But what I am really wondering is how to set buffers in alsa config and
how to have access to the "disconnect" option and leave the unit in
standalone mode. Any ideas?
Le 23.05.24 à 02:04, David Kessler a écrit :
Hey there!
Ok so interface is composed of 52pi PCIe slot and a P
lsa-project.org/wiki/Matrix:Module-hdsp
https://www.kernelconfig.io/config_snd_hdsp
Le 20.05.24 à 14:54, Peter P. a écrit :
* David Kessler [2024-05-20 12:56]:
Ok everything works well after adding the right firmware to
/lib/firmware/hdsploader! I'am very happy and thank's to the folks pointing
me in
Ok everything works well after adding the right firmware to
/lib/firmware/hdsploader! I'am very happy and thank's to the folks
pointing me in the right direction!
Cheers!
Le 20.05.24 à 10:39, David Kessler a écrit :
Adding dtoverlay=pcie-32bit-dma too /boot/firmware/config.txt is
Adding dtoverlay=pcie-32bit-dma too /boot/firmware/config.txt is helping
a lot! The card is recognized in alsa! But yet no controls but says it
wants to update firmware! I can feel I am close!
Le 20.05.24 à 10:11, David Kessler a écrit :
As of
https://git.kernel.org/pub/scm/linux/kernel/git
uot;, hdsp->card_name);
return -ENOMEM;
}|
Who is responsible for this? We must have a talk! (:
Le 20.05.24 à 09:07, David Kessler a écrit :
From what I can tell comparing to a working system is that I have all
needed kernel modules loaded. the oss are indeed no longer need
From what I can tell comparing to a working system is that I have all
needed kernel modules loaded. the oss are indeed no longer needed.
Thank's for pointing me out.
Here a paste bin with the info I could gather.
https://pastebin.com/dAfDDnkM
The missing interrupt I wonder about and still the
à 19:40, David Kessler a écrit :
I am now running a kernel with PCI soundcard hammerfall hdsp enabled
and snd-hdsp module loaded.
But it's not happy because it has no available buffer...
dsp@dsp:~ $sudo dmesg | grep snd
[ 0.00] Kernel command line: reboot=w coherent_pool=1M
does my rusty kernel compilling skills.
Thank's for reaching me out to any clue what is going on here <3
Le 19.05.24 à 11:55, David Kessler a écrit :
Ok thanks I've came across this webpage also. I'am missing the
alsa-driver tarball to do make configure as explained here:
https:/
save space?
Le 19.05.24 à 11:44, David Kessler a écrit :
I am now running ubuntu server 24.0 lts
jean@dsp:~$ modinfo soundcore
name: soundcore
filename: (builtin)
alias: char-major-14-*
license: GPL
file: sound/soundcore
author: Alan Cox
d
Looking for HDSP + Multiface or Digiface cards :
Card 0 : vc4-hdmi-0
Card 1 : vc4-hdmi-1
Looks like a trivial problem. I would be very happy if someone could
learn me the next steps involved to make this card work on a raspberrypi 5.
Much thanks!
Le 19.05.24 à 10:16, David Kessler a écrit :
I am
I am trying to bring to life a hammerfall soundcard on a RPI5. I'ts
attached through a PCI interface to PCIe 1x interface to the PI. THe
card is powered by eternal supply and is recognized with lspci like
everything is normal but snd-hdsp module is missing on that kernel.
I have alsa-firmware-
ed, with nothing about it in the
kernel log.
So, what do you think? Is there any way to restore my sound card to
health, or am I forever stuck with Bluetooth speakers? Is there any way
to reset the audio controllers?
Any advice will be deeply appreciated.
Thanks,
-=[david]=-
, I was able to expose all 4
of its inputs and all 4 outputs in JACK via the zita bridge tools.
Cheers
David
On Sun, 8 Oct 2023 at 19:05, David McNab wrote:
> Hi all,
>
> I'm hoping for some ideas to get a maya44-based audio device working in
> ALSA on a Ubuntu 20.04LTS x86-64 a
Hi all,
I'm hoping for some ideas to get a maya44-based audio device working in
ALSA on a Ubuntu 20.04LTS x86-64 audio workstation.
