On 5/9/17, remu kelly wrote:
>
> How this can be achieved, seeing that we can't have a plugin after dmix.
Couldn't you use snd-aloop and have a plugin AFTER dmix? It basically
creates a loopback interface who's output channel is the input
channel. Although I've never used it (yet). But one way
Does the playback file exist on a slow storage device? I have a few
usb sticks that pause with cd quality wav files because of the slow
I/O of the device. But the same file compressed to flac or mp3 on the
same storage device will play without pauses. Or the cd quality file
on any "faster" devic
> You can mix all 4 inputs down into one stream and then record that, but
do you really mean that you can record to 4 separate application threads
concurrently without mixing?
No, I mean you can record all four channels as input at the same time
with the same app. Hence the -c 4 aka 4 channels.
It can record from all 4. Although many applications only care about
left / channel 1 (defaults). I tend to run pulseaudio over jackd
setup. As that was the only way to have your mic be an input other
than 1 for apps like skype. Since channel 1 and 2 are typically
stereo output.
$ man arecord
I've used alsa and firefox. By default java (in debian) is configured
for pulseaudio.
FILE: /etc/java-7-openjdk/sound.properties
But both alsa and pulseaudio configs are in there (in debian). Just
comment out pulse and uncomment alsa. I switch between a lot,
depending on if I am home or using
As previously said pavucontrol to configure pulseaudio.
BITD the default sound card was index 0. Which could not be
overridden by some things. So re-indexing was the desired way to
override things. These days most things respect the .asoundrc. And
you can have a pretty short one to change your
You might also blacklist snd-pcsp
Or maybe purge and re-install the alsa-base stuffs.
# dpkg --purge --force-all alsa-base
# apt-get install alsa-base
Make sure those needed blacklist items are there and add them if need be.
blacklist pcspkr
blacklist snd-pcspkr
blacklist pcsp
blacklist snd-pcs
Just look for USB 2.x, "class compliant", 24bit + 96kHz, and for 8
channels something like 7.1 surround which is technically 8 channels.
M-Audio has a Sonica Theater, which is USB and 7.1. Which might work,
as I've seen mention of it on the pulse-audio topics of discussion.
But I don't have such
ooking for,
> but rather how do I get the "configure" script to simply look in
> /opt/staging/alsa for the directory hierarchy and header files during
> the compile process? Unless I'm completely misunderstanding what you're
> saying, it appears as though we're
plughw is probably better at sharing a device than hw would be. And
plughw probably allows for some conversion of content. Otherwise they
are functionally the same IMO. Not that I'd know since I haven't
really delved that deep into things. You might check lsof or fuser to
see if something is us
stalled) is failing to compile because
> it's looking under /... for the header files and not
> /opt/staging/alsa/... Is there a way to make the configure script look
> into that directory for the header files during the "configure" phase?
>
> Thanks again for everyone&#
Ummm. I'm not sure if I follow you.
$ make
will build the objects and stuff in the current path of your source tree.
$ make install copies the executables to the system usable locations.
/usr/bin/ /lib/modules/. /usr/share/doc/.
(which is why you need to be root in a lot of cases to run
If you're really into going it on your own. There's gentoo, and
there's LFS aka linux from scratch. Both of which impose a lot of
source compilation. The inherent problem with sources is that you run
into maintenance issues. i.e. If you use the same install for a long
enough time, it'll eventua
it's still required to install the alsa-driver
>>> package. That being said, I'm going to run into the same problem as "A"
>>> above since the version of Kubuntu I'm using to build the custom distro
>>> isn't using the same kernel version. So w
o. Not only that,
> but knowing details at this level of building an OS can also help with
> my job - so I get a two fold benefit. :) Otherwise, I'd definitely
> follow your advice! lol
>
> Thanks again for your help, I look forward to hearing back from you.
>
> Dave
>
&
A) If you want to overwrite your existing distro's versions, you
probably want the --prefix=/usr option on your ./configure commands.
If not, be sure to change your $PATH to look at /usr/local FIRST.
