It does not look like PA use 4 fragments and 25ms
http://manpages.ubuntu.com/manpages/natty/en/man5/pulse-
daemon.conf.5.html
default-fragments= The default number of fragments. Defaults to 4.
default-fragment-size-msec=The duration of a single fragment. Defaults
to 25ms (i.e.
** Patch added: Patch which call pa_alsa_dump when alsa-sink ans alsa-source
underrun
https://bugs.launchpad.net/ubuntu/+source/linux/+bug/475355/+attachment/2118374/+files/0001-Add-pa_alsa_dump-when-alsa-sink-alsa-source-undderrun.patch
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please provide pulseaudio log and a test case which can reproduce the
problem
https://wiki.ubuntu.com/PulseAudio/Log
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https://bugs.launchpad.net/bugs/798504
Title:
[CS46xx - Sound
according to 92HD73C datasheet
1.4.11. Multi-channel capture
The capability to assign multiple ADC “Input Converters” to the same
stream is supported to meet
the microphone array requirements of Vista and future operating systems.
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You received this bug notification because
*** This bug is a duplicate of bug 760632 ***
https://bugs.launchpad.net/bugs/760632
Distro-Upgraded Kubuntu 10.10 (64 bit) to 11.04 when I open the Konsole and
click the Maximize gadget the screen flashes and the mouse pointer freezes.
After a few seconds the mouse pointer moves but
your codec info show two line in pin complex 0x15 and 0x1a
This node 0x1a seem to be wrong
Node 0x1a [Pin Complex] wcaps 0x40: Mono
Pincap 0x0020: IN
Pin Default 0x918711f0: [Fixed] Line In at Int Rear
Conn = Analog, Color = Black
DefAssociation = 0xf, Sequence = 0x0
you have to provide pulesaudio.log
seem that you overwrite default device
pcm.!default {
type plug
slave.pcm surround51
slave.channels 6
route_policy duplicate
}
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this is clearly underrun when appl_ptr 56512 is behind hw_ptr 146048
PA server only wrote 56512 frames but the sound card has already played
146048 frames, you can set log-time=yes in pulseaudio/daemon.conf to
compute the time elapsed to verify that the underrun occur
the sound card is not stop
if you look at http://git.alsa-project.org/?p=alsa-
utils.git;a=blob;f=alsactl/state.c
sprintf(name, hw:%d, card);
dbg(device='%s', doit=%i, name, doit);
err = snd_ctl_open(handle, name, 0);
it only use snd_card_next() to find out all the alsa cards and open
those hw:x
[ 17.873308] HDA Intel :04:01.0: PCI INT A - GSI 17 (level, low) - IRQ 17
[ 17.873418] HDA Intel :04:01.0: setting latency timer to 64
[ 17.873423] HDA Intel :04:01.0: PCI: Disallowing DAC for device
--
[VIA VT1708] pactl stat failed to find default card
it is because you turn on Independent HP
when Independent HP is off, audio through hw:0,0,0 go to both speaker and
headphone
and when Independent HP is on, audio through hw:0,0,0 goto speaker and audio
through hw:0,0,1 go to headphone
Simple mixer control 'Independent HP',0
Capabilities:
Oct 8 09:47:15 mike-desktop pulseaudio[1771]: alsa-util.c:
snd_pcm_avail_delay() returned strange values: delay 4 is less than
avail 6.
Oct 8 09:47:15 mike-desktop pulseaudio[1771]: alsa-util.c: appl_ptr : 9764858
Oct 8 09:47:15 mike-desktop pulseaudio[1771]: alsa-util.c: hw_ptr : 9764864
can
did your PA server log contain the error message when select 5.1 profile
as in Bug #455779
I: alsa-sink.c: Starting playback.
I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed
--
5.1 on SB Live! 5.1 [SB0060] is very unreliable
https://bugs.launchpad.net/bugs/406582
You received this bug
CrunchBang Linux install on a Pentium 4, 2.8ghz with an NVidia Geeforce 5700ve
and 1gb ram.
