http://blog.dis-dot-dat.net/2005/11/more-music.html
And now I have to get some work done.
James
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)
Hi,
this is a new release of jack_convolve/libconvolve, the simple
commandline based, convolution engine for JACK.
New features are
- LASH support:
have jack_convolve be restored with the exact same commandline
parameters when you restore your LASH session. Comment out two lines in
the makefile if you don't have LASH installed.
- Pseudo multichannel support (--pseudo-multi):
This creates an input port for each responsefile channel, instead of
only one input per response. While technically i.e. a stereo response
file is a response of a mono impulse recorded with stereo equipment it
many cases it might still sound good to convolve i.e. a stereo sum's
left and right channel with the left and right channel of a response
file individually :) And what sounds good is The Right Thing To Do.
- Output mixing now optional (--mix-outputs):
previously all outputs from the same channel number of different
responses (if multiple responses were loaded) have been mixed together
to save on some IFFT's. This is now optional. You can load several
responses with varying channel counts, if your heart desires. Note that
in mix-outputs mode all response files still need to have the same
channel count.
libconvolve had to be changed somewhat for these features, thus the
upgrade. Note that dssi_convolve will not link against this libconvolve.
I consider dssi_convolve broken atm anyways (still missing gui, etc).
Will take another while to get that fixed up.
Grab them at:
http://www.affenbande.org/~tapas/jack_convolve/
Bug reports welcome, especially with odd combinations of response files
with different channel counts :)
Flo
--
Palimm Palimm!
http://tapas.affenbande.org
Hi everybody,
I'm a new one on this list, excuse-me for my english it's not my natural
languange (I've got many red line under this word)
This message is a test and a hello.
see you.
P'tit Louis
tim hall:
> On Friday 25 November 2005 19:26, Kjetil S. Matheussen was like:
> > In addition, as you say, timemachine use this (and I
> > really mean it) stupid w64 fileformat by default, and I very seldom
> > remember to use the -f flag to override that. Instead of doing all that,
> > now I can just write "jack_capture", and it starts recording immmediately.
>
> I use a custom menu entry.
Listen, jack_capture is not a competitor for timemachine. Its a
program that fills a hole that timemachine previously filled the
best. Timemachine is, as Steve Harris says, a simulation of:
"a minidisc recorder in my studio running in a mode where when you
pressed record it wrote the last 10 seconds of audio to the disk and then
caught up to realtime and kept recording."
--
james:
> On Fri, 25 Nov, 2005 at 12:34AM -0800, Kjetil S. Matheussen spake thus:
> >
> > http://www.notam02.no/arkiv/src/
> >
>
> I normally use timemachine for recording like this - how does this
> differ?
timemachine is a different kind of program. Its a GUI thing that is ment
to be running all the time.
This is just a quick tool to get the sound going into your loudspeakers as
quickly as possible into disk. It happens quite a lot when I play with PD
or jamin, for example. What happened previously, when I had something
cool going, was that I had to start timemachine (which name does not
start with jack, so I allways use a lot of time remembering its name),
connect the correct jack ports into it in qjackctl, and press the
recording button. In addition, as you say, timemachine use this (and I
really mean it) stupid w64 fileformat by default, and I very seldom
remember to use the -f flag to override that. Instead of doing all that,
now I can just write "jack_capture", and it starts recording immmediately.
--
Hello, all,
Does anyone know if there's a way to convert text files to MP3s
in Linux? I'd like to be able to take some of my text files and make
audio out of them so I can listen to them while walking.
Thanks,
Terrence
Hi Paul and All,
> if its like aplay, use the -v argument to get a verbose listing of the
> h/w config that the device ends up with.
Unfortunately this device is Big Endian so I need to use plughw also
using a play:
$ aplay -D plughw:2 /usr/share/sounds/shutdown1.wav -v
Playing WAVE '/usr/share/sounds/shutdown1.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Stereo
Plug PCM: Linear conversion PCM (S16_BE)
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 16384
period_size : 4096
period_time : 92879
tick_time : 1000
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Hardware PCM card 2 'Audiophile USB (tm)' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_BE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 16384
period_size : 4096
period_time : 92879
tick_time : 1000
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
> then try to replicate that using the hw:N device with JACK (not the
> plughw device). JACK can use plughw devices, but it is intended to
> work better with a hw device.
>> Problems rise when I try using Jack.
>> even if I managed to get it starting with jackd -dalsa -dplughw:2,0
>> -r44100 -p1024 -n4 -Pplughw:2,0 -S -o2
> this is a very very odd way to start JACK. the -Pplughw:2,0 seems
> redundant.