B) Compile alsa-lib first, alsa-driver second. Most compile options
only need --prefix=/usr if yo
It depends on the HDMI device. For my video card, the specification
of the sound that travels over that wire is pretty strict. ONLY AC3,
only 44.1kHz, only stereo / 2 channels, only... And it does work if
all criteria is met. But I much prefer to use the analog audio
(lossless / PCM). But a si
A little overkill from my description. And so forget that versioning
would alter the module sizes. But alsa-info has the needed info.
Knoppix - Kernel 2.6.37 - alsa 1.0.23
Debian - Kernel 2.6.32 - alsa 1.0.24
Is that the way your debian came, or did you try to fix things
manually? Just an odd
Finding out what's different with more detain than Yes and No would help.
boot into knoppix
$ lsmod | grep -i "snd" | sort 2>&1 | tee alsa_knoppix.log
boot into debian
$ lsmod | grep -i "snd" | sort 2>&1 | tee alsa_debian.log
save these files on a common medium (flash drive) of course.
$ diff
gh it's still the
only way to tell some soundcards to record from PCM out IME. I've
never found a way to affect that setting in any other way, in the
manner needed. Even though I can see the effect of that change in the
output of amixer.
HTH,
- James
On 5/20/11, Y P wrote:
>
That seems pretty regular at 8 to 10 minute intervals. Do you live
near a subway line? Or other electric mass transit option? Is the
computer on a UPS or power conditioner type supply line? I get a
spike like that when I use a battery box to power an electret mic. If
I turn it on after pressin
The first step would be to see if it's even an ALSA issue. With flash
video (youtube) there's a speaker icon and a slider which affects the
volume. I recently noticed hulu had my levels way low with such an
icon. With mplayer there's a softvol option which might differ from
the levels set in als
It depends on what you need. A majority of the cheap USB interfaces
are USB 1.x and only do 2 channels. USB 2.x only recently got an
audio standard (< 5 years) and devices that are starting to use that.
With USB 3.x already being out in the wild of sorts. IMO, if you need
more than 2 channels, y
> Please reread my message in this thread on 'sox' - it contains the complete
> command line I've used.
So apples to oranges? since your sox only does 4 seconds (trim 1 5)
and your arecord does 6 seconds -d 6. Statistically that's 50% more
opportunity for failure in arecord.
-
Did omitting
> No, I'm not trying to capture content from browsers; the browsers have
> no relationship to what I'm doing.
If it's running on the same computer at the same time, there is a
relationship. i.e. Fewer resources. An xrun is a lack of resources.
(or a bug)
> arecord -D hw:0,2,0 -c 2 -r 96000 -d 6
Low latency kernel?
xruns are basically a resource issue. Web browsers have flash and
java and javascript scripts that loop for infinity and other things
that strip you of your resources. Basically I'd start by closing your
browsers while recording. If you're trying to capture content from
the
http://ubuntuforums.org/showthread.php?t=286016
does that one help? Appears common to need to do a card reset for some reason.
If that doesn't work, you might try the snd-hda-intel driver, versus
the snd-intel8x0 that it says you're using. I don't know which of
those drivers go with that card.
Try:
speaker-test -c 2 -D hw:0
where hw:# is the number of your card as it shows in /proc/asound/cards.
snd-ice1712 is the driver. With a Delta 44 myself. Pulse-audio
doesn't play nice with it, so disable that if reasonable. Most apps I
use interface with alsa or jackd directly, so I just lea
ll get
> these. But just to make sure I'm talking about the same one-- what's
> the difference between the oldest one you referred to and the one
> you've got?
>
> And has anyone had success with the new shiny little one with top knobs?
>
> Thanks,
> J
I have the mobile pre (old one, but not the oldest one). It just
works. USB compliant, at least for USB 1.x standards. i.e. 2
channels input, 16 bit, 48kHz max. The gray one with buttons on
front, and pretty much any analog connection type known to man.
