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20661 discussions

[linux-audio-user] Trying to record audio from line in and additional questions

by Sylvain Robitaille

First the vitals: The computer system is a 400 MHz K6-2 with 512 MB memory, and a Hercules Fortissimo II (Cirrus Logic Soundfusion CS4624 processor) sound card. The OS is Linux-2.4.20 (built up from Slackware-8.0, with kernel patches and package upgrades). I have the following ALSA components installed: alsa-driver-0.9.0rc7, alsa-oss-0.9.0rc1, alsa-utils-0.9.0rc7, alsa-lib-0.9.0rc7, alsa-tools-0.9.0rc7. The kernel is mostly monolithic, with the following options (abridged to include only those I know are relevant; if folks want the complete listing, I can easily provide it): CONFIG_EXPERIMENTAL=y CONFIG_MODULES=y CONFIG_MODVERSIONS=y CONFIG_KMOD=y CONFIG_LOLAT=y CONFIG_LOLAT_SYSCTL=y CONFIG_RTC=y CONFIG_SOUND=y ALSA is the only component that I have modularized (only because I don't have the impression it can easily be just built-in, actually), and the following modules are loaded: Module Size Used by snd-mixer-oss 11120 1 (autoclean) snd-cs46xx 58096 1 (autoclean) snd-pcm 50752 0 (autoclean) [snd-cs46xx] snd-timer 9392 0 (autoclean) [snd-pcm] snd-ac97-codec 26912 0 (autoclean) [snd-cs46xx] snd-rawmidi 11808 0 (autoclean) [snd-cs46xx] snd-seq-device 3824 0 (autoclean) [snd-rawmidi] snd 27280 0 (autoclean) [snd-mixer-oss snd-cs46xx snd-pcm snd-timer snd-ac97-codec snd-rawmidi snd-seq-device] I've been using Linux with OSS and a Media-Vision Pro-Audio-Studio-16 soundcard for years, and have basically had very little trouble with that combination. Upgrading to Linux-2.4.x caused the sound output to be rather noisy, so I recently took advantage of the situation to buy a new soundcard, and upgrade to using ALSA (which I've been wanting to do so I can investigate certain rather exciting applications such as Ardour and the new Rosegarden). The applications I've been using for years with OSS, (notably Dap, Mixviews, Aumix, and a few others) all appear to mostly be working fine, presumably through the ALSA OSS emulation. Aumixer now shows some lables (PhoneIn and Video) which don't correspond to any inputs I have on my soundcard, while some others (IGain and Line1) don't appear to have any use, given the way I'm currently using the card (signal from a hardware mixer going to the Line input, and signal from the main, front line output going back to the mixer). I can play back sound files without any problem. In fact I've also tried a few new (to me) applications, such as Audacity, ProTux, and (though I'm having problems with it I won't cover here) Ardour, and they too play back sounds quite nicely (the sound is "grainy", though, perhaps I need to investigate the Low-Latency issues more, or increase the size of some buffer somewhere). I can send sound to the computer, and listen to that sound coming back from the computer (adjusting the level with the "line" level in Aumix, for example), but if I try to record that sound (in any of the applications I've tried, including all those listed above), with "record enable" selected in Aumix, I get no audio recorded (silent sound file). Also, I find it interesting to note that I can record-enable only one input source at a time (contrary to my old Media-Vision card). I'm assuming that's a function of the hardware, though and can't be fixed in software? Finally, we get to my questions: - Can someone point me in the right direction(s) to find a solution to this, so I can record audio (preferably from numerous sources) into my computer? I'll be perfectly happy with pointers to documentation (though I expect to get at least some to documentation I've already read, because I think I've repeatedly gone through all the documentation I already found myself), and suggestions to upgrade certain components. - I've tried to start Jackd at system boot, with the following command added to my rc.local file: /local/bin/jackd -d alsa -d cs46xx -p 512 & Jackd fails to start at that point (I'm afraid I haven't noted the exact error message, but I can make a point of doing so next time I reboot the system), but it does start when run manually from a root shell. Does anyone have any idea why jackd might not start at boot time, (perhaps the ALSA modules aren't loaded yet at that time)? If I understand my boot sequence properly, kernel modules are setup (via "depmod -a") before rc.local is run, so the modules may not be loaded, but they should by that point be loadable (they load automatically by the kernel in regular use). - Is it possible to build ALSA into the kernel? - on a slightly unrelated point, my new soundcard has a built-in synthesizer with at least some features I'd like to explore. Though I know this won't replace any of my hardware synths (or some of the software synths I've begun to play with), I'd like to play around with some of its built-in sounds, and probably use it (at least until I get something better) as my "preset playback" device, controlled from an external MIDI controller. Does anyone have suggestions for how I might go about that? (again, simple pointers to existing documentation would be more than appreciated). In case people are wondering about my specific intended application, I'd like to use a Roland Octapad to trigger percussion sounds from the soundcard's built-in synth. Thanks in advance for any pointers people can provide.... -- ---------------------------------------------------------------------- Sylvain Robitaille syl(a)alcor.concordia.ca Major in Electroacoustic Studies Concordia University Faculty of Fine Arts / Music Department Montreal, Quebec, Canada ----------------------------------------------------------------------

