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
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
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
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
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
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
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?
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
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
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
