Dave Pawson wrote:
2009/7/24 Clemens Ladisch cladi...@googlemail.com:
Does the onboard HDA controller show up in the output of lspci?
# lspci | grep Audio
00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio
Controller
01:00.1 Audio device: ATI Technologies Inc R700
Dave Pawson wrote:
2009/7/24 Clemens Ladisch cladi...@googlemail.com:
Dave Pawson wrote:
# lspci | grep Audio
00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio
Controller
01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD
4000 Series]
Both
Dave Pawson wrote:
2009/7/24 Clemens Ladisch cladi...@googlemail.com:
Is there any error message in the system log when you do this:
rmmod snd-hda-intel
$ rmmod snd-hda-intel
ERROR: Module snd_hda_intel is in use
Please kill all applications that use it, then try again.
(I guess
Dennis Borgmann wrote:
I am working with snd_pcm_mmap_begin and snd_pcm_mmap_commit - just the
way it is done in the pcm.c under the test-directory of the
alsa-lib. While working with snd_pcm_writei(), those problems did not
occur, but I want to test this using the mmap technology.
This
Christoph Köditz wrote:
is the driver in work or do you need some more informations about this
soundcard?
According to the driver's source, this card has been supported since 2005.
HTH
Clemens
--
Enter the
Marek Michalak wrote:
Is there any difference if I write in my code:
while (size 0) {
frames = size;
err = snd_pcm_mmap_begin(handle, my_areas, offset, frames);
record_buffor(my_areas, offset, frames, data-phase);
Matt Snow wrote:
On Fri, Jul 10, 2009 at 2:20 AM, Clemens Ladischcladi...@googlemail.com
wrote:
So the headphones or speakers in the front jack work, and the to-RCA
cable in the *same* jack does not? This is obviously a broken cable
(or TV).
You might think that, but its fine. I take
Matt Snow wrote:
On Thu, Jul 9, 2009 at 12:05 AM, Clemens Ladischcladi...@googlemail.com
wrote:
Matt Snow wrote:
After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out
plays audio perfectly through head phones or powered speakers, but the
3.5mm stereo-to-RCA cable going to
Matt Snow wrote:
I have a Asus Xonar DX sound card that was working perfectly in
KnoppMyth/LinHES R5.5(Knoppix/debian based) after manually compiling
alsa-driver 1.0.18b and 1.0.18 libs/utils/etc.
After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out
plays audio perfectly through
Saleem Hasan wrote:
I found that although aplay was working on my computer, arecord was not. I
played back the wav files provided with the linux distribution for various
system beeps and sounds. I am able to navigate through the alsamixer
screen using tab, space, etc in order to select the
Phil wrote:
I'm trying to find a way to get audio in over USB for a project, and I
think that the UA-101 might be about the only Linux device that supports
6 or more channels of audio over USB.
Indeed. There are other multichannel USB devices, but there are none
that have even rudimentary
scar wrote:
Clemens Ladisch @ 06/26/2009 03:30 AM:
The hardware does not have any mechanism to change the line input's
level. You have to adjust the volume after recording.
shown in section 6.3.2 of the xonar dx manual is a picture of the levels
for recording, indicating
scar wrote:
i have been experimenting, and what appears to be happening is this:
while recording using the line, if the microphone levels are adjusted,
the driver/card automatically jumps to recording from the mic.
Neither the driver nor the card do this; I'd guess this is a feature of
your
scar wrote:
Clemens Ladisch @ 06/24/2009 12:53 AM:
The microphone input is mono. There are no left/right levels; the
driver pretending that they exist is a driver bug. I'll fix that.
but that port is also the line-in, so i need stereo
When used as line in, that port is stereo
scar wrote:
so, i got my asus xonar dx ...
the problem i am running into is i cannot independently adjust the left
and right levels of the Mic capture. well, i can, kind of, but it
doesn't have any effect.
