Hi folks,
I've been trying for months now to get arecord to work with the built-in
microphone and/or mic jack of my laptop but it just won't work. Since
the alsa developers seem totally uninterested in fixing or even
understanding this problem (yes, I've posted on alsa-devel many many
times) and the built-in A/D on the laptop is unlikely to be any good
anyhow I've decided to investigate an external A/D system. It's for a
laptop so it would have to be either PC card, USB, or firewire, not
PCI. Compatability and reliability are a big deal to me -- I'm fed up
with sending mail to alsa-devel, which seems to route to /dev/null as
far as I can tell. Can anybody recommend a solution that's really
simple to set up and reliable to use under linux and doesn't cost an
arm and a leg (e.g. less than $400)? I'm not too picky about number of
ins and outs or fancy features, more picky about quality,
compatability, ease-of-use, and reliability.
Thanks,
-n8
--
>>>-- Nathaniel Gray -- Caltech Computer Science ------>
>>>-- Mojave Project -- http://mojave.cs.caltech.edu -->
GuyCLO~ wrote:
>I agree. I found that larger latencies (100 < latency < 200) are usable for
>having fun. I mean I have used softsynths on a computer without tuning
>latency and without beeing root.
Hmm. How are you measuring latency? I'm not sure how to do it (sufficiently
accurately). I'm running SuSE 8.2, which some have suggested includes the
low-latency patch, but I don't think so. I've been too busy (lazy?) to check
or implement. I find even when playing some .ogg file to jam along with, that
hiccups are "disturbing" or distracting. Maybe I've got problems (as a
musician wannabe) with my timing? I've got my .ogg files on a server, but I
believe xmms pre-buffers (I recall setting it to 1/2 second at one point?)
its compressed audio stream, so I think I'm only hearing jitter from variable
interrupt response (and temporarily blocked interrupts?)?
Your other comment (in another post) about "real life lost packets" (UDP
comparison) is interesting. You would have to transmit a time code in each
packet, so the player can continually "re-sync" the audio. I don't think the
current audio streams do that? I think they "assume" (Benny Hill?) that the
audio stream is continuous, and therefore you can derive the timing?
--
Juhan Leemet
Logicognosis, Inc.
Hello again - I hope this is the right list for this topic (so far I can only
get subscribed to two and I keep bouncing back and forth between them...)
first off, may I say thank you to the list members for bearing me, and extra
thank you still to those who take the time to respond - amazing gentlemen and
ladies you all are; I realize I have written what amounts to several novels
worth of emails in the past couple weeks! :)
well I started a couple threads earlier about strangeness (or what I think was
strangeness) with compiling and installing hdspmixer and such...the questions
I had I didn't figure out, but somehow hdspmixer was able to run - whether
it's ACTUALLY working, I don't know for sure sure...
THAT having been said,
I only know of a couple people who are using the HDSP 9652, along with a
couple who've developed for it - Mark K, Thomas, Kevin, et al - does any of
this sound familiar? :
--
a) I've been doing a lot of back and forth and reading but I'm a bit confused
as to what hdspmixer is DOING (which makes it harder to tell if it's working
:) )
b) my goal, as I may have said on other threads, is to be running ardour with
my HDSP 9652, sending each of 24 individual adat optical channels out to 1
channel input in my behringer ddx3216 (outfitted with adat i/o). An earlier
thread I started was solved by me adding -d hw:0 to my jackd command line
(duh), where I wasn't seeing 24 (actually 26) possible outputs when I clicked
on "output" in the mixer in ardour
now I'm able to see the outputs, and assign them as they should be, meanwhile
in hdspmixer i've picked the preset which assigns ins, out and "playback"
(what would be the difference between "out" and "playback" I wonder) to adat
outputs, etc:
--
first I used hdspconf to change the HDSP to 44.1 for purposes of these tests
next I ran hdspmixer
I kept hdspmixer running, then started jack from a terminal using:
su
jackd -R -d alsa -d hw:0 -r 44100 -p 2048
then started ardour from a terminal:
su
ardour -n <--- that's so no splash screen will appear
but what was showing up at the board wasn't what I was expecting *laugh* if I
routed a track to channel 1, then it showd up a 3 db too quiet at channel 1
on the mixer, and at the right volume on channel 2 (I was using a -12 1khz
test tone that I generated with the board and was able to record to ardour -
ardour indeed is playing it at -12.1 - I think the point 1 is explanable but
I'll skip that 'cause it doesn't matter just now) if I routed the track to
channel 2, it didn't show up anywhere. odd numbered tracks after that seemed
to be showing up on 1-2 just like 1 did (although I can't confirm the
consistancy of this)
I experimented with opening up qjackconnect, which was the only one of the
many patch bay programs from planet that showed 24 capture and playback
channels and connecting captures to playbacks. this didn't seem to do
anything, which made me wonder just what qjackconnect was for.