The device shows up in 'arecord -l' and 'aplay -l'. But when I try to
record from it with arecord, I get:
*arecord -D hw:5 -t wav -c 2 -f S16_LE -r 44100 /tmp/foo.
point me to steps to work these out?
Best regards,
--
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evices shown by 'aplay -l' are
listed except for the first - my headphones.
I have run the latest version of alsa-info.sh under both versions of
the OS, and I attach the outputs. I am new to the list, so I'm not
sure whether attachments are accepted, but I can paste
CPU1 CPU2 CPU3 CPU4
20: 0 0 06519080 0 IO-APIC
20-fasteoi snd_emu10k1
I'm in the process of replacing my CPU, RAM, and MB, because, as you
might have noticed above, one of the cores is dead. I'll tell you if it
works with my new motherboard and CPU.
Also, it would probably be helpful for the devs to know:
1: Is yours a PCI or PCIe card? (Mine is PCI)
2: If a PCI card, what type of adapter are you using?
Hope this helps,
David
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snd_seq,snd_hrtimer,snd_pcm
snd_rawmidi28672 2 snd_seq_midi,snd_usbmidi_lib
snd53248 8
snd_hwdep,snd_seq,snd_usb_audio,snd_timer,snd_rawmidi,snd_usbmidi_lib,snd_seq_device,snd_pcm
Thanks in advance,
David
___
Alsa-user
Not to be a pest, but does anyone know or know where I should be asking?
Thanks,
David
On Thu, 26 Dec 2019 21:55:15 -0500
David Niklas wrote:
> I can find mentions of the CMI/CM in the Linux kernel sources.
> I also found this post from 2014:
> https://sourceforge.net/p/als
I can find mentions of the CMI/CM in the Linux kernel sources.
I also found this post from 2014:
https://sourceforge.net/p/alsa/mailman/alsa-user/thread/536B28B2.1010107%40googlemail.com/
Although it only mentions the CM specifically in the patches which
are linked to.
In a 2019 post he
On Sunday 2 September 2018 20:20,
Caelia Chapin put forth the proposition:
> Hello, all --
>
> I am attempting to set up an audio stream on my home network for music
> listening. I have a desktop PC and a little headless server, both running
> Arch Linux. The server is attached to speakers and con
On Sunday 24 June 2018 13:07,
info put forth the proposition:
> I wish all my users have been migrated. Until then, is Fedora distro builder
> misusing ALSA? What do I need to tell them to either enable detection of
> attached front panel cables, or storing and properly restoring the volume and
>
On Sunday 24 June 2018 11:29,
info put forth the proposition:
> On 2018-06-24 11:24 AM, David Woodfall wrote:
> > On Sunday 24 June 2018 11:09,
> > info put forth the proposition:
> > > On 2018-06-24 11:06 AM, info wrote:
> > > > On 2018-06-24 10:51 AM, David
On Sunday 24 June 2018 16:24,
Dave Woodfall put forth the proposition:
> On Sunday 24 June 2018 11:09,
> info put forth the proposition:
> > On 2018-06-24 11:06 AM, info wrote:
> > >
> > > On 2018-06-24 10:51 AM, David Wood fall wrote:
> > > > On Su
On Sunday 24 June 2018 11:09,
info put forth the proposition:
> On 2018-06-24 11:06 AM, info wrote:
> >
> > On 2018-06-24 10:51 AM, David Wood fall wrote:
> > > On Sunday 24 June 2018 08:58,
> > > info put forth the proposition:
> > > > Good morni
On Sunday 24 June 2018 11:06,
info put forth the proposition:
>
> On 2018-06-24 10:51 AM, David Wood fall wrote:
> > On Sunday 24 June 2018 08:58,
> > info put forth the proposition:
> > > Good morning,
> > >
> > >
> > > Whenever I boot
On Sunday 24 June 2018 08:58,
info put forth the proposition:
> Good morning,
>
>
> Whenever I boot a Fedora 26 box, the front headphone jack is muted and volume
> set to zero. No insertion or removal of the cable is detected.
