Linux-audio-user May 2010

linux-audio-user@lists.linuxaudio.org

153 participants
128 discussions

by Harry Van Haaren

Hey all, I used to produce some dubstep songs (totally synthesized) using various proprietary solutions, but I'm recently attracted to the bass frequencies again. Hence I'm looking for a good synth for bass, with plenty of assignable LFO's, Portamento, and... sounding good! I've a little experience with MX44, another little bit with AlsaMod Synth, a bit of a giggle with ZynSubAdd, a play around with the recent Yoshimi, but I'm not convinced of any really heavy bass yet. Chances are that I've not got the right settings for the synth, Just wondering if there's any dubstep guys out there using linux synths, or if anybody has any PD dubstep patches? Cheers, -Harry

15 years, 1 month

Is ALSA using hrtimer?

by Grant

I have hrtimer support compiled into my kernel, but when I watch 'top' while music is playing I can see that 'timer' uses CPU cycles but 'hrtimer' never does. How can I find out if ALSA is using hrtimer? - Grant

15 years, 1 month

a NEW question about pitch and speed

by Kevin Cosgrove

This morning when I turned on my computer it started playing stuff on Pandora, but more slowly and at a lower pitch than normal. I wondered if this was just a problem with Pandora, so I checked my own song files with Amarok. Again, that music plays more slowly and at a lower pitch than I'd expect. 1-2 days ago this very same computer played at the right pitch and speed. What could have gone wrong? The built-in (only) sound card's clock is now running more slowly? Some system wide configuration changed for the sound system, and it now slows things down? If it's a configuration thing, then I didn't intend to do that, nor did I even touch the configuration. This computer runs PulseAudio on Mandriva 2009.1. The driver is snd_intel8x0. Any ideas folks? Thanks.... -- Kevin

15 years, 1 month

Campaign: "The most remarkable album on this entire planet"

by Leonard Ritter

Hi everyone, perhaps you know me as "paniq". For 10 years, I'm releasing good and free music on the web. My wife and I, we are currently organizing an album project called "The most remarkable album on this entire planet". Most likely remarkable because it will sound awesome, maybe also because the album is entirely funded by fans, but most importantly, because the album will be completely made with free software, in this particular case: Ubuntu Linux and the Linux Audio infrastructure. We hope the album will always be remembered for showing that selling copies isn't the only way to produce great work, and that the power of free software can indeed outsmart commercial software. Furthermore, our aim is to contribute to the treasure of free works, because we believe that financial interests interfere with quality, truth and honesty in art, and we wish to set an example. Since our launch last week, we have raised 50% of the basic production cost from our closest friends and fans, but we are not quite there yet. To make donating more attractive, we have thrown in some rewards such as early downloads, a final CD, access to video logs and being added to the album's booklet. We hope you'll like our idea and will support us, financially or through telling your friends. Love, Sylvia & Leonard

15 years, 1 month

Q: HOWTO: extract the beat times from an audio track

by Jeremy Henty

I want to create an animation for a track my band recorded and I'm wondering how to get it in sync. It's a straight 4/4 piece but we weren't playing to a click track so the tempo varies slightly. Most of the advice I get is "Open it in Audacity (or similar) and eyeball the beats from the waveform" but it would be very cool if there was an app that could read the audio and spit out the beat times. Is there? Regards, Jeremy Henty -- The music business is a cruel and shallow money trench, a long plastic hallway where thieves and pimps run free, and good men die like dogs. There's also a negative side. -- Hunter S. Thompson

