i am experiencing the "system can't keep up" messages
like crazy, even recording a single first track.
often when i speak into the microphone, what comes out
of the speakers sounds like it has a crazy chorus
effect on it.
i also wonder what the correct settings to use in the
alsamixer gui are.
below is some system info gleaned from a modified
shell script somebody posted on another list.
if anyone has any ideas please let me know! and if
there's any other system info i could provide.
thanks!
=========================
ALSA Audio Debug v0.0.2 - Wed Jun 2 11:06:32 EDT 2004
Kernel
----------------------------------------------------
Linux 66.149.213.131 2.4.22-21mm.2mdksmp #1 SMP Sun
Nov 2 15:20:57 CET 2003 i686 unknown unknown GNU/Linux
Loaded Modules
--------------------------------------------
snd-seq-midi 5632 0 (autoclean) (unused)
snd-seq-oss 37824 0 (unused)
snd-seq-midi-event 6784 0 [snd-seq-midi
snd-seq-oss]
snd-seq 51920 2 [snd-seq-midi
snd-seq-oss snd-seq-midi-event]
snd-pcm-oss 46852 0 (unused)
snd-mixer-oss 16016 1 [snd-pcm-oss]
snd-cs46xx 84232 1
snd-pcm 92128 0 [snd-pcm-oss
snd-cs46xx]
snd-timer 22404 0 [snd-seq snd-pcm]
snd-ac97-codec 51352 0 [snd-cs46xx]
snd-rawmidi 20128 0 [snd-seq-midi
snd-cs46xx]
snd-seq-device 6268 0 [snd-seq-midi
snd-seq-oss snd-seq snd-rawmidi]
snd-page-alloc 10228 0 [snd-cs46xx snd-pcm]
snd 50692 0 [snd-seq-midi
snd-seq-oss snd-seq-midi-event snd-seq snd-pcm-oss
snd-mixer-oss snd-cs46xx snd-pcm snd-timer
snd-ac97-codec snd-rawmidi snd-seq-device]
Modules Conf
----------------------------------------------
alias sound-slot-0 snd-cs46xx
above snd-cs46xx snd-pcm-oss
Proc Asound
-----------------------------------------------
Advanced Linux Sound Architecture Driver Version
0.9.8.
Compiled on Nov 2 2003 for kernel 2.4.22-21mm.2mdksmp
(SMP) with versioned symbols.
0 [CS46xx ]: CS46xx - Sound Fusion CS46xx
Sound Fusion CS46xx at
0xf8ffe000/0xf8e00000, irq 18
0: [0- 0]: ctl
8: [0- 0]: raw midi
18: [0- 2]: digital audio playback
17: [0- 1]: digital audio playback
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
1: : sequencer
33: : timer
00-00: CS46xx : CS46xx : playback 31 : capture 1
00-01: CS46xx - Rear : CS46xx - Rear : playback 31
00-02: CS46xx - IEC958 : CS46xx - IEC958 : playback 1
CPU
-------------------------------------------------------
model name : Pentium III (Coppermine)
model name : Pentium III (Coppermine)
cpu MHz : 864.478
cpu MHz : 864.478
RAM
-------------------------------------------------------
MemTotal: 513760 kB
SwapTotal: 1028120 kB
Hardware
--------------------------------------------------
00:00.0 Host bridge: Intel Corp. 82840 840 (Carmel)
Chipset Host Bridge (Hub A) (rev 02)
02:06.0 Multimedia audio controller: Cirrus Logic CS
4614/22/24 [CrystalClear SoundFusion Audio
Accelerator] (rev 01)
HDPARM
/dev/hda:
Timing buffered disk reads: 122 MB in 3.03 seconds
= 40.26 MB/sec
/dev/sda:
Timing buffered disk reads: 124 MB in 3.04 seconds
= 40.79 MB/sec
__________________________________
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hi all,
is there any possibility to sync ardour (probably using jack) with a
video file player? i want to compose a soundtrack to for some visuals...
cheer...
--
Tim mailto:TimBlechmann@gmx.de
ICQ: 96771783
--
The only people for me are the mad ones, the ones who are mad to live,
mad to talk, mad to be saved, desirous of everything at the same time,
the ones who never yawn or say a commonplace thing, but burn, burn,
burn, like fabulous yellow roman candles exploding like spiders across
the stars and in the middle you see the blue centerlight pop and
everybody goes "Awww!"