>
> I have to use alsamixer to un-mute and bring the volume up. Then I
On Thursday 14 June 2018 21:24,
Pablo Rodríguez put forth the proposition:
> Dear list,
>
> I get too much noise (something similar to buzz) when I record my own
> voice on my laptop.
>
> I have used both Audacity and GStreamer. When I set recording volume
> control to 0dB, recording gets too much
On Friday 8 June 2018 22:03,
Alsa User put forth the proposition:
> David Woodfall wrote:
> > I've have had a Xonar D2X for some time now and normally use coax
> > spdif output to an SMSL Sanskrit DAC.
> > [...]
> > I couldn't get any signal via the optic
On Friday 8 June 2018 22:03,
Alsa User put forth the proposition:
> David Woodfall wrote:
> > I've have had a Xonar D2X for some time now and normally use coax
> > spdif output to an SMSL Sanskrit DAC.
> > [...]
> > I couldn't get any signal via the optic
I've have had a Xonar D2X for some time now and normally use coax
spdif output to an SMSL Sanskrit DAC.
Yesterday I was messing around with different rates and tried 192k. I
had a lot of gaps in the audio. Not being sure whether this could be
caused by the software, the D2X or my DAC I thought I'd
Great! Thanks Samuel.
On 14/04/15 20:12, Samuel Engelmajer wrote:
> Hi David.
>
> I will answer only for the arecord part.
>
> Looking at alsa-utils 1.0.29 and back (1.0.26, 1.0.21...) it seems that
> Wave header has never supported S24_3BE format.
> Or at least, that are
de: linux (3.19.3-1 -> 3.19.3-3)
System info is here:
http://www.alsa-project.org/db/?f=257c31b8c899be7febffb0a34a8823e36aff2006
How can I get arecord working? (then I'll worry about recording in JACK but
I
Dear ALSA developers and users,
How can I get this recent Roland USB audio device (TR-8) to work in Linux with
ALSA?
many thanks, please see below,
David.
On 30/12/14 19:39, David W. wrote:
> On 15/12/14 23:37, David W. wrote:
>> On 07/07/14 07:31, Clemens Ladisch wrote:
>>&
Dear ALSA developers and users,
How can I get this recent Roland USB audio device (TR-8) to work in Linux with
ALSA?
many thanks, please see below,
David.
On 30/12/14 19:39, David W. wrote:
> On 15/12/14 23:37, David W. wrote:
>> On 07/07/14 07:31, Clemens Ladisch wrote:
>>&
On 15/12/14 23:37, David W. wrote:
> On 07/07/14 07:31, Clemens Ladisch wrote:
>>
>> Apparently, this device needs some vendor-specific magic to enable
>> recording.
>>
Is enough information on this USB audio device provided below to create
an appropriate entry in
On 07/07/14 07:31, Clemens Ladisch wrote:
> David W. wrote:
>> On 06/07/14 20:48, Clemens Ladisch wrote:
>>> David W. wrote:
>>>> arecord -D hw:1,0 -f S32_LE -c14 -r96000 -t wav test.wav
>>>> Recording WAVE 'test.wav' : Signed 32 bit Little E
On 07/07/14 10:26, David W. wrote:
> On 07/07/14 07:31, Clemens Ladisch wrote:
>> David W. wrote:
>>> On 06/07/14 20:48, Clemens Ladisch wrote:
>>>> David W. wrote:
>>>>> arecord -D hw:1,0 -f S32_LE -c14 -r96000 -t wav test.wav
>>>>> R
On 07/07/14 07:31, Clemens Ladisch wrote:
> David W. wrote:
>> On 06/07/14 20:48, Clemens Ladisch wrote:
>>> David W. wrote:
>>>> arecord -D hw:1,0 -f S32_LE -c14 -r96000 -t wav test.wav
>>>> Recording WAVE 'test.wav' : Signed 32 bit Little E
On 06/07/14 20:48, Clemens Ladisch wrote:
> David W. wrote:
>> arecord -D hw:1,0 -f S32_LE -c14 -r96000 -t wav test.wav
>> Recording WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 96000 Hz,
>> Channels 14
>> arecord: pcm_read:2031: read error: Input/outp
On 05/07/14 19:55, Clemens Ladisch wrote:
> David W. wrote:
>> It has ... 14 audio inputs
>>
>> arecord ... -c1 ...