I no longer have this Desktop machine so I cannot reproduce it.
--
Installer Crash
https://bugs.launchpad.net/bugs/314201
You received this bug notification because you are a member of Ubuntu
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Card2.Amixer.info:
Error: command ['amixer', '-c', '2', 'info'] failed with exit code 1: amixer:
Mixer load hw:2 error: Invalid argument
Card hw:2 'Device'/'C-Media Electronics Inc. USB Multimedia Audio Device at
usb-:02:00.0-4.2, f'
Mixer name : 'USB Mixer'
Components :
it is not wine bug because they drop OSS emulation in ubuntu 10.10
you cannot
cat any.wav /dev/dsp
cat /dev/dsp any.wav
--
OSS no longer works as a standalone sound driver for WINE.
https://bugs.launchpad.net/bugs/662472
You received this bug notification because you are a member of Ubuntu
#3* pulseverbose.log (192.6 KiB, text/plain)
Thanks for your quick reaction.
I: alsa-sink.c: Starting playback.
I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed
D: module-suspend-on-idle.c: Sink
alsa_output.pci-_02_00.0.analog-surround-51 becomes busy.
--
pulseaudio takes 100
you have to select a model for your alc887 , the auto model does not
work
[ 12.828284] hda_codec: num_steps = 0 for NID=0xc (ctl = Front Playback
Volume)
[ 12.848284] hda_codec: num_steps = 0 for NID=0xc (ctl = Front Playback
Volume)
try those model of alc883/888 which create switch at
this solution did not work for me:
{[[ Please remove model=generic, the latest alsa driver modules (if installed),
then install
http://people.canonical.com/~diwic/temp/alsa-hda-diwic-alc887-mixer-dkms_1.0.23.diwic_all.deb
- reboot and report back if it works for you or not. Thanks!]]}
I had
this solution did not work for me:
{[[ Please remove model=generic, the latest alsa driver modules (if installed),
then install
http://people.canonical.com/~diwic/temp/alsa-hda-diwic-alc887-mixer-dkms_1.0.23.diwic_all.deb
- reboot and report back if it works for you or not. Thanks!]]}
I had
this solution did not work for me
I had to reinstall module=genetic
--
[ALC887] alsamixer not open
https://bugs.launchpad.net/bugs/669092
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refer to http://git.alsa-project.org/?p=alsa-
lib.git;a=commitdiff;h=bd7b73a621a74de863f6d6f79af4131144e4f9fe;hp=73d20069e072e2b62bbc3ea12e46ae19d8b6cac9
PA developer 's definition is
For capture the delay is defined as the time that a frame that was
digitized by the audio device takes until
Using 4 controls : Volume , Balance, Fade and Bass to control six or
eight channels is not easy especially when gnome-media cannot
save/restore the settings of this four controls
This bug is in pa_cvolume_set_balance() since left and right will be zero when
you set the volume to 0%
655
0:00:12.187248302 7969 0x94c2ae0 WARN alsa
pcm_pulse.c:696:pulse_hw_params: alsalib error: PulseAudio: Unsupported
format S32_LE
0:00:12.187281533 7969 0x94c2ae0 WARN alsa
gstalsasink.c:528:set_hwparams:alsasink0 error: Unable to set hw
params for playback: Invalid argument
--
alsa Unable to
There error message is printed by the recent patch , you have to ask PA
developer about Broken alsa _delay implementation
alsa: work around slightly broken _delay implementations
Use snd_pcm_avail_delay() in pa_alsa_safe_delay() so that we can check
the delay value against the avail value and
according to HDA specification
Input Delay is a 4-bit value representing the number of samples between when
the sample is
received as an analog signal at the pin and when the digital representation is
transmitted on the
High Definition Audio Link. This may be a “typical” value. If this is 0,
I have the same bug in Maverick and Natty. Who is responsible here -
ALSA or Pulseaudio!?