22 years

Re: [linux-audio-user] alsa setup debug (was: jackd -d alsa -d hw:0 -p512 / problem to start)

by Tim Orford

> > this sounds like what happens when alsa is not running properly. > > I do the following tests: arecord (meanwhile in other xterm play some > mp3), and then when I do aplay, file.tests contains a portion of this > mp3 recorded [I suppose] using alsa. > > But you're right, alsa say 'failed' when start and all seems to go bad, i'm assuming you mean the alsa init script. did you look in the logs (eg /var/log/syslog) for why it failed? maybe you could tell us what distro and software versions you are using, together with the output of: lspci -v lsmod service alsa status (or whatever it is on your distro) cat /var/log/syslog | grep -i alsa (for the last boot) also read the documentation for your card on http://www.alsa-project.org and the quicktoots: http://www.djcj.org/LAU/quicktoots/ the archives for the alsa-user list might also be instructive... cheers -- Tim Orford

22 years

[linux-audio-user] mounting network drives for dummies?

by Mark Knecht

Hi, I've got two Linux boxes, one Gentoo and the other PlanetCCRMA. I need to move about 10GB of data from one to the other. How can I do this? I guess that over Ethernet maybe Samba or NFS might work? I don't know anything about making either of these technologies work, and obviously I don't want to start building kernels or anything like that to get there. Has anyone got a tutorial on how to do this easily. I really don't want to become an IT guy to make this work. If it's too difficult, then I could dig up and add a 1394 adapter to one box and dump it to the other that way. The second machine has 1394 already. I just didn't want to open the box up and mess with cards. Thanks very super much in advance, Mark

22 years

[linux-audio-user] (no subject)

by Sascha Retzki

unsubscribe shmaster lantis(a)linux.net _____________________________________________________________ Linux.Net -->Open Source to everyone Powered by Linare Corporation http://www.linare.com/

22 years

[linux-audio-user] CheeseTracker 0.9.0 released!