>> arecord: set_params:1239: Channels count non available
>
> Works as designed.
>
>
Oh yes! I misunderstood the message.
So using the
7;re worth a go for other users if there's a dev with enough time to
spare :-)
The recording is the most useful feature to get working (drum sounds
into the computer).
Thanks! Info below,
David W.
sh alsa-info.sh:
http://www.alsa-project.org/db/?f=1bd82a79fce0631dacb7dc057057f72c9e6909
David Vincent-Jones wrote:
> Reference:
> http://www.alsa-project.org/db/?f=b5c0d636c3870da14ab5f4e4c6f5f64d7c906507
>
> I am unable to get my laptop built-in microphone to work.
> Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In
> Connection: 7
> 0x18 0x19 0x1
On 15/01/14 12:12 PM, Bill Unruh wrote:
> On Wed, 15 Jan 2014, David Vincent-Jones wrote:
>
>>
>> Takashi Iwai suse.de> writes:
>>
>>>> David Vincent-Jones wrote:
>>>>> Reference:
>>>>> http://www.alsa-project.org/db/?f=b5c0d6
On 01/15/2014 12:12 PM, Bill Unruh wrote:
> On Wed, 15 Jan 2014, David Vincent-Jones wrote:
>
>>
>> Takashi Iwai suse.de> writes:
>>
>>>> David Vincent-Jones wrote:
>>>>> Reference:
>>>>> http://www.alsa-project.org/db/?f=
Takashi Iwai suse.de> writes:
> > David Vincent-Jones wrote:
> > > Reference:
> > > http://www.alsa-project.org/db/?f=b5c0d636c3870da14ab5f4e4c6f5f64d7c906507
> If this still doesn't work, it can be rather the way of testing.
> This digital mic might be
t found on my system.
At this point I have become stuck.
A question: I am on Fedora-20+XFCE; is it possible that this
'lightweight' desktop is less capable/complete in my case than working
with either Gnome or KDE? If so I would make a switch in a heartbeat.
David
On 01/13/2014 06:45 PM,
Thank you Dominique I was able to load both of these modules and
can confirm that they are now correctly listed. Unfortunately the
problem remains.
David
On 01/13/2014 04:06 PM, Dominique Michel wrote:
> Le Mon, 13 Jan 2014 12:48:02 -0800,
> David Vincent-Jones a écrit :
>
>
appreciated.
David
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G
In order to add information to my prior request for help I have run
"aadebug" and I attach the results:
David
[david@localhost Desktop]$ ./aadebug
ALSA Audio Debug v0.2.0 - Fri Jan 10 14:32:16 PST 2014
http://alsa.opensrc.org/aadebug
http://www.gnu.org/licenses/agpl-3.0.t
o.
I have followed several suggestions on various forums without success.
Assistance would be most appreciated.
David
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More details on the Raspberry Pi forums:
http://www.raspberrypi.org/forum/viewtopic.php?f=53&t=64936&p=478560#p478560
But essentially the problem is that playing with plug:dmix blocks waiting for
an event to arrive from /dev/snd/timer.
Should I send this to the dev list?
Cheers,
Dave
-
As per the subject I'm attempting to send single packets to, actually, a
Novation Launchpad S. So not something that conveniently makes a noise when it
receives a packet but, in theory, it can make it's switches light up.
(http://global.novationmusic.com/support/downloads/launchpad-programmers-r
I think I don't understand but whatever, I guess I'll have to look more into
some documentation. Thank you for solving my problem however. Bye.