Although the line input (3.5mm) does not
In theory you don't need the dmix thing anymore. If your applications
use ALSA natively, it will automagically mix sound from several
applications (in software). If the applications use OSS, you can
force it to use alsa with aoss. BITD you'd run esddsp or artsdsp -m
to do this sort of thing. D
-f cd is a shortcut for a STEREO track. AFAIK, the output for arecord
is ONE file, with many channels in it.
i.e. -f cd == -f S16_LE -t wav -c 2 -r 44100
and
i.e. -f cdr == -f S16_BE -t wav -c 2 -r 44100
(what it gets converted to before burning a disc)
or something like that...
$ arecord -t wav
Well depending on HOW it was obtained. The short answer is that the
kernel primarily installs to only TWO locations.
/lib/modules/`uname -r`/
and
/boot/
So check for the #.##.## of your kernel version in those locations.
Also note a few symlinks /boot/config / boot/system /boot/kernel that
mig
work anymore...
- James
On 2/12/11, Bill Unruh wrote:
> On Sat, 12 Feb 2011, Marcin Szyniszewski wrote:
>
>> On Sat, Feb 12, 2011 at 16:26, James Shatto wrote:
>>
>> $ sudo depmod -a
>> $ sudo modprobe snd-hda-intel
>> WARNING: Error inserting snd_timer
&g
> $ sudo dpkg -l '*alsa*'
> Desired=Unknown/Install/Remove/Purge/Hold
> |
> Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend
> |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad)
> ||/ Name Version
> Description
> +++-=
As I suspected, the modules aren't loaded so alsa isn't even running.
Hence your original "open" error(s).
How did you install alsa? Not that I think it is your issue, but it
could be. If you boot with lilo, you need to re-install lilo after
creating a new kernel. Even if it's technically the s
. By all means nitpick that I used grep -i
"audio", versus awk '{ print $4 " " $9 }' | grep -i audio or
something.
- James
On 2/11/11, Bill Unruh wrote:
> On Fri, 11 Feb 2011, James Shatto wrote:
>
>> Note that * is a wildcard. So /dev/dsp* is any d
Note that * is a wildcard. So /dev/dsp* is any devices that start
with /dev/dsp.
It looks like you don't have the modules loaded.
# modprobe snd-hda-intel
$ sudo modprobe snd-hda-intel
(depending on your distro / $ is user / # is root)
It might be /etc/init.d/alsasound or other named thing dep
Most times when I get something like that it has to do with the
/dev/'s not being present. Could be that udev isn't running on your
box. Or isn't configured for alsa. It could also be something else
like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss
snd-seq-oss. Basically canno
* * *
29 * * *
30 * * *
- James
On 5/30/10, Jaroslav Kysela wrote:
> On Sat, 29 May 2010, James Shatto wrote:
>
>> My debian distro comes with a 2.6.26-2-686 kernel. Which has version
>> 1.0.17 of alsa. I was hoping to just install the 1.0.23 version from
>> alsa-project
My debian distro comes with a 2.6.26-2-686 kernel. Which has version
1.0.17 of alsa. I was hoping to just install the 1.0.23 version from
alsa-project.org. But the links to download the sources don't appear
to work. Is the ftp site down? Is there some other way to get these
sources without ext
On Tue, 5 Jan 2010 01:07:21 +0300
An St wrote:
> Hello!
>
> Please help. I can't get working microphone at HDMI output.
HDMI audio normally has some sort of limit in place. For my ATI HD4550 video
card, the audio has to be transmitted in an AC3 codec(5.1 surround). AKA
compressed, it will n
On Wed, 9 Dec 2009 09:23:44 -0200
Kazuo Teramoto wrote:
> On Wed, Dec 9, 2009 at 5:16 AM, James Shatto wrote:
> > You can set the record device to PCM (aumix term, never been able to find
> > the equivalent alsamixer way). Although you'll likely need to adjust your
> &
On Tue, 8 Dec 2009 21:26:14 -0200
Kazuo Teramoto wrote:
> Hello.
>
> I like to redirect the sound I hear in the speakers to microphone, so
> it can be recorded with e.g. arecord.