can you provide a testcase which can reproduce the problem and a full
pulseaudio log with timestamp by set log-time=yes in
pulseaudio/daemon.conf
pulseaudio -
--
pulseaudio[2034]: alsa-util.c:
Mixer name : 'USB Mixer'
Components : 'USB0763:2003'
Controls : 0
Simple ctrls : 0
your usb audio seem have not volume control
post the output of lsusb - of your usb audio device
and the output of pulseaudio - since PA must complain about
missing controls
--
pulseaudio hogs
Does this give you any clue?
you have to post the output of pulseaudio server
pulseaudio -k;pulseaudio -v
--
[CA0106 - CA0106] ALSA test tone not correctly played back
https://bugs.launchpad.net/bugs/671178
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you can use strace alsactl store to find out why X11 connection
rejected
--
alsactl should ignore the pulse plugin
https://bugs.launchpad.net/bugs/557016
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seem to be related to PA set the start threshold to -1
if you look at pcm.c , the start threshold is cast as a signed number
in some write function , this mean that snd_pcm_start() still
automatically called during wirte instead of manually started by PA
server
6740 if (state
This seem to be bug in gnome-alsamixer according to
http://thread.gmane.org/gmane.linux.alsa.devel/5047/focus=5060
--
error message when gnome-alsamixer is launching
https://bugs.launchpad.net/bugs/106903
You received this bug notification because you are a member of Ubuntu
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The point is gnome-media seem does not keep the setting of volume
e.g. when you increase the volume to over 100% , log out and login , you
will find the volume change to 100%
--
Changing volume to 0% breaks channel ballance
https://bugs.launchpad.net/bugs/672420
You received this bug
but there are three capture subdevices which allow user to capture from
three input sources independently
List of CAPTURE Hardware Devices
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
Subdevices: 3/3
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
does it help if your mute [Analog Mix] ?
http://www.analog.com/en/audiovideo-products/audio-
codecs/ad1988b/products/product.html
Simple mixer control 'Analog Mix',0
Capabilities: pvolume pswitch penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front
(In reply to comment #25)
Created an attachment (id=32743) [details]
Output of alsa-info.txt on my ALC889.
Includes arecord -l output under the ARECORD section.
DAC5 is node 0x25
/* FIXME: setup DAC5 */
spec-alt_dac_nid = 0x25;
spec-stream_analog_alt_playback =
Ubuntu 10.4 LTS Could not open location file:///home/ray failed to execute
child process /usr/bin/dolphin(No such file or directory)
Clean install but existing home partition. No icons or (my)wallpaper on
desktop. could the three /// be the problem?
Using the Bash shows /home/ray exists and I
Error Type:
Error Value: The cache has no package named 'dpkg-exec'
File : /usr/share/PackageKit/helpers/apt/aptBackend.py, line 1948, in
main()
File : /usr/share/PackageKit/helpers/apt/aptBackend.py, line 1945, in main
run(args, options.single)
File :
Kubuntu 9.04 after Dist-upgrade (still 9.04). no updates in tray but 301 in
System settings add/remove.
Broken filter-used synaptic to mend.
--
package perl-base 5.10.0-24ubuntu4 failed to install/upgrade: error writing to
'standard output': No such file or directory
Refer to log in #13
I: alsa-sink.c: Using 16.0 fragments of size 4096 bytes (23.22ms), buffer size
is 65536 bytes (371.52ms)
I: alsa-sink.c: Time scheduling watermark is 20.00ms
watermark 20ms is not enough since au88x0 has 4 hardware subbuffers
you cannot let PA sleep 348ms and wakeup 20.00ms
Soun d Architecture': Could not get/set settings from/on resource.
[gstalsasink.c(666 ): set_hwparams (): /pipeline0/alsasink1:
This is because au88x0 driver require the period_size to be power of two
snd_pcm_hw_constraint_integer(runtime,
Refer to the log
PA can open mono playback and mono capture using hw:0
D: alsa-mixer.c: Profile output:analog-mono+input:analog-mono supported.
