Hi,
Couple of days ago I saw a thread about learning drums rhythms. Maybe my
website can help : www.percunivers.com (click on the uk flag for the
english version). Not only about drums, because everything is made under
64Studio (mainly ardour and lilypond). The weak point is about mp3 for
the exercices. I have no choice because this is the only format I can
use for streaming (via flash, argh!) and the only way to have my mp3
played on any Ipod-like from my students. I'm open for any open
solutions...
Maybe this can help, hope you send me any feedback.
Thanks,
PL
I've been trying to do some sound morphs between two different hits of a snare drum, one rimshot and the other a plain hit.
I tried using SMSTools, and even some windows programs under wine, but the results always seem to sound more like hitting a bucket full of water than a combination of the two drums.
I tried csdrummer, but only get a horrible scratchy noise from it.
I looked at Loris...the python-based example that comes with it seems horrifically complicated.
Maybe someone with a knowledge of Csound/Loris could be kind enough to help with a simple script? Or a step-by-step how to achieve this with Cecilia?
Are there any other programs I could try?
Hi,
I am stuck in a problem with MIDI merging.
I am currently using two external MIDI devices that connect to my notebook via
an external USB MIDI adaptor (a nice Edirol UM2).
One of these devices transmits data in running status, which means it omits
the status byte of NOTE ON messages if many NOTE ON messages are transmitted
and there is no change of the channel.
The second device transmits data in the ordinary uncompressed mode and never
leaves out the status byte.
Now I need to route these external MIDI devices to a MIDI application.
The devices show up in my notebook as physical ports UM2-1 and UM2-2 and what
I do is to simply connect these ports to the IN port of the MIDI application.
And now there is a problem: as I have connected the two devices to the IN port
the application (or the driver of the MIDI port) has to do MIDI merging. But
something is wrong with the merging, if the running status device's data
bytes have to be merged with the second device's data stream.
The application detects that there is the status byte missing in the complete
data stream that it is receiving and it does not "resync" even if the running
state device transmits a status byte because the channel has changed. So the
application indicates an error message stating there is a missing status byte
and that it only got data bytes.
1.) Which part is actually responsible for the MIDI merging of two or more
output devices into one input device? Is it the driver of the port or is it
the application, that is in the end receiving the MIDI data stream?
2.) Is it illegal to connect two output devices to one input port?
3.) If I should need some kind of virtual MIDI multiplexing/demultiplexing
driver: which one should I chose? I have debian and would like to install
from repository.
Thanks for any hint and kind regards,
Crypto.
hi...
i wrote an audio output driver for flashplayer, based on lennart
poetterings pulse driver.
git clone git://repo.or.cz/libflashsupport-jack.git
--
torben Hohn
Hi,
I understand that the module snd-virmidi can be loaded via modprobe from the
command line. BUT: I cannot get it to work. The module does not pass any data
transferred to it.
What am i doing wrong? I simply connected one MIDI output of things like the
vkeybd to the virtual MIDI port and then connected kmidimon to the virtual
port to see if something is happening. But nothing.
Any helpful hints on this?
kernel is 2.6.29-0.slh.1-sidux-686.
MIDI is working fine otherwise.
Regards,
Crypto.
Hi everyone.
I was kind of curious about the real latency of my system and did a
series of tests using jdelay (http://www.kokkinizita.net/linuxaudio/).
I share my results so others can see what to expect from such a usb
audio device and also because..
! Important !
I ran in a number of oddities that I can't explain myself. This could
mean that my results here are totally useless, so please don't take
them for real until someone can hopefully clarify those matters.
General Information:
-------------------
Audio Interface: Edirol UA-25
http://www.rolandus.com/products/productdetails.aspx?ObjectId=704http://alsa.opensrc.org/index.php/Edirol_UA-25
It can work in 44.1, 48kHz duplex and 96kHz either record or playback,
so I couldn't (and wouldn't have anyway) tested 44.1 and 48kHz.
There are Periods/Buffer from 2 to 42 possible, most common is I
believe 2 and 3 Periods/Buffer, I only tested these.
My pc is rather old, mainboard is a via km400 or km400a if that helps
anything.
On the software side there is Arch Linux with it's latest stock kernel,
2.6.28.8, jack 0.116.2 and qjackctl 0.3.4 (the patchbay made this a lot
easier).