- Original Message -----
> David Kačerek wrote:
> > why do I have to mute S/PDIF in alsamixer when in asound.conf I
> > h
Dne St 27. listopadu 17:22:40, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> > Dne St 27. listopadu 12:57:26, Clemens Ladisch napsal(a):
> >> David Kačerek wrote:
> >>>> The S/PDIF output does not support 192 kHz.
> >>>>
> >>>>
Dne St 27. listopadu 12:57:26, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> >>> Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital
> >>>
> >>> Device: name="CX20561 Digital", type="SPDIF", device=1
> >>>
> &
- Forwarded Message -
>
>
> - Original Message -
> > David Kačerek wrote:
> > >>> I wanted to list supported sample frequencies of my sound card
> > >>> and it
> > >>> shows only 2: 44100 96000, even though according t
In case of a need I attach my asound.conf.
- Original Message -
> David Kačerek wrote:
> >>> I wanted to list supported sample frequencies of my sound card
> >>> and it
> >>> shows only 2: 44100 96000, even though according to the official
>
Dne Po 25. listopadu 12:29:27, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> > isn't it weird to require an audio file if I just want to dump
> > parameters of a HW device?
>
> The aplay tool is for playing. The --dump-hw-params option just prints
> the param
Dne Ne 24. listopadu 15:25:53, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> > when executing aplay --dump-hw-params it allways kinda freezes
>
> When you run aplay without a file name, it tries to read audio data
> from its standard input.
>
>
> Regards,
&g
nses/agpl-3.0.txt
Kernel ----
Linux David-ArchLinux 3.12.1-1-ARCH #1 SMP PREEMPT Thu Nov 21 08:18:42 CET 2013
x86_64 GNU/Linux
Advanced Linux Sound Architecture Driver Version k3.12.1-1-ARCH.
Loaded Modules
snd_hda_c
e. What
happened?
Thank you.
Regards,
David.
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Hi Guys,
I started having this problem few days ago, not sure what caused it.
first my setup:
mac mini 6.1 (late 2012) running debian x64 - kernel 3.8.10
denon 4311 avr, connected via HDMI
5.0 speaker setup
Previous to a few days ago, when I am just playing mp3 tracks I will use
mplayer and it w
On Thu, Feb 07, 2013 at 11:00:13AM +0100, Daniel Vetter wrote:
>On Wed, Feb 06, 2013 at 10:35:33PM +0100, David Härdeman wrote:
>> a) switching between 1080p30 and 1080p50 or 1080p60 is enough to make
>> the sound go away (higher frame rates) or work (1080p30). So, it has
>&g
On Thu, Feb 07, 2013 at 11:00:13AM +0100, Daniel Vetter wrote:
>On Wed, Feb 06, 2013 at 10:35:33PM +0100, David Härdeman wrote:
>> I'll break etiquette here and include the entire original message below
>> (and top-post!) since I'm sending this to intel-gfx as well.
>&
60 is included below. It's interesting to note
that all the modes that don't work have fdi_lanes = 2 while the working
ones have fdi_lanes = 1 (port width in intel_reg_dumper-speak).
I'm CC:ing the intel-gfx list as well as the ALSA list since I'm not su
sure where the prob
tal), libdrm is 2.4.40, the
xorg intel driver is 2.19.0. I've also tried earlier versions of e.g.
the kernel without any success.
Suggestions?
--
David Härdeman
[1]
http://www.asrock.com/nettop/overview.us.asp?Model=Vision%20HT%20Series
[2]
http://www.pioneer.eu/eur/products/42/98/405/SC-LX76-
Hello, https://bbs.archlinux.org/viewtopic.php?id=88556.
Basically i have got same issue, same card, and a lot newer kernel,
but still, Audio Card is not being detected(shows up with correct name
under dmesg, but not in aplay/alsamixer/any place.
Is there anything i could do?
--
Regards,
David
Dne Ne 6. ledna 14:51:40, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> > So is there a way to make mplayer respect "defaults.pcm.rate_converter" or
> > force it to use speexrate_best or samplerate_best some other way?