PA can also open front:0 stereo playback
D: alsa-mixer.c: Profile output:analog-stereo supported.
But PA fail for open front:0 for stereo capture
I: alsa-sink.c: Using 32.0 fragments of size 2048 bytes (11.61ms), buffer size
is 65536 bytes (371.52ms)
I: alsa-sink.c: Time scheduling watermark is 20.00ms
: alsa-sink.c: Starting playback.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Cutting sleep
Sorry, I realized that was what it was telling me. Seeing as it was a
new installation I deleted all the hidden files in /home and
reinstalled. Its ok. now except for problems with thunderbird and imap
which I`ve sorted. (recreated account as pop)
--
cant open places ? Could not open location
so the current factory default levels (e g, what you get if you
boot a Live-CD)
The controls are muted by default by the driver to prevent accidental
damage of the speaker ,
The volume of Live-CD is the result of alsactl init
If you use alsactl store -f original.txt before you logout/reboot
when there is no sound card detected by pulseaudio server and PA server
load Dummy Output module
aplay -l
aplay : device_list :235 no soundcards found...
application can still use the pcm device of pulse with Dummy Output module
but ctl device of pulse seem disappear
alsamixer
cannot open
I'll agree with Martyn and Ralf and add that I use it at work in an
Exchange environment configured for IMAP and using Sunbird to manage
my calendar after becoming S T of Evolution's BS.
Thunderbird 3.1 looks amazing and am going to upgrade once my testing
is complete.
Raymond
On Sat, Mar 27
Public bug reported:
Binary package hint: rhythmbox
When I let the player do a station change, it tells me its paused.
ProblemType: Bug
Architecture: i386
Date: Fri Jan 29 10:40:00 2010
DistroRelease: Ubuntu 9.10
ExecutablePath: /usr/bin/rhythmbox
InstallationMedia: Ubuntu 9.10 Karmic Koala -
** Attachment added: Dependencies.txt
http://launchpadlibrarian.net/38521219/Dependencies.txt
** Attachment added: ProcMaps.txt
http://launchpadlibrarian.net/38521220/ProcMaps.txt
** Attachment added: ProcStatus.txt
http://launchpadlibrarian.net/38521221/ProcStatus.txt
** Attachment
free to mail me.
Greetings,
Raymond. P. HANSSENS.
From: Pedro Villavicencio pe...@ubuntu.com
To: rphanss...@yahoo.com
Sent: Friday, January 29, 2010 15:11:28
Subject: [Bug 514202] Re: Can't listen to station
Thanks for the report Raymond, which radio station? may
seem related to this patch
http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=18ee6dfae89d9c131e3c9952939633ba8fa86247
--
Sound card Yamaha YMF744b not working
https://bugs.launchpad.net/bugs/237925
You received this bug notification because you are a member of Ubuntu
Bugs, which is
Now what I CAN'T figure out is what exactly outputs the mmap()
function. Is it a library? Is it pulse? Something in the kernel? A
module? I don't FREAKING KNOW, and that's driving me nuts.
The reason is alsa-pulse plugin does not support snd_pcm_mmap_*
functions
so i t still a bug in pulseaudio since module-udev-detect should only
load the sound card
if it can skip the modem card , it should also skip the midi card too
--
pulseaudio module fails to ignore casio usb midi
https://bugs.launchpad.net/bugs/627758
You received this bug notification because
May 7 17:50:15 deepblue pulseaudio[9676]: alsa-util.c: stream : PLAYBACK
May 7 17:50:15 deepblue pulseaudio[9676]: alsa-util.c: access : MMAP_INTERLEAVED
May 7 17:50:15 deepblue pulseaudio[9676]: alsa-util.c: format : S16_LE
May 7 17:50:15 deepblue pulseaudio[9676]: alsa-util.c: subformat : STD
There are two demux (input source selector) , The first one in [Audio
Input] node 0x14 and the second one in node 0x17
There are three [audio input] 0x14, 0x15 ,0x16 and another selector 0x18
you will need to implement input source control which allow you to
select any input sources
Node 0x14
Refer to the pulseaudio log, udev only find the card 2 's hdmi and card
1 usbaudio
D: module-udev-detect.c: /dev/snd/controlC2 is accessible: yes
D: module-udev-detect.c:
/devices/pci:00/:00:02.0/:01:00.1/sound/card2 is busy: no
D: module-udev-detect.c: Loading module-alsa-card with
seem PCM softvol control was created when PA probe the front device but
your nvidia only have hdmi device
Does PA really support hdmi without Master Volume control ?