You can set the record device to PCM (aumix term, never been able to find the
equivalent alsamixer way). Although
On Tue, 03 Nov 2009 18:25:24 -0600
"Jonathan E. Brickman" wrote:
> OK. I now find myself happily educated in "card" names (HD2 in my
> case), "devices" as being items on cards (HD2,0 et cetera), and
> "subdevices" whose names appear to be used in rather different
> locations. My next questio
On Mon, 2 Nov 2009 20:42:48 +0100
"Y.A. Bolawy" wrote:
> Hi all,
>
> I'd like some advice on a USB soundcard. The reason for getting one is
> that I'd like to have good quality sound on all the computers I use or
> will use. The quality should be good enough to allow speech
> recognition. Of cou
On Fri, 28 Aug 2009 10:38:16 +0200 (CEST)
Julien Claassen wrote:
> Hi!
>I'm not sure, if alsa does it, still. But you can do it with jackd (Jack
> Audio Connection Kit). It's a low latecny audio server and a lot of Linux
> Audio software support it. You can find packages in your distro.
>
> (/lib/modules/2.6.24-19-generic/ubuntu/sound/alsa-driver/acore/seq/snd-seq-device.ko):
>
there's your problem
Alsa from source will likely install to:
/lib/modules/`uname -r`/kernel/sound/
Which means you likely have two versions:
/lib/modules/`uname -r`/ubuntu/sound/
find /lib/modules/
On Mon, 29 Sep 2008 23:22:27 -0500
John Beavers <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> I have a few problems. My main problem is that I have installed an
> M-audio Delta 66, and it will output no sound, and does not recognize
> input from sound sources, either. But before we get to that
> I need to make multitrack recordings; I' m looking for a sound card
> usb2 model
> of at least 4/6/8 balanced inputs, XLR with phantom power to 48V and
> audio resolution 24-bit/96kHz
> and with many analog audio outputs maybe XLR balanced, SPDIF in / out
> and MIDI in / out / trough.
For t
> Hello everyone! I now resigned to not being able to use
> the m-audio fast track ULTRA usb soundcard with my "LINUX-DAWs",
> someone can recommend another card usb I can afford
> multitrack audio recordings of quality, which is working with Linux?
As said before, my M-Audio Mobile Pre seems clas
On Thu, 28 Aug 2008 17:03:44 +0200
brunal <[EMAIL PROTECTED]> wrote:
> I have to precise that I'm using a M-Audio fatst track pro sound
> card, which works fine if I only record stereo files.
Is there some reason the application doesn't do mono? Almost all I've seen
allow you to record only o
> I ended up recompiling an old version of xmms from source under Debian
> to get the real-time support I wanted, then compiled plugins from source
> as I needed them, using the checkinstall utility to create .deb files.
>
> Since this is the alsa-user list, does anyone know of current media
>
> > mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob
>
> I'd have figured that out :-) Didn't work with the 96/24 audio...
Well, you could probably do the arecord method.
arecord -D copy -t wav -c 2 -f S24_BE -r 96000 audio_track.wav
(unverified syntax)
Set record to the PCM / VOL devi
> Those are the two I have here. mplayer was giving me endless grief
> actually ripping the tracks from the DVD so I haven't yet done that in
> fact, but:
mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob
Granted that my default distro supplied version of mplayer didn't work for
this.
It looks like this might be your issue:
> La periferica di riproduzione è hw:0
> I parametri dello stream sono 48000Hz, S16_LE, 2 canali
> Using 16 octaves of pink noise
> ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition
> 'defaults.namehint.extended'
> ALSA lib conf.c:3513:(_s
It looks like you have all of the parts as far as kernel modules loaded. What
are you trying to use the the card with? What does /proc/asound/ say about the
card?
cat /proc/asound/cards
speakertest -c 2 -D hw:0
(change the 0 to match your cards index number, since usb probably isn't the
pri
I picked up a Delta 44 (pci / ice1712) off of craigslist for $100. I would
check to see what they need first. The Delta is a fine card, but by the time
you add in a microphone preamp, headphone preamp, and stuff just to record /
playback stuff. Not to mention the costs of cables and adapters
On Tue, 19 Aug 2008 01:28:46 +0200
V Gabriele De Palo <[EMAIL PROTECTED]> wrote:
> Has someone got the Fast Track Ultra from M-Audio working with linux?