The microphone input is mono. There are no left/right levels; the
driver pretending
Malte Gell wrote:
Clemens Ladisch cladi...@googlemail.com wrote
Malte Gell wrote:
Do you think I could get rid of this Skype issue when I upgrade to a
CMedia 8768 chip?
That's exactly the the same hardware interface (and driver), so I'd
guess you'd get the same error.
Ok, same
Malte Gell wrote:
when I use Skype 2.0.0.72 on Linux I get an error:
RtApiAlsa: callback thread error (RtApiAlsa: audio write error for device (C-
Media CMI8738 (hw:CMI8738,0)): unknown error 405.) ... closing stream.
Unknown error isn't quite helpful.
I did not have problems before when I
Grant wrote:
Are you sure the hardware actually supports S24_3LE? Most 24 bit
soundcards require 24 bit audio to be packed into 32 bit words. See
if you can output S24_3LE directly to the hw device.
I don't think a USB DAC could do 24-in-32 because of the USB 24-bit
limitation.
Mike Knichel wrote:
I have recently set up a computer for my daughter from old parts. It
uses the VIA 8237 chip. I can get no sound from speakers.
I even ran the alsa script and the results are here...
Your mixer settings seem to be correct.
However, some mainboards use another than the
Mike Knichel wrote:
I do not have front mounted speaker jacks. Only the ones in the back.
I meant front as opposed to surround or center/LFE.
If you have only one jack intended for speakers, then your description
looks as if that jack is connected to the surround output of the
codec. Please
Raena wrote:
Clemens Ladisch wrote:
Raena wrote:
checking for ncursesw5-config... yes
This indicates that ncursesw is installed ...
checking for new_panel in -lpanelw... no
configure: error: panelw library not found
... but actually it isn't.
What is the output of the following
scar wrote:
i was basically just looking for verification that the m-audio 2496 card
would suit my needs.
Yes, it certainly would.
gary wrote:
I often wonder has VIA came out of nowhere with a DSP audio chip
(Envy24). Oversampled ADCs are not exactly rocket science, but to make
one
Raena Lea-Shannon wrote:
Is there another ncurses-devel I have missed?
libncursesw5-dev?
HTH
Clemens
--
OpenSolaris 2009.06 is a cutting edge operating system for enterprises
looking to deploy the next generation of
robert crowther wrote:
Can ymfpci access the WF-192XG wavetable?
No. While documentation for basic DS-XG wavetable functionality exists,
nobody has as yet added support for this to the driver.
Best regards,
Clemens
Aidan Dixon wrote:
I have a MOTU micro lite (USB, 5x MIDI IN and 5x MIDI OUT, Vendor ID
0x07fd, product ID 0x0001). According to the kernel usb-audio device
driver source code, this device should be supported as its USB vendor
and product ID are listed in a table in linux/sound/usb/usbmidi.c.
Michael B Allen wrote:
Does capture work with the M-Audio Fast Track USB?
Maybe. Does some capture device show up in the output of arecord -l?
Can someone give me some advice or a pointer to some documentation
that explains how the whole ALSA tool chain works wrt external USB
sound
Raena wrote:
I also installed the snapshot alsa-lib and that was fine but alsa-utils
would not configure. configure: error: panelw library not found I have
not been able to find out what package has this library. Here is my
configure output;
...
checking for ncursesw5-config... yes
This
Ari Moisio wrote:
Thanks:-) I downloaded the relevant firmware package from alsa-project
site, configured, compled and installed. Everthing is working again:)
I only wonder if the driver itself loaded the firmware earlier because
there were no such issues with 2.4 kernels?
In older
Paul Hartman wrote:
On Fri, Apr 17, 2009 at 2:27 AM, Clemens Ladisch
cladi...@googlemail.com wrote:
Phil Gorbett wrote:
I am having difficulty getting the microphone(s) going with this
monitor, and get the get the following results when I disable pulseaudio
and run arecord
Michał Kowalczyk wrote:
SND_PCM_FORMAT_U8, //changing this type for any other gives an error in
the output: segmentation fault.
