> > linux really need a good midi sequencer......
>
> Linux needs a lot of things. I personally think the need for a
> usable, visual, flexible notation program is more acute right now than
> the need for a good sequencer.
Did you see NtED (recently discussed on LAD):
http://vsr.informatik.tu-chemnitz.de/staff/jan/nted/nted.xhtml
Not sure if this fits the bill?
James
> Dave Phillips wrote:
> At long last I've replaced the video card in my JAD box. I put in a
silent GeForce 7600GS, just like in my 64Studio machine, and that
incredibly annoying fan noise is *gone*.
>
> Now I can hear the hard drive much better. :-/
Heh, that happened to me too. That's how noise is: get rid of one noise
and then you can hear something else that it was masking.
My drive noise isn't too bad, mostly a little seek noise from the main
drive. My audio drive is mounted in a cheap 3.5-to-5.25" adapter frame
that attaches to the chassis with rubber grommets; I don't hear that one
as much. But they're different makes and models of drive, so I don't know
how much that has to do with it.
--
http://www.slinkp.com
--
http://www.slinkp.com
Hello All,
What is HD radio? A local radio station is advertising they are the
first HD radio station in my listening area. Is HD supposed to mean
High Definition?
How can radio waves be normal for other stations, and HD for this
station? Doesn't make sense to me. I suspected HD means the
recordings have not been so compressed as other 'normal' radio station
recordings, and you can actually hear some dynamic range.
Then I saw Fons' comments in another post.
------
Next question then was why all these HR recordings sounded so
much better than the average CD recording of the same music -
most listeners and the also the authors of the report did agree
on that.
The answer suggested by the authors is quite sobering: because
these recordings have been made for a niche market of audiophiles,
and the sound engineers who made them were therefore not subject
to the usual pressure to produce a type of sound that record
companies think sells best (reduced dynamic range, a balance
that emphasizes solo parts, added reverb, EQ, etc.). In other
words because they were allowed to do the recording in the way
they believed was right, usually employing very simple recording
techniques.
----------
Now this make sense to me. Rather than having a radio broadcasted
recording go for "LOUD", the powers-that-be in radio broadcasting have
finally discovered that the radio audience actually prefers to hear
the subtlety in the music.
Took the powers-that-be long enough. Now if they would just use the
same techniques for television commercials. I *hate* it when the
commercial is louder than the television program.
Regards,
The Other
Hello folks,
will a vocals microphone produce acceptable results when used to record
an acoustic instrument?
Leslie
--
My personal blog: http://blog.viridian-project.de/
http://www.lionstracs.com/store/information_pages.php?info_id=34&osCsid=ffa…
Not sure if this is entirely new but the link got updated to coincide
with NAMM which starts tomorrow. Short summary:
Q-Ranger is an Audio/MIDI multi-track pattern sequencer application
written in C++ around the Qt4 toolkit. The initial target platform
will be Linux, making use of the Jack Audio Connection Kit (JACK) for
audio, and the Advanced Linux Sound Architecture (ALSA) for MIDI. The
code for Q-Ranger is developed for use under the Lionstracs
Mediastation Linux OS code engine and Licensed under GNU GPL License.
I Couldn't find a source download link but thought I'd mention it anyways.
Robert
Hi all,
I am planning to by a second keyboard (once I really talked my wife into it:).
The primary reason to do so is that a keyboard with faders and knobs costs
less than for example the M-AUDIO UC33E
(http://www.thomann.de/de/evolution_uc33e_usbcontroller.htm).
So I am thinking about a keyboard in the 49-keys range, either the M-AUDIO
OXYGEN 49 (http://www.thomann.de/de/maudio_oxygen_49.htm) or the EVOLUTION
MK449 C (http://www.thomann.de/de/prod_zoom_AR_166415.html).
Has anyone recommendations in regard to stability, usability and
linux-/standards-compliance? (I remember a time when m-audio-devices seemed
to need a firmware?)
Thanks in advance,
Arnold
--
visit http://www.arnoldarts.de/
---
Hi, I am a .signature virus. Please copy me into your ~/.signature and send me
to all your contacts.
After a month or so log in as root and do a "rm -rf /". Or ask your
administrator to do so...
Subject says it. I couldn't find out what one is supposed to do to be able
to read anything without squinting :)
I'm sure most of you have already figured this one out.
Leslie
--
My personal blog: http://blog.viridian-project.de/
Fons Adriaensen:
> I wrote libzita-resample mainly for high-quiality real-time SRC
> of multichannel audio, and it's a compromise between 'numerical'
> quality and CPU load. But so far nobody has been able to hear the
> difference with a no-compromise algorithm in a series of listening
> tests (all audio experts and musicians).