I have an M-Audio Mobile Pre working. I just used it this morning to digitize
some judging tapes.
modprobe snd-usb-audio
If it's USB cla
> I started to experimenting with Software Defined Radio so I need some higher
> quality sound card. I could buy Creative SB Audigy SE which has 24bit stereo
> sampling @96kHz but it is not yet supported.
>
> Can somebody here recommend me some other 24b/96kHz card which is supported
> by ALSA
> I used to record streamed audio using this:
>
> arecord -D copy -f cd -t wav out.wav -d 10
Same here. My card has some bleed into the mic port so I do get something
regardless. I am able to set PCM record from aumix. No ideal how to do it in
alsamixer, it doesn't seem to be an option. I'v
> I have a new mother-card with a MCP61
> module and the Linux OS is Suse 10.2.
> Alsa driver is the 1.0.13.22 version.
I have that on one of my machines. It should use the snd-hda-intel module(s).
And you should run at least alsa version 1.0.14. There were some mixer
additions in 1.0.15 iirc
> I reconfigured the alsa-driver package like this
>
> ./configure --with-cards=hda-intel,usb-caiaq --with-sequencer=yes
> --with-moddir=/lib/modules/2.6.22-14-rt/updates/alsa
try adding a --prefix=/usr in there. Also since you're specifying a non
default modules location, you'll probably need
> > > record -D copy -f cd -t wav outfile.wav
> > >
> > > ecasound -i:/dev/dsp -o outfile.wav
I found a way around this. Sort of. Since the files do get cached by the web
browser, even if they don't finish downloading. I extracted their URLs from
about:cache for the disk cache.
about:cache
> > record -D copy -f cd -t wav outfile.wav
> >
> > ecasound -i:/dev/dsp -o outfile.wav
One question. Is there an alsa dummy driver/package that might capture this
through the above methods in it's intended form? I realize I wont hear
anything locally. But it'd be nice to capture it without
> Whether this works will depend on your hardware. Some devices support
> capturing the audio output, some don't.
Well, I swapped it around so that the onboard sound was card 0 and it works
that way. But the quality of what gets recorded is bad, actually hideous is
more appropriate. From stud
record -D copy -f cd -t wav outfile.wav
This does not seem to capture any of the sounds from /dev/dsp.
ecasound -i:/dev/dsp -o outfile.wav
Nor does this.
It's been a while since I've done this, what am I missing? Or is there
something about usb-audio the prevents this from working? Or some .
> Sadly, I've already tried all of these to no avail.
What recording application are you using? I've had issues where audacity would
give me that metalic sound and ardour+jackd would not. And vice versa.
Depending on versions and whatnot.
Beyond that I really can't offer any more insight w
> Each time you start to record you have a
> 10% chance of having the recording completely
> distorted and having a metallic sound.
I know that sound. And it is quite ugly. On my snd-hda-intel board(nVidia
MCP61), I have to increase the number of periods to overcome this sound.
default of 2 i
> /home/saurav # aplay -D plughw:0,0 Vincent.wav
>
> /home/saurav # speaker-test
>
> /home/saurav # arecord -D plughw:0,0 -t wav file.wav
have you tried to be a little more specific on your usage?
$ speaker-test -c 2 -D hw:0
$ aplay -D hw:0 -t wav -f S16_LE -r 48000 Vincent.wav
$ arecord -D h
> Anyway, I tried your suggestion and it
> recreated asound.state. Exactly the same
> as the previous one (diff gives no changes),
> and alsamixer gives the same error message.
> By the way, the USB sound card is disconnected,
> so that's not what's messing things up.
Some distros do the alsactl s
> Anyone else have any experience with this kind of thing?
IF /dev/dsp is already locked, aoss isn't going to help. Unless you redirect
it to another device not in use / locked. Also aoss does NOT cover any
children launched by the app started with aoss. i.e. Firefox, any popups are
not cove
> I'm a bit confused about the rest of your comment.