15 years, 1 month

qjackctl fails to exit gracefully on UbuntuStudio 10.4

by John Ouzts

I recently did a fresh UbuntuStudio 10.4 amd64 install. I found that I could only start qjackctl once without rebooting. When I attempted to close jackd with the Quit button on qjackctl, the next attempt to run it results in the following message: JACK compiled with System V SHM support. loading driver .. Enhanced3DNow! detected SSE2 detected apparent rate = 48000 creating alsa driver ... hw:0|hw:0|256|2|48000|0|0|nomon|swmeter|-|32bit control device hw:0 23:08:49.544 JACK was stopped successfully. 23:08:49.544 Post-shutdown script... 23:08:49.545 killall jackd 23:08:49.549 JACK has crashed. jackd: no process found 23:08:49.993 Post-shutdown script terminated with exit status=256. 23:08:50.025 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Neither "sudo kill -9 <PIDs of jackd, qjackctl and qjackctl.bin>" nor "sudo pkill -9 qjackctl" nor closing the terminal in which qjackctl was executed eliminates this problem. Since this was a clean install with very little added beyond ambdec, tetraproc and mhwaveedit, I don't have any explanation for this problem except bad karma between the current UbuntuStudio and qjackctl. On one occasion when starting the reboot, I got an error message asking if I wanted to wait for process Unknown to complete. The good side to this is that UbuntuStudio 10.4 boots up quickly. Does anyone have any ideas? Thanks, John

15 years, 1 month

Re: [LAU] qjackctl fails to exit gracefully on UbuntuStudio

by alexander

On 05/31/2010 07:57 PM, John Ouzts wrote: > alexander wrote: > | What output do you get when doing: lsof | grep snd > > alexander, > > Below is the data that you requested. My current thinking, following > Raine's lead, is that it is not qjackctl that is > not exiting gracefully but pulseaudio, which leaves too much memory > tied up in /dev/shm for qjackctl to come up > twice in a row on my machine (1GB of RAM). Once I run "sudo rm > /dev/shm/pulse*" qjackctl cycles at least 3 times > (all I've tried). The problem may still lie with qjackctl, since I am > using the "artsshell -q terminate" in qjackctl>Setup> > Options>Execute script on Startup. While that seems to stop > pulseaudio, it does not release the memory that > pulseaudio tied up in /dev/shm. > > If someone has superior insight, please let me know. > > John > Hmm, well, it doesn't seem to be the classic problem that some app is hugging your soundcard.. but I wonder, what are all those processes doing with libsndfile? then again I sit on archlinux, either way it shouldn't be a problem as far as I know.. There is also something called "pulseaudi" in there too, typo? Btw, Can't you just remove pulseaudio altogether? or is it trapped in some kind of nasty dependency hell?