You're right, I'll try to start with a better setup
using the parameters above, first block, and using plughw because BE,
jackd -v -dalsa -dplughw:2 -r44100 -p4096 -n4 -S
the problem is that I get the card playing few milliseconds of audio
continuously even without using any player connected to Jack. A sort of
tic -tac - tic -tac ....
if I reduce p to 1024 then it seems to work fine but the quality is
still lousy.
See below part of the output, any suggestion is most welcome.
$ jackd -v -dalsa -dplughw:2 -r44100 -p4096 -n4 -S
getting driver descriptor from /usr/lib/jack/jack_dummy.so
getting driver descriptor from /usr/lib/jack/jack_alsa.so
getting driver descriptor from /usr/lib/jack/jack_oss.so
jackd 0.100.1
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 ... plughw:2|plughw:2|4096|4|44100|0|0|nomon|
swmeter|-|16bit
control device hw:2
configuring for 44100Hz, period = 4096 frames, buffer = 4 periods
You appear to be using the ALSA software "plug" layer, probably
a result of using the "default" ALSA device. This is less
efficient than it could be. Consider using a hardware device
instead rather than using the plug layer. Usually the name of the
hardware device that corresponds to the first soun
nperiods = 4 for capture
registered builtin port type 32 bit float mono audio
new client: alsa_pcm, id = 1 type 1 @ 0x8746a78 fd = -1
You appear to be using the ALSA software "plug" layer, probably
a result of using the "default" ALSA device. This is less
efficient than it could be. Consider using a hardware device
instead rather than using the plug layer. Usually the name of the
hardware device that corresponds to the first soun
nperiods = 4 for playback
new buffer size 4096
registered port alsa_pcm:capture_1, offset = 16384
registered port alsa_pcm:capture_2, offset = 32768
registered port alsa_pcm:playback_1, offset = 0
registered port alsa_pcm:playback_2, offset = 0
++ jack_rechain_graph():
client alsa_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
4167 waiting for signals
late driver wakeup: nframes to process = 8192.
load = 0.2853 max usecs: 530.000, spare = 92349.000
.....
Thank You in advance
Ronny
Greetings!
My name is Sean Edwards. I have used Linux on my
desktop at home since 1997. My computer career
started in 1993, and I have done System Admin work
since earning a Novell Certification in 1995. In
2002, my career became Linux focused on a full-time
basis. I have done both contract and full-time work
in Chicago, Dublin Ireland, and now Omaha.
I learned to play guitar 30 years ago, and took a
break from it for 10 years when I started my family.
Before the kids came along, I used to do sequencing on
a 486 DOS computer, connected to external MIDI
equipment. For years, the only Linux audio I used was
for ripping and creating CD's. Last year, I
discovered the Planet CCRMA project, and was a lurker
on that list for several months. This Autumn, I
decided to get back into computer music and I find
what is available for Linux to be overwhelming!
Here is my equipment list:
Yamaha TX81Z (2)
Alesis MIDI Data Disk
Alesis HR 16 Drum Machine
Yamaha MJC8 MIDI Junction Controller
Alesis 1622 Mixer
A.R.T. MultiVerb
Behringer V-Amp Pro
Behringer UltraTwin
The computer is an AMD K7 500Mhz, 768MB RAM, VooDoo 3D
AGP Video Card, and a 20GB hard drive. My previous
employer provided me with a PowerBook G4 with 1GB RAM,
and Garage Band could not play more than 3 synthesized
voices simultaneously. My 6 year old desktop does
more music with Linux than Garage Band on a newer
PowerBook!
The software I mostly use:
Qjackctl
Rosegarden4
Qsynth
Hydrogen
Ecasound
-=Sean Edwards=-
__________________________________
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
Hi,
I've just bought myself (€130, in Italy) a Zoom 2G.u1 guitar effect
pedal (http://www.zoom.co.jp/english/products/g21u/index.php).
It is a very good sounding effect if you're on a budget, like me, and
with my strat plugged in it sounds like this:
http://www.cesaremarilungo.com/download/music/Hope_IN_PROGRESS_20051125.mp3
What makes it interesting is that it has a usb interface so it works
like an audio interface. You can listen to the mix from the DAW from its
headphone jack along with the direct output of the effected guitar (no
additional latency). The sampling is done at 24bit/96kHz.
I bought it to complete my transition to linux and I would really like
to use it with jackd. In windows and macos x it works without any driver
(it is seen as a generic usb audio device), and in fact through USB it
is limited to 16bit/44.1kHz.
The question is: how can I get it to work with alsa, and therefore jackd?
I compiled my kernel (2.6.13) without the usb-audio support, as I read
somewhere that it can interfere with alsa usb implementation. And I
guess that recompiling the kernel with it wouldn't allow me to use it
with jackd anyway.
Thanks in advance.
c.