>
> mplayer never uses ALSA'
Dne So 5. ledna 22:37:47, Clemens Ladisch napsal(a):
> David Kačerek wrote:
> > ... to prevent pointless and counterproductive resampling
>
> Resampling happens only when needed. And how can it be counterproductive?
>
> To check whether resampling is done, use apl
Dne Čt 3. ledna 21:50:32, David Kačerek napsal(a):
> Hi,
> I've got Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)
> sound card which support most of sample rates used in my music library so I
> have set my sw players to send audio streams straight to the &quo
Hi,
I've got Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)
sound card which support most of sample rates used in my music library so I
have set my sw players to send audio streams straight to the "hw:0,0" device
(or to one of virtual devices containing chain of ladspa plugi
Hi,Does anyone of you able to run Hdspe AES on Ubuntu 12.04 or 12.10?
I have just done a fresh install of 12.04 n 12.10, tried kernel 3.2.X, 3.4.0
and 3.5.19 unfortunately none of them run properly with Alsa 1.0.25.
However it works perfectly on 10.04 with alsa 1.0.25, so basically same
hardware
I have a Turtle Beach Amigo II USB audio adapter connected to my Raspberry Pi.
I have already commented out the following line in
"/etc/modprobe.d/alsa-base.conf":
# options snd-usb-audio index=-2
Now my USB audio adapter is first in the list when I run "aplay -l", or "aplay
-L", or "cat /pr
On (13:39 23/02/12), David Woodfall put forth the
proposition:
>I have just bought a soundcard which uses the ice1724 module and had
>to put the following in my .asoundrc to get optical out to work:
>
>pcm.ice1724 {
> type hw
> card 0
>}
>
>ctl.ice1724 {
I have just bought a soundcard which uses the ice1724 module and had
to put the following in my .asoundrc to get optical out to work:
pcm.ice1724 {
type hw
card 0
}
ctl.ice1724 {
type hw
card 0
}
pcm.!default {
type plug
slave.pcm "spdif"
}
It works, but my questio
On (16:54 17/02/12), David Woodfall put forth the
proposition:
>On (17:14 17/02/12), Clemens Ladisch put forth the
>proposition:
>>David Woodfall wrote:
>>> I recently bought a DAC unit with one of these chips and although
>>> alsamixer shows PCM volume/mute the
On (17:14 17/02/12), Clemens Ladisch put forth the
proposition:
>David Woodfall wrote:
>> I recently bought a DAC unit with one of these chips and although
>> alsamixer shows PCM volume/mute they have no effect on the output.
>
>Are there any error messages in the system
I recently bought a DAC unit with one of these chips and although
alsamixer shows PCM volume/mute they have no effect on the output.
I have alsa 1.0.25 installed.
As I mentioned in another post I have added a softvol but would
obviously like to get the proper control working.
Any ideas about thi
Hi, I added a softvol for my USB DAC since the PCM wasn't working. The
softvol works fine, but how to I add a mute to it?
--
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On (22:54 30/01/12), David Woodfall put forth the
proposition:
>I've found a few things on the net about raising volume by adding a
>softvol using dmix, but is it possible to do something like this while
>still keeping to hw: or plughw: ?
>
>Basically my DAC is much quieter
I've found a few things on the net about raising volume by adding a
softvol using dmix, but is it possible to do something like this while
still keeping to hw: or plughw: ?
Basically my DAC is much quieter than the rest of my Hifi and I'd like
to get some kind of balance there.
Cheers
--
Studio
pling rate while using external
clock. So my question is installing the alsa-firmware will help to solve the
problem? Or is there any solution to this?
Btw I'm using Ubuntu 10.04LTS console-mode only and do not have any x-windows
installed.
Thanks so m
Support for SPDIF on card:
Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02)
Can't seem to get it working. Is it not supported yet?
--
Don't look back, the lemmings are gaining on you.
--
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On (12:45 05/11/11), Torsten Schenk put forth the
proposition:
> > Is there any way of finding the sample rate being output from alsa?
