Could you please give me a hint what to do? Thanks in advance, Mirko Swillus

19 years

[linux-audio-user] Re: [PD] Sound splitter

by Chris McCormick

On Fri, Feb 09, 2007 at 08:35:34AM +0000, Lots of people wrote: > The attachment? Whoops! Attached. Chris. ------------------- chris(a)mccormick.cx http://mccormick.cx

19 years

[linux-audio-user] gstreamer, rhythmbox, and mp3

by Victor Roetman

I've been using rhythmbox for listening to my mp3 files on my Ubuntu laptop. Mostly this was because it supports daap sharing, and gives me a nice interface for choosing songs. It works well except I've been having sound quality problems. I seem to get some minor distortion, and some pops and cracks when playing a lot of my mp3 files. I took one especially problematic file and played it with alsaplayer and had no troubles (at a louder volume even). I'm about 98% sure that somehow gstreamer is peaking/clipping, even when the volume is way down. Banshee, I've noticed, does the same thing. mpg123 and mpg321 seem to have no problems. xine engine of amarok seems to work. Has anyone else noticed anything like this? Is this a problem with gstreamer in general? What's going on here? What's with the levels that it would clip when it's not even loud? (and I really wish there was a jack backend for gstreamer) Any comments, suggestions? Does anyone here follow gstreamer at all? -- -= =- vroetman(a)myrealbox.com -= =-

19 years

[linux-audio-user] Re: Spring reverb?

by philicorda

"I am working on a tune that needs surf guitar. The CAPS AMP IV does a great Fender Twin, and the #4 cabinet in CAPS Cabinet II seems to do pretty well as a 2x12 open-back Fender cab. But the classic surf guitar sound of course includes the high-pitched plinking of a Fender Twin reverb spring slapping against its metal case. If you haven't heard it, it's kind of like entering the V'Ger central core, but really high-pitched and fast, and in 16th notes while double-picking. Picking out a muted barre chord into a spring reverb turned up way too high, you can definitely *hear* the springs." Despite the way this should be a very non linear and random kind of reverb, convolution reverbs with a spring reverb sample do this surprisingly well. There are plenty of convolvers around, and loads of samples out there (noisevault is good). I made my own stereo impulses by recording both ends (both in and out as outs, impedances way off, but who cares) of a spring reverb tray and giving it a thwack with a drum stick. I use SIR running on ardour-vst on a computer dedicated to reverbs and other heavy DSP. Connect via netjack, chuck artificial latency on the sends so it comes back in time, and bob's your uncle. (may not be as easy as it sounds, contents may settle in transit). I've tried some other non vst native Linux convolvers, but have not found one as efficient yet. Recommendations welcomed.

19 years

[linux-audio-user] cart chunk writer?

by Dan Easley

are there any facilities on linux to create cart chunks within a wave file? <http://cartchunk.org:8080/> i'd love to be able to add cart chunk metadata to a wave file from a command line. it would save my butt at work. i haven't been able to find anything via google. -- daneasley(a)gmail.com dan(a)towndowner.com dan(a)burntpossum.com http://towndowner.com http://burntpossum.com

19 years

[linux-audio-user] [ANN] Releases : Tapeutape and Tranches

by Florent Berthaut

Hi everyone, I'm pleased to announce the release Tranches and Tapeutape. (my first release !!! ) Tapeutape is a midi-controlled virtual sampler.It is highly configurable,lets you create instruments/kits/setups and is designed for live performance.It works on Gnu/Linux with the jack sound server. There's a command-line and a gui version. It can be found at http://www.hitmuri.com/Tapeutape An example of what can be done with it : http://www.hitmuri.com/Tapeutape/demo_tapeutape.ogg Tranches is a midi-controlled multi-(inputs|outputs) live beat repeat|redirect tool for the jack sound server on Gnu/Linux !!! There's a command-line and a gui version. website : http://hitmuri.com/Tranches/ audio example : http://www.hitmuri.com/Tranches/demo_tranches.ogg Regards, Flo

19 years

[linux-audio-user] record from the monitor mixer on ice1712?

by Thomas Kuther

Hello LAUs, maybe someone here knows how I can record from an ICE1712's Monitor Mixer using ALSA? I routed the HW in 1/2 to the digital mixer in envy24control, and now I want to record that, but I'm perhaps just too stupid to find out how the monitor mixer is called in ALSA. That device is R/W for sure: http://www.alsa-project.org/alsa/ftp/datasheets/icensemble/envy24.pdf Using 4Front's OSS (not the in-kernel) it is as simple as recording from /dev/dsp4, which is the monitor on my audiophile 2496. But OSS is a pain when trying to use jack or any recording app like rezound or audacity with it. This somehow must be possible using ALSA too, not? I also tried using ALSA's jack plugin, and record with jack_capture, but a) I get total sound distortion as soon jack_capture starts, and b) that's one layer too much anyway, as the card can do "loopback" itself. So i'd prefer ALSA. Any help would be appreciated. Thanks, Tom

19 years

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Linux-audio-user February 2007