15 years, 1 month

Re: [LAU] [alsa-devel] Crash 2.6.32.x, Echo 3G, Alsa, SCSI

by F. Heitkamp

On 02/02/10 17:09, Giuliano Pochini wrote: > On Tue, 02 Feb 2010 16:30:27 +0100 > Clemens Ladisch<clemens(a)ladisch.de> wrote: > > >> Frederick V Heitkamp wrote: >> >>> Repeatable Hard Crash. >>> What else do I need to provide? >>> >>> [ 5484.995249] WriteControlReg: not written, no change >>> [ 5485.069621] divide error: 0000 [#1] pcm_hw_params ok >>> [ 5485.070159] Prepare rate=44100 format=2 channels=2 >>> [ 5485.070161] set_audio_format[14] = 5 >>> [ 5485.070166] Prepare rate=44100 format=2 channels=2 >>> [ 5485.070167] set_audio_format[14] = 5 >>> [ 5485.070003] PREEMPT SMP >>> ... >>> [ 5485.070003] EIP is at pcm_pointer+0x37/0x70 [snd_echo3g] >>> ... >>> [ 5485.965788] [<c10041c0>] ? do_divide_error+0x0/0x90 >>> [ 5485.980619] [<f87aa037>] ? pcm_pointer+0x37/0x70 [snd_echo3g] >>> [ 5485.998047] [<c104764e>] ? run_timer_softirq+0x17e/0x2e0 >>> [ 5486.014175] [<f87ac9bf>] ? snd_echo_interrupt+0x11f/0x240 [snd_echo3g] >>> [ 5486.033940] [<c107a5d5>] ? handle_IRQ_event+0x45/0x190 >>> >> bytes_to_frames() divides by runtime->frame_bits which is not set until >> after the hw_params callback has succeeded, but the corresponding >> chip->substream[] entry is set in that callback, by init_engine(). It >> should probably have been set in the prepare callback. >> > I've just had another look at my code. Although it never happened to me, it > is indeed possible when hw_params() completes if another substream is > already running. The reason is that the card delivers an irq when it > executes an irq instruction in any of the running s-g lists. The irq > handler cannot know which substream caused it, so it has to call the > pointer() function for each of the configured substreams (ie. the ones > which completed one of the pcm_*_hw_params() callbacks. > > There is another possible fix. I tested it briefly. It looks ok wrt race > conditions because pipe->state is set only in the trigger callback. I hope > I didn't overlook anything again... > > > Signed-off-by: Giuliano Pochini<pochini(a)shiny.it> > > --- alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c__orig 2010-02-02 22:37:33.000000000 +0100 > +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-02-02 22:44:03.000000000 +0100 > @@ -1821,7 +1821,9 @@ static irqreturn_t snd_echo_interrupt(in > /* The hardware doesn't tell us which substream caused the irq, > thus we have to check all running substreams. */ > for (ss = 0; ss< DSP_MAXPIPES; ss++) { > - if ((substream = chip->substream[ss])) { > + substream = chip->substream[ss]; > + if (substream&& ((struct audiopipe *)substream->runtime-> > + private_data)->state == PIPE_STATE_STARTED) { > period = pcm_pointer(substream) / > substream->runtime->period_size; > if (period != chip->last_period[ss]) { > > > I've tried some newer kernels. Still having problems with the echo 3G. This is kernel version: 2.6.32.13. The above patches posted to the linux kernel list seemed to get rid of the crashes, but evidently did not go into the main kernel tree. Any help appreciated. I am willing to help to the best of my ability. Thanks! Fred This segment keeps looping to infinity: [ 9331.528043] pcm_hw_free(0) [ 9331.529200] free_pipes: Pipe 0 [ 9331.545230] pcm_hw_freed [ 9331.552808] pcm_hw_freed [ 9331.560365] pcm_close [ 9331.567145] pcm_close oc=0 cs=1 rs=1 [ 9331.578595] pcm_close2 oc=0 cs=1 rs=0 [ 9490.665630] pcm_analog_out_open [ 9490.675009] max_channels=6 [ 9490.683109] pcm_analog_out_open cs=1 oc=1 r=44100 [ 9490.698046] allocate_pipes: ch=0 int=2 [ 9490.702428] allocate_pipes: ok [ 9490.718350] allocate_pipes()=0 [ 9490.727466] pcm_hw_params (bufsize=131072B periods=2 persize=65536B) [ 9490.746450] SetSampleRate: 44100 clock d63 [ 9490.756870] WriteControlReg: Setting 0xd63, 0x3bfe [ 9490.768468] WriteControlReg: not written, no change [ 9490.787532] pcm_hw_params ok [ 9490.796155] Prepare rate=44100 format=2 channels=2 [ 9490.810464] set_audio_format[0] = 5 [ 9490.820922] Prepare rate=44100 format=2 channels=2 [ 9490.835233] set_audio_format[0] = 5 [ 9490.845758] pcm_trigger start [ 9490.847582] start_transport 1 [ 9497.317297] pcm_trigger stop [ 9497.318517] stop_transport 1 [ 9497.334516] pcm_hw_free(0) [ 9497.335483] free_pipes: Pipe 0 [ 9497.351709] pcm_hw_freed [ 9497.359278] pcm_hw_freed [ 9497.366836] pcm_close [ 9497.373613] pcm_close oc=0 cs=1 rs=1 [ 9497.385061] pcm_close2 oc=0 cs=1 rs=0 [ 9503.442232] pcm_analog_out_open [ 9503.451611] max_channels=6 [ 9503.459713] pcm_analog_out_open cs=1 oc=1 r=44100 [ 9503.474663] allocate_pipes: ch=0 int=2 [ 9503.484394] allocate_pipes: ok [ 9503.494962] allocate_pipes()=0 [ 9503.504081] pcm_hw_params (bufsize=131072B periods=2 persize=65536B) [ 9503.523069] SetSampleRate: 44100 clock d63 [ 9503.532174] WriteControlReg: Setting 0xd63, 0x3bfe [ 9503.545805] WriteControlReg: not written, no change [ 9503.564143] pcm_hw_params ok [ 9503.572765] Prepare rate=44100 format=2 channels=2 [ 9503.587074] set_audio_format[0] = 5 [ 9503.597490] Prepare rate=44100 format=2 channels=2 [ 9503.611839] set_audio_format[0] = 5 [ 9503.622355] pcm_trigger start [ 9503.623236] start_transport 1 [ 9510.095872] pcm_trigger stop [ 9510.096578] stop_transport 1 [ 9510.113193] pcm_hw_free(0) [ 9510.114163] free_pipes: Pipe 0 [ 9510.130405] pcm_hw_freed [ 9510.137982] pcm_hw_freed [ 9510.145541] pcm_close [ 9510.152317] pcm_close oc=0 cs=1 rs=1