U8 has one byte per sample; if you try to use a bigger sample format,
you have to enlarge the buffer accordingly.
1, //argument responsible for
MK wrote:
I want to write a simple ALSA application. I have downloaded pcm_min.c
from ALSA documentation and try to do some modifications in code like
change channels to stereo or sampling rate to 44100Hz. But every time
program fails. Can anybody explain me why this is happening?
Please show
immanuel litzroth wrote:
Apr 13 12:08:39 voodochile kernel: [ 169.622067] usb 2-10: new full speed
USB device using ohci_hcd and address 5
Apr 13 12:08:39 voodochile kernel: [ 169.844566] usb 2-10: configuration #1
chosen from 1 choice
Apr 13 12:08:39 voodochile kernel: [ 169.847594]
Phil Gorbett wrote:
I am having difficulty getting the microphone(s) going with this
monitor, and get the get the following results when I disable pulseaudio
and run arecord:
arecord -v -f cd -D plughw:0 file.wav
Recording WAVE 'file.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Matt Garman wrote:
Is there a way to query alsa to see what sample rates and formats
the sound hardware natively supports?
Try the attached program.
HTH
Clemens
/*
* hw_params.c - print hardware capabilities
*
* compile with: gcc -o hw_params hw_params.c -lasound
*/
#include stdio.h
Geoffrey Leach wrote:
I'm havind a problem with alsamixer. It appears not to save the
changes. The man page does not mention this, so perhaps its doing what
it is designed to do.
It is only designed to change the sound card's mixer settings.
In that case, can anyone point me to a tool that
Jonathan Black wrote:
I've been struggling as of lately to get ANY sound through the lime
green output on the back of my motherboard.
lsmod | grep snd shows:
snd_mpu401 6252 0
snd_cs4232 11380 0
snd_via82xx20920 2
Please show the output of
Geiger Ho wrote:
I would like to build the WM8510 SOC sound driver for my ARM platform.
After downloading alsa-driver-1.0.19, ./configure
--with-cross=arm-linux- --with-kernel=/home/source/linux-2.6.22.19
--with-build=/home/source/linux-2.6.22.19 --with-cards=all, and then
make. I get
Oliver Stephenson wrote:
Now though, I'm trying to open the PCM output in 6-channel mode
non-interleaved, and it all falls over here:
if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params,
SND_PCM_ACCESS_RW_NONINTERLEAVED)) 0)
{
fprintf(stderr, cannot set access type
Poulet Fou wrote:
I have two sound cards, one attached to pc speakers and one attached to a
sound system in another room.
Following instructions on the alsa project wiki, I am trying to use two
sound cards (V8237 and CA0106).
I configure my ~/.asoundc but I still can't manage to send a sound
Jens Rutschmann wrote:
I have a C-Media USB Headphone Set which is working well. One issue though is
that when plugging it in the mixer levels are initialized to bad values. The
Speaker level is set to 100% (*very* loud, like in ouch !!!) while the
Mic
Capture level is set to 0%.
Matias D'Ambrosio wrote:
I have been using this system for months, no problem (and many distros
before), but a few days ago sound stopped working, probably after an update.
I have an SB Live, I'm running Debian testing AMD64 kernel 2.6.26-1-amd64
#1
SMP.
lspci shows the card:
Antony Gelberg wrote:
I have an onboard NVidia sound card using snd_hda_intel. Full info about my
setup is at
http://www.alsa-project.org/db/?f=f407c7aceb5abe36b68731ff50fd1d21f272fb98
I can output everything to a HT amp via the IEC958 optical out by means of the
.asoundrc that you see in
(please don't top-post)
sto...@safe-mail.net wrote:
Any suggestions with this output?
The AC'97 controller does not show up. Please make sure that SB
emulation is disabled.