> The only file called devfs is a directory which has
> two subdirectories, neither of which seems to have
> anything interesting in it (one is empty). And I
> can't find a file called snddevices anywhere, but
> might have mistyped (I'm not curre
> Typing "alsamixer" in a terminal window gives:
> "alsamixer: function snd_ctl_open failed for default: No such device"
This error normally happens when it can't find the devices. Is your user in
the audio group? Are you running udev/devfs? If not, did you run ./snddevices
? Are the sound m
Is there any way to monitor arecord's progress?
In terms of time recorded, space used, and mixer levels? I've got two like USB
devices and can only record from both at the same time with arecord. I would
like to able to monitor what is recorded, so I can adjust the gain / mic level
if it's
I recently purchased a second M-Audio Mobile Pre. So I have two usb devices.
I wish to record from both interfaces at the same time. One has a phantom
powered LDC on it, the other a batterybox enabled electret mic. I'm not so
much worried about sync issues at this point. I'm mainly just try
> Can anyone recommend a PCCard/Cardbus soundcard, or possibly a
> USB card supported by alsa and which you've been able to run with
> low latency?
The USB bus speed probably isn't going to ensure low latency. Most USB
soundcards seem limited to two channels and 48kHz. I'd recommend a
PCCard/C
> Simple mixer control 'V_REFOUT',0
> Capabilities: enum
> Items: 'High-Z' '3.7 V' '2.25 V' '0 V'
> Item0: '0 V'
What type of mic are we talking? Electret type mics(the kind that most PC
style mics seem to be/use) need a voltage(called a bias voltage, or plug-in
power for voice recorders)
> When I do "arecord -d 10 -f S16_LE -r 48000 -c 2
> -t wav foobar.wav" and I literally put one of the
> speaker at full volume in one of my ears I can
> hear a very vere low volume version of myself.
That's always a good sign.
> _REFOUT=High-Z
High-Z is high impedence(Mega Ohms). Generally mea
> $ arecord -d 10 -f S16_LE -r 48000 -c 2 -t wav -D hw:0 foobar.wav
> Recording WAVE 'foobar.wav' : Signed 16 bit Little Endian, Rate 48000
> Hz, Stereo
> arecord: pcm_read:1347: read error: Input/output error
>
> ("-c 1" gives a different error).
The I/O error is an issue. What does dmesg or ot
> > > strace arecord -d 10 -f cd -t wav -D copy foobar.wav
> >
> > try:
> >
> > arecord -d 10 -f S16_LE -r 44100 -c 2 -t wav -D copy foobar.wav
> > or
> > arecord -d 10 -f cd -t wav -D hw:0 foobar.wav
>
> I tried the second and got:
>
> $ arecord -d 10 -f cd -t wav -D hw:0 foobar.wav
> Recording
> strace arecord -d 10 -f cd -t wav -D copy foobar.wav
try:
arecord -d 10 -f S16_LE -r 44100 -c 2 -t wav -D copy foobar.wav
or
arecord -d 10 -f cd -t wav -D hw:0 foobar.wav
Make sure copy is the right alias in your .asoundrc, otherwise you might need
to use the card number (/proc/asound/cards (
> $ echo "here kitty kitty kitty" | aoss festival
> Segmentation fault
$ echo "here kitty kitty kitty" | aoss festival --tts
(left off the --tts, doh)
-
This SF.net email is sponsored by: Microsoft
Defy all challenges. Micro
I can't seem to get festival to use any card other than the default card 0.
$ echo "here kitty kitty kitty" | aoss festival
Segmentation fault
Without aoss, it generates sounds to card 0. Even with the following .asoundrc.
#--- START .asoundrc ---#
pcm.atiixp {
type hw
card 0
}
ctl.atiixp
> > I have:
> > Guitar -> Behringer Xenyx mixer -> M-Audio Audiophile 2496 RCA inputs
> >
> > When the setup is sitting there, idle and active, there is a very very
> > high-pitched light squeal coming from the PC speakers. It's so
> > high-pitched that only myself and so far two other people I've
> The Mia now appears to work every time I boot up,
> but I only managed to do this by disabling the
> built in Intel sound card in the BIOS. Not an
> ideal solution.