Jack Kerouac
Malcolm Baldridge writes:
>It's most likely a consequence of being run through a filter. Remember,
>filters generally REMOVE power rather than add it. And even though you're
>upsampling, some filtering is being done to eliminate synthetically created
>aliasing noise.
That turned out to be sort of what was going on. I noticed
the reduced audio on both the up-sampling script and a script that did
no sample conversion but was supposed to send one mono feed to two
stereo channels.
I tried the -e -stat function and it showed a gain factor of
1.0 in the mono to stereo script. I stuck a gain conversion of 2 in
the output and my level came right up to where it should be except for
lots of clipping.
After another look at the manual, I found that I had
overlooked the avg settings. These settings control how much audio is
sent to each channel. sox invokes this function but with defaults if
you don't set it and I had not. Apparently, the default is to send
half the audio to each channel.
avg 1,1 did the trick and my stereo from mono recording is now as loud
as the original mono .wav without clipping.
That script I included that produces a stereo 44100 .wav out
of 8000-HZ PCM probably also needs the avg settings adjusted. I just
didn't realize that I had lost some level at the time.
My thanks for answers that got me pointed in the right
direction.
Martin McCormick WB5AGZ Stillwater, OK
OSU Information Technology Division Network Operations Group
Hi all,
I've been sending two channels of audio via SPDIF from an RME Hammerfall
DSP multiface to to a Delta 66. Listening to the Delta, I notice some
little periodic glitches which seem to indicate a synch problem [these
glitches are not present when receiving from the same card via analog].
Can anybody suggest how I can troubleshoot this? Is it necessary to make
one a master and one a slave? If so, how is this done?
thx,
d.
--
derek holzer ::: http://www.umatic.nl
---Oblique Strategy # 72:
"Find a safe part and use it as an anchor"
I recorded a mono .wav file at 14400 samples per second and it
sounds loud and clear. Then, I got fancy and decided to convert it in
to a stereo file suitable for putting on a CD. It still sounds clear,
but about 3DB down from the original mono recording. Just guessing at
this, but it is noticeably lower in volume.
Here's how I did all that:
#! /bin/sh
arecord -d 454 -t wav -r 44100 -c 1 -f S16_LE \
| sox -t .wav -c 1 - -c 2 -w -r44100 outputcd.wav
The top half of that shell script originally outputted to a
file which was mono and did play at 44100 samples/sec.
If I take that file and use it as input to the lower half of
the script, I get exactly the same lower audio level so I know it
isn't the way I am piping the arecord output in to sox. Besides, that
is normal UNIX practice and there is no modification to the data.
I expected sox to simply duplicate the mono data in to both
left and right channels. Is this normal or do I need to tell sox to
increase the level. I may accidentally be causing sox to expect
another channel of audio which would be right if one was mixing two
channels. In this case, 100% of the mono signal should appear on both
channels as if I had a Y connector.
Sox is amazing in that I was able to produce a CD-ready audio
track from some unsigned 8-bit PCM audio and it sounded as good as it
did originally. This isn't all that good for anything but voice, but
the conversion didn't make it any worse. For those who are curious,
that script looks like:
#! /bin/sh
sox -t ub voxaudio.ub -t wav -c 2 -w -r44100 output.wav resample .95
I don't have any particular question about this script, but I
do wonder why my mono/stereo conversion is a few DB down?
Thank you.
Martin McCormick WB5AGZ Stillwater, OK
OSU Information Technology Division Network Operations Group
Hi:
I'm fairly new to ALSA, and I'm having a problem with capturing sound on
the SBLive cards line in jack.
Long story short: using alsa 1.0.5 with kernel 2.4.26. Recompiled all
the drivers including SBLive when I rebuilt the kernel. Ran alsaconf to
do automatic set up.
Altered the defaults like so:
amixer sset Capture 33%,33%
amixer sset Line cap 100%,100%
...and what I'm able to record is one channel of what's coming into the
Line In. I know both channels are coming in -- if I set the playback of
Line to 100% I can hear both channels fine.
Has anyone had this problem? I'd appreciate any pointers to a solution.