Best regards,
Clemens
--
Apps built with
Ron Pitts wrote:
Analogue Output is working fine now however no luck with the HDMI audio.
But HDMI video does work?
Does your HDMI display show if any audio is sent, or in which format?
What sample format did you try to play?
I've rebuilt the alsa lib with full debugging turned on:
No sound
Landis McGauhey wrote:
I've decided to replace the old ISA card with a PCI card (several PCI slots
are available). Can anyone happen to tell me if you have any ALSA
experience with the Diamond Xtreme XS51?
According to the driver's .inf file, that card uses a CMI8738/8768
chipset, which is
Ron Pitts wrote:
[ 653.504508] snd_mpu401_uart: Unknown symbol snd_rawmidi_receive
...
The sound modules cannot be loaded because other sound modules they
depend on cannot be loaded.
Please show the first few of these messages (for module snd).
Best regards,
Clemens
Ron Pitts wrote:
Its loading now without error however doesnt report its found my ASUS
HDAV1.3 ..
It doesn't report anything when there is no error.
Does it appear in /proc/asound/cards?
Anyway, I've fixed a bug after 1.0.19 was released; please try
Landis McGauhey wrote:
ALSACONF recognizes the soundcard as SBAWE Creative SB AWE64 PnP and
completes its work without errors. ... Command ALSAMIXER results in
the error message function snd_ctl_open failed for default: no such
device.
In theory, all those old ISA cards are still supported.
Ron Pitts wrote:
...
FATAL: Error inserting snd_virtuoso
(/lib/modules/2.6.27-11-generic/kernel/sound/pci/oxygen/snd-virtuoso.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
The most likely cause of this problem is that you are trying to load a
module of a new driver version
dennism...@gmx.net wrote:
Can please a admin have a look at my mail?
There is no moderator; all such mails go into a black hole.
the message was over the max size
Compress it, ot put it somewhere on the web.
HTH
Clemens
Mårten Gustafsson wrote:
mor...@linux-hyfd:~ arecord -D spdif:0 -f dat -t raw | ac3dec -D hw:0 -6
Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz,
Stereo
2.0 Mode 48.0 KHz 32 kbps Complete Main Audio Service
unsupported 1/1 channels 6
Apparently, the recorded Dolby
sto...@safe-mail.net wrote:
The card is enabled in the BIOS,
Is the entry Enabled or Auto? The latter means the BIOS
automatically disables it when it finds another sound card, because,
apparently, it doesn't expect anybody would want to use it when there
is a _real_ sound card.
This is common
rek...@holisticode.se wrote:
I have been searching the web for an example of how to get sourround
sound from my toslink input [...] I realise that an a52 filter should
be used,
The a52 plugin encodes PCM data into Dolby data, and as a plugin, it is
only used when an application uses the
jed wrote:
Damn, that's a substantial amt of stuff that won't be available don't u
reckon?
Yes.
Best regards,
Clemens
--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the
deadrabbit wrote:
I'm new to using ALSA, or at least to the internals of it. I'm trying
to create a Icecast stream encoding computer for two audio streams.
The computer as a single stereo line in, so I'm planning on feeding
each mono stream through the single stereo input. I've looked at the
sonof...@iinet.net.au wrote:
I'm looking to move over to a device that I can use with both my linux desktop
and a 2nd hand powerbook i bought recently. I have been more focussed on FFADO
compatible as from what I understand I am more likely to get latency issues
on a
USB device...
gary wrote:
gary wrote:
Now for the problem. I can't talk to the iMic. Using
krecord -d /dev/dsp4 .
I get the response
ioctl SNDCTL_DSP_SETMFT: Input/output error
There should be an error message in the system log (probably
/var/log/messages, or at the end of the output of dmesg).
I
jed wrote:
Is support for this coming soon?
Yes.
How does using ALSA compare to using drivers developed by the vender
specifically for Windows?
i.e.
