Hi all,
I just upgraded to Mdk 9.1 and reinstalled all of the MIDI-related
stuff, as well as the Alsa 0.9.2 (from sources, after uninstalling Mdk's
RPM's). I also installed Fernando's modified alsasound script that
enables smart unloading of snd- modules.
The end result is that the midisport works, well sort of.
The firmware loads fine, and the midisport is properly recognized. The
problem is that I only have one MIDI port available even though this is
the 2x2 version (I had 2 ports in Mdk 9.0, usually /dev/midi(0) and
/dev/midi1). Now I only have /dev/midi.
The interesting thing is that the /proc/asound/card1/midi0 lists the
following:
Output 0
Tx bytes : 0
Output 1
Tx bytes : 0
Input 0
Rx bytes : 5 (this number changes, most likely because
it's the only port that has anything hooked up to it)
Input 1
Rx bytes : 0
So, it seems that there are 2 I and 2 O ports, but they are all
allocated within the midi0 device according to the proc stuff (or
/dev/midi1 according to the dev stuff). If this is the case, then how do
I access the second port.
Cat /dev/midi1 only outputs stuff coming into the first port, so that
tells me something's fishy.
I am using CCRMA's hotplug 8_26_1 and ezusbmidi 2002_11_17.
There were no alterations to the kernel or any other part of the system
(apart from the alsa install and other audio software installs).
My modules.conf looks like this:
# ALSA portion
alias char-major-116 snd
alias char-major-14 soundcore
alias snd-card-0 snd-intel8x0
alias sound-slot-0 snd-card-0
alias snd-card-1 snd-hdsp
alias sound-slot-1 snd-card-1
alias snd-card-2 snd-usb-audio
alias sound-slot-2 snd-card-2
options snd major=116 cards_limit=4 device_mode=0666 device_gid=0
device_uid=0
options snd-intel8x0 index=0
options snd-hdsp index=1
options snd-usb-audio vid=0x763 pid=0x1001 index=2
# CS4205
alias sound-service-0-0 snd-mixer-oss
#alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
#alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
#HDSP
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-8 snd-seq-oss
alias sound-service-1-12 snd-pcm-oss
#MIDISPORT
alias sound-service-2-1 snd-seq-oss
alias sound-service-2-8 snd-seq-oss
In this situation the hdsp is not in use, so midisport assumes the 2nd
card position.
Upon midisport reconnect, the /var/log/messages spits out the following:
Apr 2 23:59:49 badass-bukvic kernel: usb.c: USB disconnect on device
00:1d.2-1 address 29
Apr 3 00:00:00 badass-bukvic kernel: hub.c: new USB device 00:1d.2-1,
assigned address 30
Apr 3 00:00:00 badass-bukvic kernel: usb.c: USB device 30 (vend/prod
0x763/0x1001) is not claimed by any active driver.
Apr 3 00:00:03 badass-bukvic /etc/hotplug/usb.agent: Setup ezusbmidi
for USB product 763/1001/1
Apr 3 00:00:03 badass-bukvic /etc/hotplug/usb.agent: Module setup
ezusbmidi for USB product 763/1001/1
Apr 3 00:00:03 badass-bukvic /etc/hotplug/usb/ezusbmidi: load
/usr/share/usb/ezusbmidi/ezusbmidi2x2.ihx for 763/1001/1 to
/proc/bus/usb/002/030
Apr 3 00:00:03 badass-bukvic kernel: usb.c: USB disconnect on device
00:1d.2-1 address 30
Apr 3 00:00:03 badass-bukvic kernel: hub.c: new USB device 00:1d.2-1,
assigned address 31
Apr 3 00:00:07 badass-bukvic /etc/hotplug/usb.agent: Setup
snd-usb-audio audio usb-midi for USB product 763/1110/1
Apr 3 00:00:07 badass-bukvic /etc/hotplug/usb.agent: ... blacklisted
module: audio
Apr 3 00:00:07 badass-bukvic /etc/hotplug/usb.agent: ... blacklisted
module: usb-midi
Apr 3 00:00:07 badass-bukvic /etc/hotplug/usb.agent: Setup alsasound
for USB product 763/1110/1
Apr 3 00:01:00 badass-bukvic modprobe: modprobe: Can't locate module
sound-service-2-2
(NB: what's up with the sound-service-2-2, is there such a thing?)
Upon doing this, cat /dev/midi1 is the only one that works.