>Yes, you can use
>cat /proc/asound/card0/pcm0p/sub0/hw_params
>
>to display you current card mode. If you have multiple cards/subdevices per
>card/... just brows
On (10:39 05/11/11), David Woodfall put forth the
proposition:
>Is there any way of finding the sample rate being output from alsa?
>
>Scenario is that I have bought a small DAC which works fine, but I
>would like to use the alsaequal mixer with it. Does the fact that is
>going th
Is there any way of finding the sample rate being output from alsa?
Scenario is that I have bought a small DAC which works fine, but I
would like to use the alsaequal mixer with it. Does the fact that is
going through alsa mean that the rate will be upsampled?
Details on alsaequal are here (great
Hi
I'm working with a couple of guys on a project to bring all the best
audio apps to Slackware Linux. The project is called Studioware and we
are based at www.studioware.org and #studioware on freenode for those
slackers out there that might be interested. Anyway.. the question:
Someone pointed
is not at fault. I'm happy
to collect further data, but I'm not really sure where to start!
On Fri, Sep 23, 2011 at 2:06 AM, Wouter Verhelst wrote:
> On Thu, Sep 22, 2011 at 07:28:37PM -0400, David O wrote:
>> Hello all,
>>
>> I recently set up a home theater PC runni
d asked for. Please let
me know if anyone has any suggestions for how I can get ALSA to
properly handle my HDMI output.
Thanks,
David
alsa-info
Description: Binary data
amixer_c0
Description: Binary data
eld
Description: Binary data
---
d asked for. Please let
me know if anyone has any suggestions for how I can get ALSA to
properly handle my HDMI output.
Thanks,
David
alsa-info
Description: Binary data
amixer_c0
Description: Binary data
eld
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---
evel mailing
> list for guidance. Although I'm not on that list. alsa-project.org
> should have info on that one.
>
> - James
>
>
> On 6/21/11, David Henderson wrote:
>> Hi James, if I use "configure --prefix=/opt/staging/alsa" then I'll have
>> t
s already there, not re-invent it in your
> image. And yes a bit OT at this point.
>
> - James
>
>
> On 6/20/11, David Henderson wrote:
>> I think your statement here "
>>
>> i.e. how exactly would you create your tarball? From a diff of an
>> entire bac
y hell I guess, of sorts.
>
> Lots of little things that will keep you from succeeding. It's
> probably time better spent learning an existing package management
> system IMO. Than to create your own. Especially if you're on your
> own and not part of team. But it'
On 06/20/2011 11:52 AM, Pierre Lorenzon wrote:
> Hi,
>
>
>
> From: David Henderson
> Subject: Re: [Alsa-user] First post
> Date: Sun, 19 Jun 2011 15:28:48 -0400
>
>> Thanks for the reply Pierre. I checked into the blfs book, but
>> it merely says "thes
wheel as previously said. But
> sometimes your distro doesn't package things in a way that you want to
> use them. i.e. Timidity with sequencer support. Jackd with sequencer
> support. Alsa with OSS emulation. And other fine tuning type needs.
> Or your distro is on such
ess you have a lot of time to waste, or just need the learning,
>>> I'd recommend going with existing distros. There's enough of them
>>> that one might suit your current needs. www.distrowatch.com
>> I suggested something intermediary : lfs or gentoo wh
th like software requires the
> --with-sequencer=yes if your card doesn't have native midi abilities
> (most don't these days). And various pulse-audio and browsers and
> other things that just need --with-oss=yes or things might not work as
> expected, if at all. Little thin
unless you have a lot of time to waste, or just need the learning,
> I'd recommend going with existing distros. There's enough of them
> that one might suit your current needs. www.distrowatch.com
>
> HTH,
> - James
>
>
>
> On 6/19/11, David Henderson wrote:
>
06/19/2011 11:22 PM, Pierre Lorenzon wrote:
> Hi,
>
> It looks like to me such questions are well answered in the
> blfs book. I personnaly think that the latter is a very good
> tool to build his own custom distro.
>
> Bests
>
> Pierre
>
>
> From: David Henderson
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