Card hw:0 'NVidia'/'HDA NVidia at 0xfbf78000 irq 21'
Mixer name: 'Nvidia MCP77/78 HDMI'
Components:
If they have removed oss emulation, they should also remove the option
to select OSS Driver in winecfg to avoid confusion
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https://bugs.launchpad.net/bugs/662472
Title:
OSS no longer
if they had removed oss emulation, they should also remove padsp and OSS
driver in winecfg
http://bugs.launchpad.net/bugs/579300
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https://bugs.launchpad.net/bugs/681572
Title:
Use of
* alsa-info.txt.FbRz3szq16 (28.0 KiB, text/plain)
Alsainfo for Maverick
There is only one capture subdevice but two capture volume
controls/switches and two input sources
ARECORD
List of CAPTURE Hardware Devices
card 0: Intel [HDA Intel], device 0: ALC259 Analog [ALC259 Analog]
MMarking wrote on 2009-12-02: #36
* Output of cat /proc/asound/card0/codec#0 (13.1 KiB, text/plain)
List of CAPTURE Hardware Devices
card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog]
Subdevices: 3/3
Subdevice #0: subdevice #0
Subdevice #1:
wine need to implement dsound 's speaker-configuration to find out your
speaker arrangement is 2.0 , 4.0 , 5.1 and 7.1 from Sound Preference and
use alsa's analog front , surround40, surround51 and surround71 devices,
iec958 for digital audio pass through coxial/spdif to your digital
receiver,
http://www.intel.com/support/motherboards/desktop/sb/cs-020642.htm
Those are motherboard with only 3 audio jacks at rear panel
Take a look at videos
For example, you can listen to one audio source through the back panel speakers
and a second audio source through front panel headphones or
according to the pdf , second audio source through front panel
headphones can be accessed through dsound
does your snd-hda-intel provide this independent headphone device ?
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The Multistreaming option of realtek and idt codec is similar to the
Independent Headphone switch in snd-hda-intel if you are using VIA HDA
Codec
http://www.viaarena.com/forums/showthread.php?t=41015
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Bugs, which is
alsa seem only implement the capture devices only
do you have additional capture device (hw:0,2) ?
arecord -l
Take a look at block diagram of alc889 datasheet and the codec-info
section in the output of alsa-info.sh
Ten DAC channels support 16/20/24-bit PCM format for 7.1 sound playback,
(In reply to comment #27)
Alsa works very well with multi-channel. I would take a look at the alsa
sources for this alsa program that works out of the box:
speaker-test
I am not a c/c++ programmer so I leave it to you.
ALSA require sound card to use surround40 , surround51 to play
(In reply to comment #18)
Thanks Raymond.
Doesn't UT3 use OpenAL?
Just took a look at the OpenAL WikiPedia page, and it does. So does UT2004
though, but that didn't work.
I noticed BioShock was on the list, so I tried a BioShock demo I downloaded
ages ago. It had an option for 5.1
(In reply to comment #2)
Amazingly this bug doesn't seem to have been filed before (support request for
multi-channel audio) so correcting the details to make more sense. This isn't
likely to be done for a while thought due to a) drivers not being that great,
b) EAX missing entirely from
That accelerated alsa driver for emu10k1 only provide stereo control of
those hardware mixing volume controls provided by snd-emu10k1 similar to
http://git.alsa-project.org/?p=alsa-
tools.git;a=blob;f=hwmixvolume/hwmixvolume
It actually playback different openal mono source through different
(In reply to comment #34)
A lot of people still have a Realtek ALC650 sound card so please support
multichannel for it too (for example I have a VIA KT333 chipset including it).