Other pertinent info is (in abbreviated form):
/proc/asound/version -> ALSA=0.92 kernel=2.4.21-0.13mdk
../cards -> 0 i810, 1 2x2 (Midisport) at usb-00:1d.2-1
../card1/ has id and midi0 files only
All I also remember the last time when I was using 0.9.0 is that I had
to upgrade from CVS in order to get both ports (it was the snd-usb-audio
bug IIRC). But now, I did d/l and install 0.9.2 which should have those
fixes but I can't seem to get the second port to work. Could it be that
there's something leftover from the mdk 0.9.0 version of Alsa even after
I did rpm -e and reinstalled 0.9.2 from sources?
Any help on this issue is greatly appreciated! Sincerely,
Ivica Ico Bukvic, multimedia sculptor
http://meowing.ccm.uc.edu/~ico
i got my new sound card today. It tests fine in Windows XP. Now
i want to get it working in Linux. (i've heard rumors that the
alsa driver hasn't been updated for the latest firmware.)
i'm running kernel 2.5.66 (alsa 0.9.2).
My card is marked "Dec 2002 Rev 1.3".
snd_hammerfall_mem allocates memory for 1 card. The hdsp driver
seems to load without errors, but the card doesn't appear
under /proc/asound/.
If there is a patch available then point me to it and i'll test it.
Thanks ...
--
Victory to the Divine Mother!! after all,
http://sahajayoga.orghttp://why-compete.org
Hi,
Every time I get this wonderful magazine in the mail it's like
Christmas all over again. I think this is the best magazine I read on
recording, bar none, and believe it or not, it's free!
For those of you that love this stuff like I do, go check out
http://www.tapeop.com and sign up for a free subscription. I don't know
if they mail over seas, but I hope so. If not, find out where you can
buy a copy. It's well worth it. Nearly 100 pages of articles written by
engineers and musicians. Reviews done by people really using the
equipment. The letters section is often surprising, like a recent letter
to the magazine from Pete Townshend.
Great stuff.
Mark
Greetings,
Can anyone give me an idea how to record the audio from a quicktime
movie? I've had success recording mp3 streams using grecord or xsox,
and thought I could do the same with qt, but no go. I'm using the
crossover plugin to play the movie in the browser.
Thanks,
James
--
James Hughes
jhughes(a)kos.net
http://jpathu.net/
Hello,
I just started playing around with RTMix. But I can't figure out the real-time MIDI stuff in RTMix. I've set it up to read from the correct MIDI port. The MIDI is working and MIDI logging enables me to see incoming MIDI. However, MIDI events seem not to trigger any action. I'm probably missing some basic understanding...
this line:
text([>say]={"MIDI received"} [>at]={"rt[0]"} [>kill]={"none"} [>accel]={"midi: 128, 255, 0, 127, -1, 127"});;
doesn't print 'MIDI received' no matter what I do. With MIDI logging on I can see all sorts of midi messages flying by. The same thing using [>accel]={"a"}, for instance, works as expected.
Also, I was under the impression that if I filter MIDI events the log will only display the events that get through. However, the line below results in the exact same log as the log above.
text([>say]={"MIDI received"} [>at]={"rt[0]"} [>kill]={"none"} [>accel]={"midi: 144, 144, 0, 127, -1, 127"});;
I'm lost.
./MiS
--
_
__ __ (_)___ Michal Seta
/ \/ \ _/^ _|
/ V |_ \ @creazone.32k.org
(___/V\___|_|___/
http://www.[creazone]|[noonereceiving].32k.org
On 01 Apr 2003 22:57:28 -0500
Nick <nicktsocanos(a)charter.net> wrote:
> I am writing audio applications that are real-time.
> I had been using Port Audio which is really very good, even without root
> priviledges, it does a very good job. I want to support other sound
> drivers, in case someone doesn't have Port Audio or can't get it to
> work.
>
> I have just built my ALSA driver, it works ok, but I am getting terrible
> latency problems and buffer underruns. The thing is it is a real-time
> application, it must get audio to the card as soon as possible. Why is
> it that OSS with Port Audio works so well, and my ALSA setup is
> suffering?
>
> Is there something I need to do to ALSA to get near real-time
> performance from it? I have heard people mention patching their kernel
> for low latency. I have Jack but I have not started working with it yet.
> Should I dump ALSA and just go for Jack? Is there something I need to do
> to my kernel to get better performance? I would get Planet CCRMA if it
> supported my OS, but it doesn't, so I have to tweak my box manually.
High Nick !