Will I not be getting most out of this card (hardware-wise) if I'm not
using the drivers Asus developed, or is the difference negligible?
jed wrote:
Is support for this coming soon?
Yes.
Awesome, is it likely to be within the next 6-months?
Actually, it looks like this week ...
Please note that the Linux driver does not have the driver features,
which are implemented in software (Dolby, SVN, Xear, etc.).
Can you
Floris wrote:
Is there a way to encode a file on the fly?
http://wiki.alsa-project.org/main/index.php/A52_plugin
HTH
Clemens
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James wrote:
$ speaker-test -D iec958 -c 6
...
Channels count (6) not available for playbacks: Invalid argument
S/PDIF does not support uncompressed surround sound; you have to play
stereo data (use -c 2) or AC-3/DTS-compressed data.
Best regards,
Clemens
Evan Leibovitch wrote:
# arecord -D hw:1,0 /dev/null
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
arecord: set_params:932: Broken configuration for this PCM: no configurations
available
Try arecord -D default:1.
HTH
Clemens
davide.bonfa...@bticino.it wrote:
I'm searching for a textual tool in order to play, pause and resume the
streaming on a certain device.
Try mplayer.
HTH
Clemens
--
This SF.net email is sponsored by:
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Ari Moisio wrote:
# Usb-headset
options snd-usb-audio index=2 vid=0x046d,0x0a02
# M-audio fasttrack
options snd-usb-audio index=3 vid=0x0763,0x2012
This does not work because the second 'options' line overrides the
first, and the product ID must be put into the 'pid' parameter.
Use this:
Bill Unruh wrote:
On Mon, 15 Dec 2008, Paulo Moura Guedes wrote:
[...]
The ASRC, as the name implies, is not syncronized to the clock of
the incoming digital signal. Therefore, its performance is independant of
the
This makes no sense at all. If the incoming signal is a digital signal,
Paulo Moura Guedes wrote:
On Thursday 11 December 2008 07:35:25 Clemens Ladisch wrote:
The default device (named default) uses automatic resampling, but the
spdif device does not.
So, I have to manually set the sample-rate, depending on the files I will
play?
No, ALSA automatically
Laurent E wrote:
- According to the command cat /proc/asound/card0/codec#0, the chip
is a SigmaTel ID 7645. [...]
1/ Is it possible that the chip is incorrectly detected?
Your driver doesn't know about your chip.
Support for this chip was added in May:
Paulo Moura Guedes wrote:
- does Linux/ALSA features dynamic sample rates?
The default device (named default) uses automatic resampling, but the
spdif device does not.
- is it possible to set the bit-depth? (in my case to 24 bit)
Yes, if the hardware supports it.
- what other variables do i
Lindsay Roberts wrote:
I have a board
Which one?
This is an Intel 945GC chipset (ICH7) with a Realtek ALC662 on it,
both of which seem to be supported in alsa.
Leaving the dist to its own devices sees it load snd_intel8x0, and
error out as follows:
codec_ready: codec is not
Yan Seiner wrote:
I've added a line to /etc/bash.bashrc that sets ALSA_CARD depending on
the value of $DISPLAY.
Now this works fine if I open a terminal window and run my apps. But
apps started from the menu don't have ALSA_CARD set.
You could set ALSA_CARD when X is started (in .xinitrc or
[EMAIL PROTECTED] wrote:
But when I try to replay a DVD with AC-3 data xine only tells me that
the audio device is not available.
What happens when you try a command like this:
aplay -D spdif something.wav
Best regards,
Clemens
Bankim Bhavsar wrote:
For a stereo DAC channel with 16-bit samples, number of bytes per
frame = 4 bytes, correct?
Yes.
HTH
Clemens
-
This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the
Vedran Miletić wrote:
http://www.via.com.tw/en/products/audio/controllers/envy24/
Claims that it has 20-channel, 26-bit wide built in mixer. What does
that mean?