then I closed all, closed the hdspmixer program, and started just jack and
ardour. the results from that are:
routed to channel 1, it shows up at channel 1, 3 db quiet, channels 3, 7, 9,
11, 17, 19 and 23 at 30 db quiet and channels 5, 13 and 21 at 12 dbs quiet -
wha?
routed to channel 2, it shows up only at channel 2! 3 dbs under - actually,
this 3 dbs may be a non-problem issue in the board, so possibly it can be
ignored
routed to channel 3, it shows up on 1, 2, 3 and 4 and some other channels at
half volume - wha wha?
channel 4, nothing
channel 5 --> 4, 5, 6 and others
ok you get the picture. randomness. a weird mess.
I don't know if this is a driver problem, a HDSP 9652/alsa driver patch
problem, an hdspmixer problem (I did, as I mentioned, have oddities on
compiling that program), a jack issue, an alsa issue, a bug in ardour, a bug
in alsa drivers, etc. etc. etc.
I only know of a couple people who are using the HDSP 9652 - Mark K, does any
of this sound familiar?
clearly there are thousands of details - anyone who wants to talk about this
and wants other info, as usual, ask and ye shall receive :) (although in
some cases, a request for details may yield first the question "how do I find
that?" :) )
thanks and I hope this is interesting and stimulating, maybe even
educational!!! :)
--
--------------
Aaron Trumm
NQuit
www.nquit.com
--------------
Hi,
Anybody know of an application that allows streaming of midi and/or audio over
the net for the purpose of allowing several people to jam together?
I seem to remember having heard of some such app for windows ages ago... but
that doesn't count does it? ;)
/Robert
trying to compile and install Thomas' hdspmixer 1.3 - anybody know what to do
about the following?:
[root@JamaisQuitteAudio hdspmixer-1.3]# ./configure
loading cache ./config.cache
checking for a BSD compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking whether make sets ${MAKE}... yes
checking for working aclocal... missing
checking for working autoconf... missing
checking for working automake... missing
checking for working autoheader... missing
checking for working makeinfo... missing
checking for c++... c++
checking whether the C++ compiler (c++ ) works... yes
checking whether the C++ compiler (c++ ) is a cross-compiler... no
checking whether we are using GNU C++... yes
checking whether c++ accepts -g... yes
checking whether make sets ${MAKE}... (cached) yes
checking how to run the C preprocessor... cc -E
checking for ANSI C header files... yes
./configure: line 1: fltk-config: command not found
./configure: line 1: fltk-config: command not found
checking for alsa/asoundlib.h... yes
checking for fltk-config... no
configure: error: fltk-config is required
[root@JamaisQuitteAudio hdspmixer-1.3]#
--
--------------
Aaron Trumm
NQuit
www.nquit.com
--------------
Hi,
I run across this problem a lot in Linux. Maybe someone can help me
make my Gentoo system do audio better. All the straight Alsa stuff works
well. However, when I'm browsing around I come to web sites that
probably want to play some audio, but this is Linux and Alsa, so things
don't work easily. At the web site
http://www.skale.org
I start getting this repeating glitch noise every 8 seconds or so. It
hasn't happened every time I've visited the site today, but it's
happened a lot of times.
In hdspmixer I see this glitch on all 26 playback channels, by the
way, so I'm assuming that this means it's some sort of OSS problem, but
I'm not sure.
What's wrong with my setup of Alsa, which in most other ways works
just fine, and why can I also not get mp3 (well, really xmms) to work
under Linux?
BTW - much non-Alsa audio from Games works fine, but also drives all
26 playback channels. Should it really work this way?
Thanks,
Mark
Let me first say I'm not terribly used to compiling Linux apps. I
usually intall Debian packages with apt-get. So I'm probably making a
very simple error, but I'm asking here because someone else might well
have built Audacity from source.