At the moment there are two different approaches for low-latency in the
linux-kernel.
The first is from Ingo Molnar; it's quite simple but efficient (conditional
scheduling at critical places).
You can get it at
http://people.redhat.com/mingo/lowlatency-patches/lowlatency-2.4.0-test7-A0
The second is made by Robert Love, it makes real heavy Changes in the Kernel
which have a noticeable negative impact on overall System-througput, but the
results are quite amazing !
The Patch is integrated int the 2.5-Kernel (CONFIG_PREEMPT), patches for 2.4
are available at
ftp://ftp.kernel.org/pub/linux/kernel/people/rml/preempt-kernel/v2.4/
I use Robert's patch with the 2.5-Kernel since half a year without Problems;
The interactive response of the Sytem under high loads is now MUCH better.
Lacking real real-time Software i don't know if it helps there too.
If the low-latency patches for the Kernel don't cure your Problem, consider
testing a new 2.5.6x-Kernel; the scheduler was really improved in the last
months ...
Hi,
Thanks for quick the feedback.. Sox seems to come the closest to what I want
such a program to do.
---
Roel / Utopia Sound Division
http://www.utopiasd.com
I am writing audio applications that are real-time.
I had been using Port Audio which is really very good, even without root
priviledges, it does a very good job. I want to support other sound
drivers, in case someone doesn't have Port Audio or can't get it to
work.
I have just built my ALSA driver, it works ok, but I am getting terrible
latency problems and buffer underruns. The thing is it is a real-time
application, it must get audio to the card as soon as possible. Why is
it that OSS with Port Audio works so well, and my ALSA setup is
suffering?
Even if I set the frame rate higher (the number of frames to the card) I
still get buffer underruns. I am talking maybe 70 milliseconds now. This
is sad, because I know with DirectX I can get about 35 ms timing with it
on this machine.With ASIO I have done better than that.
Even running as root and scheduling the app to real-time priority (50) I
get buffer underruns. So, something must not be right with my setup.
I have tried setting the frames written to all sorts of latencies, and
it does reduce the drop outs, but I am still getting them. I tried
buffering the sound and dropping it in blocks, but it wasn't working
correctly. This is a bit disappointing so far.
Is there something I need to do to ALSA to get near real-time
performance from it? I have heard people mention patching their kernel
for low latency. I have Jack but I have not started working with it yet.
Should I dump ALSA and just go for Jack? Is there something I need to do
to my kernel to get better performance? I would get Planet CCRMA if it
supported my OS, but it doesn't, so I have to tweak my box manually.
Thanks
--
Nick <nicktsocanos(a)charter.net>
Hi,
I'm looking for a command-line/console mode program which is available for
Linux and Windows which can resample samples (e.g. 48 Khz to 22 Khz) and to
some 'equalizer' processing to those samples. The result should of course be
written to the disc again. I'm looking for such a program for the use in the
installer of a commercial set of piano instruments which we want to make
available under windows and linux (currently only in the soundfont format,
but the product isn't released yet).
The program doesn't have to be free to use and I'm willing to pay for it's
use as long as the price isn't above my budget (around 100 euros).
---
Roel / Utopia Sound Division
http://www.utopiasd.com
Hello all,
I'm preemptivly asking a question that'll hopefully be put to
good use this week. I found a nice deal on a Delta44 off Ebay the other
night, and jumped on it. Can't wait until it shows up!
However, I'd like to *add* it to my machine, and keep me SB Live
for some stuff like everyday listening and perhaps some cheap output
monitoring. I've had friends tell me this doesn't work in Windows
because Windows will only record at the lowest-common-denominator
capabilities of all cards in the system (ie, 16/44 in a system with an
Delta44 and an SBLive).
Is this an issue with either ALSA or OSS in Linux 2.4? Also, I
know there are emu10k1 audio routing utilities, and I know there are
envy24 audio routing utilities. Is there a way I can route audio
between the two cards at all?
--
Ross Vandegrift
ross(a)willow.seitz.com
A Pope has a Water Cannon. It is a Water Cannon.
He fires Holy-Water from it. It is a Holy-Water Cannon.
He Blesses it. It is a Holy Holy-Water Cannon.
He Blesses the Hell out of it. It is a Wholly Holy Holy-Water Cannon.
He has it pierced. It is a Holey Wholly Holy Holy-Water Cannon.
He makes it official. It is a Canon Holey Wholly Holy Holy-Water Cannon.
Batman and Robin arrive. He shoots them.