The chip plays 10 channels in parallel (8 analog, 2 digital) and records
12 channels in parallel (8 analog, 2 digital, 2
Vedran Miletić wrote:
Drivers that support hardware mixing:
* snd_emu10k1 - ...
* snd_cs46xx - ...
* snd_au88x0 - ...
* snd_ymfpci - for YMF7xx chips. Like most of the others, has been
discontinued and must be bought used. Used on several cards, e.g.,
Hoontech SoundTrack Digital XG
Julien Claassen wrote:
If I'd like to use the PCM slave timer, how do I access that? Or were the
instructions you gave me at the end of your mail already for that prupose?
Yes. For the values of the CARE/DEV/SUBDEV parameters, see
/proc/asound/timers.
HTH
Clemens
Julien Claassen wrote:
Does the ALSA timter interface take its timing info from the soundcard
(sample clock) or from the system's timer (RTC)?
ALSA's timer interface can use
1) the system timer (which is not the RTC timer),
2) the RTC timer (if it is available),
3) a sound card timer, if the
Florian Winter wrote:
I can't derive from the documentation when exactly my poll() call will
wake up.
It can wake up whenever the device generates an interrupt, i.e., at
period boundaries.
Is there a way to tell ALSA something like: I want to write at
least N frames to the device without
Florian Winter wrote:
Suppose, an ALSA playback device is opened in blocking mode, and one
thread calls snd_pcm_writei. If the snd_pcm_writei call blocks, because
the internal buffer of the ALSA device is full, is there a way by which
another thread can interrupt the call, so it returns
Robert Vincent Krakora wrote:
em28xx_alsa: disagrees about version of symbol snd_pcm_new
This means that the em28xx_alsa module that you're trying to load and
the currently loaded snd_pcm module have been compiled for different
kernels.
It is possible that the old ALSA modules are still loaded;
Jason Tackaberry wrote:
I am testing with mplayer, and ensuring the digital device (as shown by
aplay -l) is being used. In my case it's device 1, and I am specifying
-ao alsa:device=hw=0.1 on the mplayer command line.
Try spdif instead of hw:0,1.
HTH
Clemens
Eliot Blennerhassett wrote:
can anyone tell me how to find out which processes are using an ALSA
driver and preventing it from being unloaded.
I know that lsof /dev/snd will tell me who has these files open, but
often this shows nothing and still the driver can't be unloaded.
Where else
Jason Gauthier wrote:
I can't find many cards except for the Creative Labs X-Fi that seems to
fit both.. and from what I read SPDIF passthrough doesn't work.I'm
not 100% sure what that means. (Pretty new to this level of audio)
SPDIF was designed to transport two channels of 16-bit
V Gabriele De Palo wrote:
# cat /proc/asound/cards
...
1 [Ultra ]: USB-Audio - Fast Track Ultra
M-Audio Fast Track Ultra at usb-:00:1d.7-3, high
speed
It seems the device is more or less class compliant, so, in theory, it
could work.
# speaker-test -c 2
Mark A Jenks wrote:
I have a Bluegears b-Enspirer 7.1 Sound Card CMI8788 that I am looking
for some help/advice for.
I would like to seperate out the stereo outs to multiple outputs, so I
can use it as a backend for my house with zones.
The hardware cannot control the four analog outputs
Artem Makhutov wrote:
I have a Philips SPC520NC webcam, which works fine with the linux UVC driver.
The only problem I have is to record audio using the build in
microphone. (Under Windows the microphone works well.)
After arround 5 seconds I am getting an read error: I/O Error message,
Stevens, Peter wrote:
For alsa-lib-1.0.17rc2 the dmix plug-in is enabled by default (for the
default stereo device). Would anybody know which setting I must
override in /etc/asound.conf file to enable dmix for the surround51
device (among others)? I wish to be able to open any of the : stereo,
Florian Winter wrote:
- What is the dmix plugin and what are the benefits of using it?