Thanks,
Doug
+-----------------------------------------------------------------+
| ______ ______ _ _ _ |
| /\ / _____) ___ \| | | | | /\ |
| / \ | / ___| | | | | | | | / \ |
| / /\ \| | (___) | | | | | | | / /\ \ |
| | |__| | \____/| | | | |___| | |_____| |__| | |
| |______|\_____/|_| |_|\______|_______)______| |
| |
+-----------------------------------------------------------------+
[Sorry for cross-posting. Feel free to forward around]
Florence, 08 June 2004
+++ AGNULA/DEMUDI 1.2.0-beta0 IS OUT
AGNULA/DeMuDi 1.2.0-beta0, the Debian-based GNU/Linux distribution for
audio/video, has been released.
+++
AGNULA/DeMuDi 1.2.0-beta0, the Debian-based GNU/Linux distribution for
audio/video, has been released.
This version is the first beta of the 1.2.0 series, which sport
tighter integration with Debian, using the Sarge Debian Installer and
the CDD (Custom Debian Distributions) framework.
You can download AGNULA/DeMuDi 1.2.0-beta0 here:
http://download.agnula.org/1.2/1.2.0/demudi_1.2.0-beta0_i386.iso
MD5SUM files are available here:
http://download.agnula.org/1.2/1.2.0/MD5SUMS
But PLEASE, PLEASE, PLEASE use the relevant mirrors:
http://freesoftware.ircam.fr/mirrors/agnula/ (IRCAM, Paris)
http://ccrma.stanford.edu/mirrors/agnula/agnula-iso/ (CCRMA, Stanford)
Please report all bugs, requests, criticisms using our development
portal [0]. Instructions on how to report bugs and requests are
available here:
http://www.agnula.org/development/agnula_bugs_requests/
We hope you enjoy AGNULA/DeMuDi! For any information, do not hesitate
to contact us writing to:
<info(a)agnula.org>
And/or visiting our web site, http://www.agnula.org/.
+++
About AGNULA: Agnula (acronym for A GNU/Linux Audio distribution,
pronounced with a strong g) is the name of a project funded until
April 2004 by the European Commission (number of contract:
IST-2001-34879; key action IV.3.3, Free Software: towards the critical
mass). After the end of the funded period, AGNULA is continuing as a
volunteer based project, aiming to spread Libre Software in the
professional audio/video arena.
Best regards,
--
The AGNULA Team info(a)agnula.org
Our mailing lists: http://lists.agnula.org/
Our web site: http://www.agnula.org/
"There's no free expression without control on the tools you use"
[0] http://devel.agnula.org/
> -----Original Message-----
>
> > find . -name "*.wav" -print -exec oggenc -q 5 {} \;
>
> It does exactly the trick I need for making .ogg files out
> of my wav's. Perfect! for ogg...
>
>
> hack through them. But maybe also somebody could explain to
> me what the above magic command line is doing so I can
> subvert it to my own evil purposes. ;-)
It doesn't look like anyone explained the command line to you so I'll do
it real quick.
First the entire command is based on find (one of the most underused and
oft looked over commands I have come across, granted I have only been
doing this for a year or so)
http://www.die.net/doc/linux/man/man1/find.1.html
so basically:
find
(name of the command)
.
(this directory)
-name "*.wav"
search for the pattern *.wav in the name of the files in this
dir
-print
prints the full filename of any matching files
-exec oggenc -q 5 {} \;
and the most powerful part of find, basically what this does is
that any files it finds run the following command line [up to the \;]
replacing {} with the files
So basically the commands runs through all the fiels in the current
directory and everytime it finds a wave runs
oggenc -q 5 mywave.wav
cool huh? I love linux :).
m.
_________________________________________________
Scanned on 01 Jun 2004 17:34:31
Scanning by http://erado.com
After a fun-filled weekend of doing nothing but downloading and
compiling software (after struggling with a flaky wireless card and
having to install a new one and build the drivers for it), I have
synchronization working beautifully between the two killer apps,
Rosegarden and Ardour (Ardour is the master, Rosegarden is the slave).
It works even better than expected, and the performance of everything is
way better than what I saw on Windows XP (same machine) using Cakewalk
and Finale (which I couldn't synch together at all). And the vast
number of plugins I have now... I would have had to spend a huge wad of
cash to get the same amount of VST of or DirectX plugins.
Using JACK and the applications that can use JACK remind me why I was
drawn to Unix-ish OSes in the first place so many years ago -- it's not
just the individual programs that are important, but the relationship
between the programs and how they work together that are important.
-- Brett