by Juan Linietsky

Cheesetracker is a portable Impulse Tracker clone. It supports all the main Impulse Tracker and FastTracker/SoundTracker features, plus many more. It is licensed under the GNU Public License. It runs under Linux/BSDs, MacOSX(QT/Mac or Qt/X11), or Win32(Cygwin). It can be obtained at http://cheesetronic.sf.net For those unfamiliar with trackers, this is basically an all-in-one sequencer/sampler/sample editor/mixer/fx processor bundle, wich provides fast and flexible means for professional grade music composing. Including in the release are tutorials and documentation. Volunteers to help to better documment it would be highly appreciated. The ChangeLog for this release follows: v0.9.0 ------ -Removed sample mode (Scream Tracker 3 mode) as It's obsolete and not needed for backwards compatibility. -Instruments are now layered and can perform up to 4 simultaneous voices with individual parameters each. -Added an effect buffers system. Instruments are now routed to custom buffers (each with individual effect chains), which can also re-route to other buffers. This allows to create very complex effect routes for realtime processing. -Effect buffers are "process on demand", which means they are smart enough to notice when they are doing nothing, thus disabling themselves. -Added a few internal effects: Amplifier, Clipping/Distortion, Recursive Delay Line, Stereo Enhancer, Chorus and Reverb. -Added a LADSPA effect source plugin. LADSPA plugins can be added to the chains. -Created new file formats that save all the new features: .CT (CheeseTracker Module) .CI (Cheesetracker Instrument) and .CS (CheeseTracker Sample) -Added preview to the sample file selection box, just hilite a file and use your keyboard to play notes (/ and * work in there too). -Readded JACK Driver (Kasper Souren) -Added RTAUDIO driver, allows for porting to Win32/ASIO and OSX/CoreAudio -Fixed some big endian compatibility issues. CheeseTracker should work fine again on big endian machines. -MacOSX port and build system/build fixes courtesy of Benjamin Reed -Fixed tons and tons of bugs. AND NOW PLEASE READ: How fast CheeseTracker reaches version 1.0 depends on YOU. The focus of this version is STABILITY. Because of this, I need to receive as many bug reports as I can, both program and build system. If you find a bug, I'd be enormously grateful if you submit it. Even if it is an obvious bug to you, chances that other people will find and report the same bug are much smaller than what you may think. If you dont report a bug that annoys you, the chances of it reappearing in the next version will allways be higher. Planned for 1.0.0: -=-=-=-=-=-=-=-=- -Rock Solid stability -WAV exporting -A hopefully working Windows port. This depends mainly on the Qt-Win32 project. If you are a good Windows programmer and would like to see CheeseTracker working in there sooner, please give those guys a hand! Enjoy!! Juan Linietsky

22 years

[linux-audio-user] Writing Audio Applications with JACK

by Daniel James

http://dis-dot-dat.net/jacktuts/starting/ Cheers Daniel

22 years

[linux-audio-user] qjackctl compilation errors

by Vescovi Christophe

Hi, I just try to complile qjackctl 0.1.0 and I get some compilation errors : In file included from src/qjackctlMainForm.cpp:26: src/qjackctlMainForm.ui.h: In member function `void qjackctlMainForm::transportStart()': src/qjackctlMainForm.ui.h:1281: `jack_transport_start' undeclared (first use this function) src/qjackctlMainForm.ui.h:1281: (Each undeclared identifier is reported only once for each function it appears in.) In file included from src/qjackctlMainForm.cpp:26: src/qjackctlMainForm.ui.h: In member function `void qjackctlMainForm::transportStop()': src/qjackctlMainForm.ui.h:1293: `jack_transport_stop' undeclared (first use this function) In file included from src/qjackctlMainForm.cpp:26: src/qjackctlMainForm.ui.h: In member function `void qjackctlMainForm::refreshStatus()': src/qjackctlMainForm.ui.h:1316: `jack_is_realtime' undeclared (first use this function) src/qjackctlMainForm.ui.h:1323: `jack_position_t' undeclared (first use this function) src/qjackctlMainForm.ui.h:1323: parse error before `;' token src/qjackctlMainForm.ui.h:1324: `tpos' undeclared (first use this function) src/qjackctlMainForm.ui.h:1324: `jack_transport_query' undeclared (first use this function) src/qjackctlMainForm.ui.h:1336: `JackTransportStarting' undeclared (first use this function) src/qjackctlMainForm.ui.h:1358: `JackPositionBBT' undeclared (first use this function) make[1]: *** [qjackctlMainForm.o] Erreur 1 make[1]: Leaving directory `/usr/src/qjackctl-0.1.0' make: *** [qjackctl] Erreur 2 All those undeclared functions are declared in <jack/transport.h> which exist on my machine in /usr/include, and are include in src/qjackctlMainForm.h .... well in fact <jack/jack.h> is include in qjackctlMainForm.h and <jack/transport.h> is include in jack.h. Does someone experiment these errors before ? I'm runing Mandrake 9.1, qjackctl 0.0.9a from thac's rpms works fine. Regards, Christophe

