I posted this to alsa-devel but since my previous post on this list
generated a lot of interest, I am just reposting it here.
As promised, here's an updated patch to add real multichannel playback
support (and improved multichannel capture) to the emu10k1 driver.
http://www.alsa-project.org/~rlrevell/emu10k1-multichannel-v001.patch
Please test it and report any problems. I am especially interested in
any regressions that impact regular PCM playback (the hw:0,0 device).
QuickStart:
$ jackd -R -v -d alsa -P hw:0,3 -C hw:0,2 -S
I tested this and it works well with 16in/16out at 128, 256, 512 frames.
32 and 64 should work too but I can't test as I'm running a stock 2.6.10
kernel for now ;-). You can check that the routing is correct by
connecting a JACK client to the playback ports corresponding to the FX
buses described in Documentation/Audigy-mixer.txt and
Documentation/SB-Live-mixer.txt, and verifying that the output appears
on that channel (the FX buses are numbered from 0 but JACK numbers
clients from 1). For example (from SB-Live-mixer.txt):
name='Music Playback Volume',index=0
This control is used to attenuate samples for left and right MIDI FX-bus
accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
So "alsaplayer -o jack -d alsa_pcm:playback_5,alsa_pcm:playback_6"
should output to FX buses 4 and 5, which you can test by lowering the
'Music' control in alsamixer. With an SBLive, use ports 1 and 2 for the
front channels, 3 and 4 for the rear channels. The Audigy uses
different channels, see the above docs for more info.
In addition to multichannel recording applications, this should also be
useful for OpenAL implementations, which are currently restricted to
using 21 sources due to the use of an extra voice per stereo PCM. This
should allow up to 63 sources.
This also adds some new register info including a per channel half loop
interrupt that I have discovered by reverse engineering the Windows
drivers.
Improvements over previous versions:
- Routes the 16 channels to the 16 FX buses by default.
- Enables the first 16 FX capture outputs by default, required for
full duplex operation at latencies lower than 512 frames.
- Rewrote the voice allocator to use a more efficient round
robin algorithm, eliminating the need to reserve the
first 16 voices for the multichannel device. The next free voice
is maintained in the card record and the search starts from there.
- Use an extra voice for playback timing rather than the EFX capture
interrupt. I was only ever able to get that to work at 64 frames. Also
there are definite advantages to being able to use the capture and
playback devices independently.
- Use the newly discovered per-channel half loop interrupt source for
the extra voice rather than the channel loop interrupts. For unknown
reasons, this works better for multichannel playback, and does not seem
to affect regular PCM playback at all.
TODO:
- Fix the send routing and volume controls for the multichannel device.
The current (copy and paste) solution assumes either one or two voices
per PCM. So the default settings work fine but changing them with the
mixer is likely to have unpredictable effects.
- EFX capture should capture output channels 16-32 (mostly unused now)
by default, so that we only capture the sources the user has connected
to the multichannel recording inputs in the DSP manager. Typically FX
buses 0-15 would be connected directly to FX outputs 16-32 so the
capture channels would correspond directly to the playback channels. In
order for this to work the default DSP configuration has to be changed
slightly.
Lee
So.. exams gone OK this semester, so i'm returning to linux-audio world :)
unfortunately my tascam us122 still dont work, and the new jack backend
doesn't even compile to me.. dont know what to do.. but, since someone
is using this thing right now on linux, i think there's hope even for me :)
willy@Zeryn:~$ uname -a
Linux Zeryn 2.6.10-mm3 #1 Wed Jan 12 21:01:17 CET 2005 ppc GNU/Linux
willy@Zeryn:~$ cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.8rc2 (Wed Jan 05
06:44:40 2005 UTC).
willy@Zeryn:~$ jackd -d alsa -d hw:0
jackd 0.99.0
Copyright 2001-2003 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
loading driver ..
creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
control device hw:0
configuring for 48000Hz, period = 1024 frames, buffer = 2 periods
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
Sorry. The audio interface "hw:0" doesn't support any of the hardware
sample formats that JACK's alsa-driver can use.
ALSA: cannot configure capture channel
cannot load driver module alsa
willy@Zeryn:~$ cd Samples/HistoricalBeats
willy@Zeryn:~/Samples/HistoricalBeats$ aplay -v -D hw:0 amen.wav
Playing WAVE 'amen.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Hardware PCM card 0 'TASCAM US-X2Y' device 0 subdevice 0
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
willy@Zeryn:~/Samples/HistoricalBeats$
willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ ./configure
[...]
willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ make
[...]
make[3]: Entering directory
`/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y'
if /bin/sh ../../libtool --mode=compile --tag=CC gcc -DHAVE_CONFIG_H -I.