15 years, 1 month

Complete MIDI confusion

by Julien Claassen

Hello everyone! Sorry to bother again, but I'm confused. I want to do a little skullduggery. So I started these: jackd - for basic audio/midi and transport a2jmidid -e - To expose my ALSA MIDI ports to JACK j2amidi_bridge - to expose my JACKMIDI stuff to ALSA jack_midi_clock - to generate a good clocksource, that's bound to the transport system of JACK Now I want to start a midi sequencer (alsamidi capable) and connect the jack_midi_clock to it. The midi sequencer somehow has it's ports hidden, but can connect to any input/output alsaseq port. Now how do I connect the jack_midi_clock to finally end up with it's signal exposed to ALSA, so I can connect it to my sequencer? Here's my list of jack_midi_ports: a2j:Virtual Raw MIDI 0-0 [16] (capture): VirMIDI 0-0 a2j:Virtual Raw MIDI 0-0 [16] (playback): VirMIDI 0-0 a2j:Virtual Raw MIDI 0-1 [17] (capture): VirMIDI 0-1 a2j:Virtual Raw MIDI 0-1 [17] (playback): VirMIDI 0-1 a2j:Virtual Raw MIDI 0-2 [18] (capture): VirMIDI 0-2 a2j:Virtual Raw MIDI 0-2 [18] (playback): VirMIDI 0-2 a2j:Virtual Raw MIDI 0-3 [19] (capture): VirMIDI 0-3 a2j:Virtual Raw MIDI 0-3 [19] (playback): VirMIDI 0-3 a2j:M Audio Delta 1010LT [20] (capture): M Audio Delta 1010LT MIDI a2j:M Audio Delta 1010LT [20] (playback): M Audio Delta 1010LT MIDI j2a_bridge:playback a2j:j2a_bridge [129] (capture): capture Jack MIDI Clock:midi_out Can any body please help me clear up the mess a bit? Kindly yours Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

15 years, 1 month

Jack support in Mumble

by Daniel Worth

Hello all, This is an appeal for help in solving the last great gap in the audio tool chain for podcasters. The Linux audio world now has a robust compliment of professional tools to record an independent radio show. We all know that the open source and free software community is under reported on in the media and the effort it takes to put out a show it greater than it should be. Recently on the show Shotofjaq they mentioned Mumble as a possible solution to voip woes that we currently have for podcasting with hosts in different locations. Currently the only solution we have been able to get to work with our Jack setup is an outdated version of Skype and compiling the alsa-utilities by hand to include jack support . That as well as some custom work in .asound to make something that is stable enough to get us through an hour long show without headaches and break downs. We have tried going the SIP route as well and the issues are thus: 1. Asterisk is a pain in the butt to set up. 2. It's difficult to set up your firewall on the client side to get reliable connections. 3. There are no sip clients that support jack either and no other hacky work arounds have gotten us a reliable setup anyways. We have filed a feature request for every SIP client we could think of and have been turned down by them all. Mumble is quick and easy to set up on server and client end. All it is lacking is Jack support. If it's a question of money or developers, then I would love help in identifying what hurdles need to be overcome. So please, if you have any incite into how we can get this accomplished let me know. I will not let this be the one thing preventing quality content about the floss world. The new feature request is here. https://sourceforge.net/tracker/index.php?func=detail&aid=3008867&group_id=… please show your support in any way you can. Daniel Worth Host Open Source Musician Podcast http://opensourcemusician.com

15 years, 1 month

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Linux-audio-user May 2010