- Is it possible to disable the dmix plugin?
- What consequences does disabling the dmix plugin have? What essential
features of ALSA will be missing without it?
The dmix plugin allows multiple
stan wrote:
Florian Faber wrote:
You want hardware monitoring - there are sound cards that support
hardware mixing. With good converters you have latencies down to 5
samples at 192kHz, that would be 0.026ms for each way, 0.052ms over
all.
I'm not the original poster, but I'm curious
Alexander Carôt wrote:
3.) Rather than using a double buffer for the playout wouldn't it be
possible to choose only one physical playout buffer and parse the
captured data in right at the interrupt.
It's unlikely that any code could be fast enough to write the entire
buffer before the hardware
Florian Winter wrote:
Is there another way to determine whether a certain hardware supports
snd_pcm_pause without having to test the hardware?
$ grep -rl SNDRV_PCM_INFO_PAUSE sound
sound/arm/pxa2xx-pcm.c
sound/arm/sa11xx-uda1341.c
sound/core/pcm_native.c
sound/drivers/vx/vx_pcm.c
Rangel Reale wrote:
modprobe snd-serialmidi sdev=/dev/ttyUSB0
Hmm the serialmidi driver seems to be broken with the kernel Ubuntu 8.04
uses (2.6.24-17-generic) (I had to enable CONFIG_BROKEN on configure to
be allowed to compile it)
/usr/src/alsa-driver-1.0.16/drivers/serialmidi.c:117:
Germano Carella wrote:
I have an edirol UA-101 usb sound card.
The UA-101 is not fully supported; capturing may work, but the driver
does not correctly synchronize to the device's sample frequency when
playing, and there are no mixer controls.
Regards,
Clemens
Karl Schmidt wrote:
The linux kernel is no longer accepting closed firmware blobs
The kernel does contain sourceless firmware blobs.
Some distributions (Debian and derivatives) remove such drivers from
the kernel.
A list of audio chip sets that require firmware that is not GPL
friendly
Mr. Man wrote:
Dmesg says:
Maestro3: probe of :02:03.0 failed with error -2
-2 means no such file or directory.
I'd guess that it did not find the firmware files. Either enable
CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL when compiling the kernel, or
install the alsa-firmware package.
HTH
Peter A. Friend wrote:
I have a bGears b-Enspirer card supported by the snd-oxygen driver of
the latest ALSA release. Things work fine for a short time then the
machine hangs and I am forced to reboot.
...
Usually the system hangs when adjusting the volume or skipping tracks
in Rhythmbox or
Jerry Geis wrote:
When compiling Mplayer rc1 I am getting a link error about undefined
reference to snd_config_search_alias_hooks()
That code was changed over a year ago. Use rc2.
HTH
Clemens
-
Check out the new
Stephen Stocker wrote:
With a fresh checkout of the hg repository (alsa-driver and alsa-kernel),
I'm getting a compile error.
../alsa-kernel/core/vmaster.c:258: error: parse error before
this_object_must_be_defined_as_export_objs_in_the_Makefile
Now fixed:
I wrote:
However, an even better solution for your problem would be to enable the
input monitoring function of your sound card.
Okay, monitoring is now supported, with this patch:
http://hg.alsa-project.org/alsa-kernel/rev/0fff88f7f05b
To enable monitoring from a script, run
amixer
Helge Fredriksen wrote:
I just bought a couple of MAudio cards to test out (Revolution 5.1 and
Audiophile 2496). I'm trying to test the sound using arecord and aplay, but
no matter which format I choose I get the message:
aplay: set_params:900: Sample format non available
Use aplay -v
John Sigler wrote:
format : S32_LE
msbits : 24
...
The first sample is stored as 00 00 a5 f5.
Does this represent the value 0xf5a5?
Yes.
For the majority of samples, the first two bytes are 00 00.
Is this because the input stream was, in fact, a 16-bit stream?
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