-I. -I../.. -I../../config -I../.. -I../.. -D_REENTRANT
-D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config -I../.. -I../..
-D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT usx2y_driver.lo
-MD -MP -MF ".deps/usx2y_driver.Tpo" -c -o usx2y_driver.lo usx2y_driver.c; \
then mv -f ".deps/usx2y_driver.Tpo" ".deps/usx2y_driver.Plo"; else rm -f
".deps/usx2y_driver.Tpo"; exit 1; fi
gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../config -I../.. -I../..
-D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config
-I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT
usx2y_driver.lo -MD -MP -MF .deps/usx2y_driver.Tpo -c usx2y_driver.c
-fPIC -DPIC -o .libs/usx2y_driver.o
usx2y_driver.c: In function `alsa_driver_setup_io_function_pointers':
usx2y_driver.c:292: error: `sample_moveswap_dS_s16' undeclared (first
use in this function)
usx2y_driver.c:292: error: (Each undeclared identifier is reported only once
usx2y_driver.c:292: error: for each function it appears in.)
usx2y_driver.c:300: error: `sample_moveswap_dS_s24' undeclared (first
use in this function)
usx2y_driver.c: In function `alsa_driver_configure_stream':
usx2y_driver.c:334: error: `SND_PCM_FORMAT_S16BE' undeclared (first use
in this function)
usx2y_driver.c:334: error: initializer element is not constant
usx2y_driver.c:334: error: (near initialization for `formats[3].format')
usx2y_driver.c:334: error: initializer element is not constant
usx2y_driver.c:334: error: (near initialization for `formats[3]')
usx2y_driver.c:335: error: `SND_PCM_FORMAT_S16LE' undeclared (first use
in this function)
usx2y_driver.c:335: error: initializer element is not constant
usx2y_driver.c:335: error: (near initialization for `formats[4].format')
usx2y_driver.c:335: error: initializer element is not constant
usx2y_driver.c:335: error: (near initialization for `formats[4]')
make[3]: *** [usx2y_driver.lo] Error 1
make[3]: Leaving directory
`/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y'
[...]
willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$
if someone has some clue... :)
thanx for the time spent reading this
wil
---
Il jazz e' lasciare che la luce brilli, lasciarla essere.
-- Keith Jarrett
Hi!
I did try the following things to get current cvs to work.
I used the appropriate configuration method and then edited the
Synthesizer.cpp and exchanged "ASM_X86_MMX_SSE" with "CPP" as in the other
synthesizer_mode functions. The sampler compiled cleanly. But when running it
(with OR WITHOUT the --no-tune option) it crashes after a few seconds. It
says:
Zombified - calling shutdown handler.
Then it keeps idling there. I have to use killall -9 linuxsampler to
terminate it. CTRL-C doesn't work.
Any idea. I use jack 0.99.47 (CVS from sunday.
Is there anything I can do to help find this "bug?"? Some debug information,
etc...
Kindest regards
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
Has anyone developed any tools to capture voice conversations via a
voice capable modem on linux?
I am interested in using a speakerphone and phone line to webcast a
meeting where I don't have a decent Internet connection.
Steven Clift
Steven Clift - http://publicus.net - Reply to: clift(a)publicus.net
Join DoWire: http://dowire.org
E-Democracy: http://e-democracy.org
Liblo, the Lite OSC library, is an implementation of the Open Sound
Control [1] protocol for POSIX systems. It is written in ANSI C and
released under the GNU General Public Licence. It is designed to make
developing OSC applictions as easy as possible.
http://plugin.org.uk/liblo/
Changes:
Patch from Dave Robillard that fixes bundles of more than 4 messages.
Some const char * for C++ compatibility.
Added a source field to messages that represents the source from
which a message was received. Useful in method handlers to
determine which client sent the message.
Added patch from Walco van Loon and Pix that fixes a bug in the
hostname detection fallback code
- Steve
[1] http://www.cnmat.berkeley.edu/OpenSoundControl/
Hi everyone. It's an honor to be read by you. I'm
not a professional programmer, but I wrote some
text-based C++ programs that quiz me on musical stuff,
and was so pleased that I want to do something similar
with musical tones.
All I'll ever need is a library that will let me
playback a sample at a certain set of frequencies --
like 6 at a time (bass tone, high tone, and a
four-note chord) -- until keyboard input signals it to
stop. In fact, even that's more than I need -- if
playing a sample is hard, I'd be perfectly happy
listening to square waves.