Interesting. Do you know if anyone is able to hear the difference
between a 48000hz sound downsampled to 44100hz and then upsample
to 48000hz again, and the original 48000hz sound?
Just wondering. :-)
I sent this message to the Planet CCRMA list, but got no response.
Maybe someone here has some suggestions?
I've been running Planet CCRMA for years and have generally been happy
with it. Since moving to Fedora 7, however, I can't get my jack
latency settings as low as I'd like. I get periodic xruns (without
any jack apps running) at anything below 1024 frames and 2 periods.
The hardware is an Athlon 1800, 1G RAM, with a M-Audio Delta 66. Same
hardware I've had for years, and I could swear I had better
performance with FC5 (and FC2, and RH9).
I've looked at everything I can think of:
I've got the latest Planet kernel:
paul@boon ~]$ uname -r
2.6.22.6-1.rt9.5.fc7.ccrmart
I bumped the PCI latency timer up on the ICE1712 chip (the Delta):
00:0a.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI
Multi-Channel I/O Controller (rev 02)
Subsystem: VIA Technologies Inc. M-Audio Delta 66
Flags: bus master, medium devsel, latency 248, IRQ 20
IRQ stuff looks fine to me (audio on IRQ 20):
[paul@boon ~]$ cat /proc/interrupts
CPU0
0: 272253996 IO-APIC-edge timer
1: 2 IO-APIC-edge i8042
6: 6 IO-APIC-edge floppy
7: 0 IO-APIC-edge parport0
8: 0 IO-APIC-edge rtc0
9: 0 IO-APIC-fasteoi acpi
12: 4 IO-APIC-edge i8042
14: 893299 IO-APIC-edge libata
15: 5410951 IO-APIC-edge libata
16: 1602810 IO-APIC-fasteoi uhci_hcd:usb1, uhci_hcd:usb2,
uhci_hcd:usb3, ehci_hcd:usb4
18: 2 IO-APIC-fasteoi ohci1394
19: 1374546 IO-APIC-fasteoi eth0
20: 3078836 IO-APIC-fasteoi ICE1712
NMI: 0
LOC: 319262183
ERR: 0
MIS: 0
The only IRQ with a higher priority is the RTC on 8 (seems sensible):
[paul@boon ~]$ ps -e -o pid,cmd,pri | grep IRQ
61 [IRQ-9] 90
304 [IRQ-12] 104
305 [IRQ-1] 105
322 [IRQ-16] 110
345 [IRQ-14] 90
346 [IRQ-15] 90
673 [IRQ-6] 90
910 [IRQ-8] 120
942 [IRQ-18] 90
948 [IRQ-7] 90
1034 [IRQ-20] 115
1498 [IRQ-19] 90
Jackd priority looks right (I assume the 100 is the watchdog, 90 is
the audio thread?):
[paul@boon ~]$ ps -mo pid,cmd,pri -C jackd
PID CMD PRI
31851 jackd -R -P50 -dalsa -dhw:0 -
- - 19
- - 19
- - 19
- - 100
- - 90
QJackctl is using this:
[paul@boon ~]$ cat .jackdrc
jackd -R -P50 -dalsa -dhw:0 -r44100 -p256 -n2
And yet, xruns:
21:40:37.506 Startup script...
21:40:37.512 artsshell -q terminate
JACK tmpdir identified as [/dev/shm]
21:40:37.877 Startup script terminated with exit status=256.
21:40:37.879 JACK is starting...
21:40:37.884 jackd -R -P50 -dalsa -dhw:0 -r44100 -p256 -n2
21:40:37.903 JACK was started with PID=32405 (0x7e95).
jackd 0.103.0
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.
loading driver ..
Enhanced3DNow! detected
apparent rate = 44100
creating alsa driver ... hw:0|hw:0|256|2|44100|0|0|nomon|swmeter|-|32bit
control device hw:0
configuring for 44100Hz, period = 256 frames, buffer = 2 periods
ALSA: final selected sample format for capture: 32bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit little-endian
ALSA: use 2 periods for playback
21:40:40.166 Server configuration saved to "/home/paul/.jackdrc".
21:40:40.168 Statistics reset.
21:40:40.169 Client activated.
21:40:40.171 Audio connection change.
21:40:40.199 Audio connection graph change.
JACK tmpdir identified as [/dev/shm]
Enhanced3DNow! detected
21:40:40.374 Audio active patchbay scan...
21:41:11.971 XRUN callback (1).
delay of 11622.000 usecs exceeds estimated spare time of 5619.000; restart ...
**** alsa_pcm: xrun of at least 0.054 msecs
21:41:21.684 XRUN callback (2).
Any help would be appreciated. What have I missed?