Measurement Results:
-------------------
SR 44100
-----
Frames Periods F*P/SR Measured
64 2 2.9 8.2
128 2 5.8 11.1
256 2 11.6 15.9
512 2 23.2 28.5
1024 2 46.4 52.7
2048 2 92.9 102.1
4096 2 185.8 187.1
64 3 4.4 9.7
128 3 8.7 14.0
256 3 17.4 22.7
512 3 34.8 30.1
1024 3 69.7 65.0
2048 3 139.3 138.6
4096 3 278.6 279.9
SR 48000
-----
Frames Periods F*P/SR Measured
64 2 2.7 7.9
128 2 5.3 10.5
256 2 10.7 15.9
512 2 21.3 26.5
1024 2 42.7 48.9
2048 2 85.3 94.5
4096 2 170.7 171.9
64 3 4 9.2
128 3 8 13.2
256 3 16 21.2
512 3 32 27.2
1024 3 64 60.2
2048 3 128 127.2
4096 3 256 257.2
Questions to the wise guys:
--------------------------
1)
As probably everyone has noticed, with 3 periods/buffer and
frames/period of 512, 1024 and 2048 the measured latency is smaller
than the calculated latency. I don't see how this can be possible
since, as far as I understand it, the measured latency should be the
calculated + some.
2)
When doing those measurements I noticed that with 44100 the Frames
reported by jdelay always ended in .732 Frames and with 48000 always
ended in .761 Frames.
I found out that I can change that value by changing the sensitivity of
the input or the amplification of the output. In the case the variance
is below one frame, but I find it very strange that it's always the
same at a given amplification/sensitivity.
3)
Is it normal that the difference between calculated and measured
latency varies so much? Shouldn't it just always be the same?
4)
Should these results be similar with any usb interface?
What does have most influence on such results?
I hope this can be of use after the remaining questions have been
answered.
Best regards,
Philipp
Hi
It seems at least part of my problem with my edirol fa-66 firewire card
is due to the fact that it shares interrupt with the graphic card :-(
atte@vestbjerg:~$ lspci -v
<snip>
00:02.0 VGA compatible controller: Intel Corporation Mobile GM965/GL960
Integrated Graphics Controll
er (rev 03) (prog-if 00 [VGA controller])
Subsystem: Toshiba America Info Systems Unknown device ff50
Flags: bus master, fast devsel, latency 0, IRQ 17
Memory at f0000000 (64-bit, non-prefetchable) [size=1M]
Memory at d0000000 (64-bit, prefetchable) [size=256M]
I/O ports at 1800 [size=8]
Capabilities: <access denied>
<snip>
00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI
Controller #4 (rev 03) (prog-if 00 [UHCI])
Subsystem: Toshiba America Info Systems Unknown device ff50
Flags: bus master, medium devsel, latency 0, IRQ 17
I/O ports at 1820 [size=32]
<snip>
0a:01.0 FireWire (IEEE 1394): Ricoh Co Ltd R5C832 IEEE 1394 Controller
(rev 05) (prog-if 10 [OHCI])
Subsystem: Toshiba America Info Systems Unknown device ff50
Flags: bus master, medium devsel, latency 32, IRQ 17
Memory at f0400000 (32-bit, non-prefetchable) [size=2K]
Capabilities: <access denied>
<snip>
I looked in the bios, but it seems interrupts can't be moved around
(it's a toshiba laptop). Is there anyway to move the interrups by
software? Anything else that could be done? Or should I simply admit I
wasted my money and use the fa-66 as paper weight?
--
Atte
http://atte.dkhttp://modlys.dk
I have this USB audio interface working with my Ubuntu 9.04 system,
but I get very frequent clicks and pops, whether I run it via jack or
directly through ALSA. I don't get these pops with either a Native
Instruments Audio Kontrol 1 or a Novation Xiosynth.
Could this be related to the playback sync issue mentioned here?
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h…
Does anybody have this interface working without these glitches?
--
miles
http://kuniklo.bandcamp.com
Hi,
Music is digitalized now and so I use more and more my pc for listening
to music. I guess this are just modern times....
But this are also the times of a planet what get destroyed and less
important money just disappears...
So I want to save as much energy as possible without buying another pc.
Is there a way to handle this? What are the tricks or are there 'green'
distro's?
Thanks in advance,
\r