I've looked at some audio libraries (OpenAL,
Penguinsound, Sound Object Library) but they're all
very intense, designed to do far more than I could
even understand, let alone want.
Is there a non-threatening, perhaps even
easy-to-install, C++ library that will do what I have
in mind?
(In case it's relevant, I run Mandrake Linux 10.1
Official, KDE, ALSA, and an old Sound Blaster Live. I
often go to the San Fernando Valley LUG, in Los
Angeles, California.)
Thanks in advance,
Jeff
__________________________________
Do you Yahoo!?
All your favorites on one personal page � Try My Yahoo!
http://my.yahoo.com
Hi!
There is an article on slashdot today about KORGs Linux based music-
workstation christened OASYS.
The java-script on KORGSs page kills my Mozilla (well, actually the
whole Xsession goes bunkers ...)
Is this machine somehow related to the prototype HiEnd keyboard Benno
introduced a while ago? Price and features seems to be about the
same ...
--
(
)
c[] // Jens M Andreasen
Hi all,
if you are not a gentoo user stop here, otherwise read on. All the english
linux-audio-lists please accept my apologies for crossposting to german
gentoo list. [DE: Gentoo-user-de, bitte vergebt mir das Crossposting an die
Englischen Listen.]
Today I decided to make my little but constant gentoo-portage overlay
available for the public. It contains only some apps not in already in
portage. Currently available are aeolus-0.3.1 with aeolus-stops-0.1.1,
fmit-0.9.[89], museseq-0.7.0, tuneroid-0.9.4 and (not an linux-audio-app)
ktechlab-0.1.2.
You can access the repository via svn, the address is:
http://roederberg.dyndns.org/svn/apps/portage-arnold/
If there is interest, I could also create a tarball, just ask...
Feel free to use it and feel free to send patches or to apply for write
access.
Currently I am trying to follow the anounces on this list, filter away the
apps, where gentoo.org is fast enough and add the remaining into my repo. The
chances are better if I personally use this app. :-)
Thanks for your patience,
Arnold
PS:
[EN: This will be the last time I do such a crossposting. Promised!]
[DE: Das ist das letzte mal, das ich so ein Crossposting mache. Versprochen!]
--
There is a theory which states that if ever anyone discovers exactly what the
Universe is for and why it is here, it will instantly disappear and be
replaced by something even more bizarre and inexplicable.
There is another theory which states that this has already happened.
-- Douglas Adams, The Restaurant at the End of the Universe
Hi,
QjackCtl 0.2.14 has been released. No big features, only a bunch of
optimizations and cleanups. Check it out:
http://qjackctl.sourceforge.net
From the change log:
- Put a limit on XRUN callback messages and statistics report rate,
preventing the potential hosing of the GUI due to a XRUN cascade storm.
The maximum reasonable report rate has been fixed to be one XRUN callback
occurrence per second.
- Set to ignore the SIGPIPE ("Broken pipe") signal, where available, as
the default handler is usually fatal when a JACK client is zombified
abruptly.
- All conection view items are now sorted in natural case insensitive
order, not just as audio port names as was before.
- Got rid of those nonsense paranoid and rather stupid pseudo-mutex
guards on the connections management framework and event notifications
(nuff said :).
- Optional confirmation warning on audio server shutdown, if there's some
audio clients still active and connected (as suggested by Sampo
Savolainen).
- Check for <jack/statistics.h> on configure time (as of JACK 0.99.42+ CVS).
- "Unlock memory" server setup option was added, allowing the release of
memory used by common toolkit libraries (GTK+, Qt, FLTK, Wine) that were
being superfluously locked on every GUI JACK client; number of periods
has now the minimum allowed value of 2; server start delay widget
converted to spinbox; setup dialog layout slighly changed.
- Removed stand-alone usx2y driver support. Since JACK 0.99.41+ CVS, the
special "rawusb" support on the Tascam US-122/224/428 USB Audio/MIDI
interface controllers have been merged and properly integrated into the
regular alsa backend driver. Being still experimental, this special mode
of operation is now triggered only when "hw:N,2" is specified as the alsa
device name (N = soundcard index of snd-usb-usx2y module).
- Messages window limit is now enforced only when the line count exceeds
in one third the user configured line count maximum; if Qt 3.2.0+ is in
use, the QTextView widget is otherwise set to the optimized Qt::LogText
format.
- XRUN status items are kept double-dashed if none has been detected.
Enjoy.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