Sound like you might just need to index the alsa modules so they load in a
specific order. Automations scripts might be working i
> Not really insight suggestions: But if you can negotiate with PCI try the
> M-Audio delta series. I've got a delta 1010lt and it works great. There are
> smaller models, the delta44 and delta66.
> They have XLR, chinch and digital I/O and MIDI in/out.
> My card cost about 300 EUR (about 6
> Nothing that could construed as phantom voltage
> (which is normally 50 volts in the broadcast industry)
> is visible for either the intel_8xx on the motherboard,
> or for the audigy 2 value I use for everything. I also
> have an sb16 but don't recall it as having such a feature
> either.
Phant
> High-Z
> 3.7 V
> 2.25 V
> 0 V
> I have no idea what does the first one means. Does it provides any voltage?
Not sure about the voltage. But things like electric guitar need high-Z (high
impedence). In the Mega Ohms. As opposed to your standard microphone preamp
that only has a few Kilo Ohm
> Problem is that I'm able to use an old (crappy) labtec microphone, but the
> two different headset microphones I tried, don't pick up a thing.
> I've tried the headsets on another computer so I know the headset mics are
> working.
You're probably missing the DC bias aka bias voltage aka plug-in
> I just got a new HP laptop. Nvidia based. It has MCP67 components from
> lspci.
>
> I installed alsa 1.0.15 and it does not find any sound cards.
# modprobe snd-hda-intel
At least that's the one for my MCP61. Does the card show up in lspci?
--
> Apps that use the old OSS API block the soundcard if your sound device
> lacks hardware mixing.
>
> Upgrade to the latest Flash plugin which uses ALSA and software mixing
> will work.
Also try starting firefox with aoss. Note that aoss will not be wrapping any
child instances (pop-ups) launch
> Here's what I have not been able to get to work:
> Full duplex,
Not sure about the full duplex part, as not all cards are full duplex. My
M-Audio Delta 44 says Full Duplex on the box. And seems to be just that. My
laptop(ATI IXP) on the other hand can do full duplex, BUT... there's some ble
> I don't see it in the aplay -l, but lsusb confirms its existence.
Does it show up in /proc/asound/cards ?
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> I somehow messed up my audio settings with
> amixer, at least that's what I believe. Not
> I have the problem that my voice is really
> quiet when I talk into the mic, but that
> there is *a lot* of white noise!
I get a lot of white noise, if I use the mic boost setting. If you have a
powered
> On my Dell Inspiron 1720 (Sigmatel based
> intel hda) I cannot enable audio passthrough
> as I record. On playback the recordings have
> worked fine, but its kinda tricky to punch-in
> a bass guitar track in Ardour if I cannot hear
> the bass while I'm playing it! Even jack cannot
> patch the cap
I'm wondering on a couple things with asoundrc. If I want the default alsa
device to be something other than card 0, is there a simple entry I can put in
asoundrc to make that happen? I can get the cards to switch in the driver
configuration in /etc/modprobe.d/ but I'd rather have something us
> even with Mic Boost (+20db) on, the sound is just not
> loud enough.
Mic in is a combination of mixer settings. Make sure you have mic +
gain/capture + mic boost to have maximum input levels. It may also be that
your mic requires a bias voltage(+5V) and you soundcard doesn't supply one.
HTH
> But it seems MIDI is not supported:
Many cards don't have onboard midi support these days. As long as you compiled
with --with-sequencer=yes, you can run timidity to emulate midi.
# modprobe snd-seq-oss
$ timidity -Oi -iA
Assuming timidity is installed and configured with useable sound patch
> [EMAIL PROTECTED] ~]$ cat /proc/asound/cards
> 0 [Intel ]: HDA-Intel - HDA Intel
>HDA Intel at 0xd044 irq 22
Have you tried:
$ alsamixer -c 0
You may need to press m to mute/unmute and cursor the selection/levels. Plus
tab to get to the capture/all/pla
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