22 years

[linux-audio-user] Ardour mixdown questions/issues

by Greg Reddin

I have a small project that I'm recording for a friend and this is my first use of hopefully several projects using ardour as my DAW. (BTW, these are not paying gigs, just low-risk projects.) I'm having some mixdown problems I need help with. I haven't been very good about keeping up with releases. For this project I'm on 0.9beta3, but I will upgrade to 0.9beta6 before this weekend for the next project. Anyway, I can't figure out how to mixdown tracks with effects. I'm using a single-band PEQ and plate reverb in the pre-fader column for each track and using the outputs of each track as the inputs of the stereo track that eventually goes to disk, but for some reason the stereo track is recorded dry. If I solo it, the mix is dry. If I listen only to the mix of the individual tracks, it's wet. So here's my questions: 1. Is it possible (and how) to bounce a wet mix to a stereo track that will be exported to a .wav file? 2. Is there currently any way, or are there plans to add support for exporting an entire mix to disk instead of bouncing to a stereo track to be exported? (Of course you can do that now, but it's a dry mix.) I guess I'm asking for a non-realtime mixdown feature. 3. Somewhat related, is it possible to delete or undo automations? Thanks, Greg __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

22 years

RE: [linux-audio-user] ... like an old vinyl

by Reuben Martin

Granted, that is harsh. But then again you have to consider that the person listening to it may not know that you are trying to make it sound like old vinyl. Unless you make it particularlly obvious that you actually want it to sound like old vinyl the listener may just think you did a crappy recording job. Personally I've used this meathond and the less of a rolloff on either end and the less mix from the fuzz from the record the newer it will sound and vice versa. It's ultimately up to you though how you want it to sound. regards -Reuben -----Original Message----- From: Florin Andrei [mailto:florin@andrei.myip.org] Sent: Fri 10/24/2003 1:22 PM To: linux-audio-user(a)music.columbia.edu Cc: Subject: RE: [linux-audio-user] ... like an old vinyl On Wed, 2003-10-22 at 20:14, Reuben Martin wrote: > First: Using an equalizer, kill all frequencies below ~600 and above ~2k. Use a graphic EQ for this. (The two values are just examples, you can play with both frequencies to get the desired sound you're looking for.) That's pretty harsh. Recent vinyl records actually have a pretty good frequency response. > Second: get a recording of the inside blank track that comes after the last song of a noisy record. Loop this noise and mix it over top of the song once you've done the band pass filter from step one. It's actually the dust motes that contribute the most to the "vinyl feel". It should be fun to simulate that, actually. Should be like... lemme see... white noise shot through narrow and tall envelopes. Pretty much like the attack phase of the piano sound, except that the parameters are different (hence it sounds more like "plastic" than "metal"). -- Florin Andrei http://florin.myip.org/

22 years

RE: [linux-audio-user] Ardour mixdown questions/issues

by Reuben Martin

> Anyway, I can't figure out how to mixdown tracks with effects. I'm > using a single-band PEQ and plate reverb in the pre-fader column for > each track and using the outputs of each track as the inputs of the > stereo track that eventually goes to disk, but for some reason the > stereo track is recorded dry. If I solo it, the mix is dry. If I > listen only to the mix of the individual tracks, it's wet. You might try sending channel and FX outputs to a bus and then using that bus as the input for your stereo track. (Just a shot in the dark) regards -Reuben

22 years

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