So here's the problem:
Whe I try compile Audacity 1.2 beta,
I get screefuls of messages like this:
libaudacity.a(PCMAliasBlockFile.o): In function
`PCMAliasBlockFile::BuildFromXML(wxString, char const **)':
/usr/include/wx/filename.h:100: undefined reference to
`wxFileName::Assign(wxFileName const &)'
/usr/include/wx/filename.h:100: undefined reference to
`wxArrayString::~wxArrayString(void)'
The system is Debian, mostly stable but I've installed quite a few bits
and pieces from testing. In response to earlier problems with
config and compilation I installed:
libwxbase2.4-dev
wxwin2.4-headers
libwxgtk2.4
zlib1g-dev
libwxgtk2.4-dev
But I can't see what's missing now and I don't understand enough about the
error messages, not being intimately falmiliare with libwx (it must have
something to do with that)
Any ideas, please?
--
Anahata
anahata(a)treewind.co.uk -+- http://www.treewind.co.uk
Home: 01638 720444 Mob: 07976 263827
On Monday 29 September 2003 12:50, Rob wrote:
>On Monday 29 September 2003 12:50, Robert Jonsson wrote:
>> Anybody know of an application that allows streaming of midi
>> and/or audio over the net for the purpose of allowing several
>> people to jam together?
>
>I seem to remember something like that for Windows too, but
>remember that latency that would be more than acceptable for
>gaming (30-40ms) could make it impossible to jam as you're
>envisioning.
Yes, I remember someone (non-technical) telling me about this great system
that allowed musicians on both coasts of North America to perform a "live"
piece together, across the internet. It was some kind of university project.
They even pulled out a newspaper or magazine article about it. I read it
through several times. I could not believe that you can get latencies (esp.
cross-continent) down low enough to allow "interactive jamming", where both
sides hear each other in real time.
Let's see (calculating on the back of an envelope): 3000 miles / 186000
miles/second... that's nominally 16msec... that could be tolerable... but
what about slowdown due to dielectric (speed of electric fields is less than
speed in vacuum)? what about delays in electronic circuits? what about
store/forward digital gear? At one point some traffic went via sattelite,
which adds 2 x 22K miles (or about 1/4 second). Hmm, that's why the delay on
some speech circuits (like when I phone my sister in the Dominican Republic)
is very noticeable! Lately, I think ground fibre is cheaper (and faster) than
satellite. For the moment, ignoring costs, I'm not even sure the network
wizards are able to splice together a dedicated circuit coast-to-coast with
audio hi-fi stereo bandwidth, even for "proof of concept". Even if they could
(like a permanent phone call?), there would be little point, because in a
digital (internet) network there would be real traffic, and hence variability
(jitter), which can only be smoothed by buffering and delay. Stutter is
usually worse than delay. So, I conclude that I'm mystified! Huh?
Now, if one is only concerned with one-way traffic, one can "cheat"! I
concluded from the article (and thinking hard about it) that they must have
used one site as a "reference site", piped their (partial) performance across
the continent (with whatever additional delay/buffering), and then had the
other orchestra "dub in" their part, and have that played at the 2nd site for
their "live" audience. Or the audience might have been at a 3rd site, at this
point it does not matter, just as long as it's not the 1st site! I seem to
recall that the audience was seated in an auditorium on 2nd coast. So, yes
they were "playing together" in some sense. And the audience was hearing the
performance "live". However, I cannot believe that the orchestra at the 1st
site was able to hear the 2nd site "live" at the same time they were playing?
Has anyone else heard about this? Details? Thoughts?
p.s. I used to do some sound recording for 16mm newsreel film stuff, decades
ago. Have you (with headphones on) ever tried to speak into a Nagra tape
recorder with a true read head after the write tape monitoring head? You hear
yourself about 1/2 second later. I had to take my headphones off, so I
wouldn't hear a delayed "echo". I think that is also true for "real" (long
delay) echo in recording? It can be paralyzing! Is that like stutterers?
p.p.s. I have recently been thinking a bit about psycho-acoustics, as I'm
(re)learning some guitar playing. If you consider nerve transmission speeds,
being able to play those real fast weedle-weedle-weedle guitar leads would
seem impossible. What must be happening is that you are telling your fingers
to move a fraction of a second before they actually move. Now add in the long
echo delay, and I suspect that's too much to handle: 3 time bases: what you
want to play, what you are playing (feel?), and what you hear. Comments?
--
Juhan Leemet
Logicognosis, Inc.
Hi, I am very stuck, I have been trying hard for a long time to try to
get digital out sound working on my Gentoo Linux kernel 2.6 test-5
machine. I do not have any conventional speakers, just an optical
digital lead going from a spdif out to an external dolby digital reciever.
Can anyone please describe the exact process to me of getting digital
out working on alsa drivers using a 2.6xx kernel? I am aware that the
2.6x kernels come with alsa included and I have enabled it and my
via82xx soundcard as modules.
Your advice is welcome.
Q