Thank you for your growing interest.
it the application does not provide 6 channels (e.g. openal only pan the
(In reply to comment #36)
-older games that rely much on multichannel to inform the player of the
position of things all around like: Counter Strike, Star Trek Armada II,
Homeworld 1 2, Giants - Citizen Kabuto, Command and Conquer 1 2 3
Those 3D game work best work with headphone when your
(In reply to comment #0)
Winamp with Winamp AC3 Filter 1.01a performs better than Amarok 1.4.
Also the very old game Starcraft has support for surround.
Therefore please give more importance to the following:
Allow DirectSound acces to Alsa, for the rear left, rear right, center and
woofer
To enhance the multi channels support of winealsa
you also need to enhance the mixer.c to add those Front, surround ,
center , lfe and side playback volume controls
Take a look at the output
WINETEST_INTERACTIVE=1 wine winmm_test mixer
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Cool, I now have microphone 1, microphone 2, and microphone 3.
Microphone 1 seems to be the digital mic. It would be nicer if it were
called digital (to match alsa) but this is a huge improvement on before.
Can you provide info about your hda codec ?
THe driver check Conn since one internal
The device 3 is not used for playing surround40 or surround51
card 0: x , device 3: emu10k1 [Multichannel Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
./alsacap -d hw:0,3
*** Exploring configuration space of device `hw:0,3' for playback ***
type : HW
16 channels
Sampling rate
you have to change max_buffer_size if you want to load a larger
soundfont as default value is 128Mb
http://git.alsa-project.org/?p=alsa-kernel.git;a=blob_plain;f=Documentation/sound/alsa/ALSA-Configuration.txt
Module snd-emu10k1
--
Module for EMU10K1/EMU10k2 based PCI
you have to turn on the Tone switch , otherwise Terble and Bass
has no effect
Simple mixer control 'Tone',0
Capabilities: pswitch
Playback channels: Front Left - Front Right
Mono:
Front Left: Playback [off]
Front Right: Playback [off]
--
You received this bug notification because you
I can load more than 16Mb soundfont into my sb live! platinum ct4760p on
my 32bits machines, so Is this bug specific to 64bits machine only
asfxload -M
DRAM memory left = 131068 kB
asfxload -M PCLite.sf2
DRAM memory left = 100523 kB
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The internal mic is at node 0x1e and it has no connection to Node 0x13
[Audio selector] connected to [Audio Input] 0x12
Node 0x1e is connected to Node 0x1f [Audio Input] , so you may need to
use hw:0,0,1
List of CAPTURE Hardware Devices
card 0: Intel [HDA Intel], device 0: VT1702
seem to be subdevice 2 and Digital Mic Capture Volume/Switch
static hda_nid_t vt1702_adc_nids[3] = {
/* ADC1-2 */
0x12, 0x20, 0x1F
};
/* capture mixer elements */
static struct snd_kcontrol_new vt1702_capture_mixer[] = {
HDA_CODEC_VOLUME(Capture Volume, 0x12, 0x0,
How about arecord -v -f cd -Dhw:0,0,2 test.wav ?
Simple mixer control 'Digital Mic',0
Capabilities: cvolume cswitch penum
Capture channels: Front Left - Front Right
Limits: Capture 0 - 12
Front Left: Capture 11 [92%] [16.50dB] [on]
Front Right: Capture 11 [92%] [16.50dB] [on]
Simple
How about arecord -v -Dhw:0,0,2 -f cd test.wav ?
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https://bugs.launchpad.net/bugs/677734
Title:
[VIA VT1702] Recording problem / Sound Recorder does not pick up sound
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ubuntu-bugs
There is problem if the driver group the three [Audio Input] 0x1f ,
0x12 and 0x20 in the same device since node 0x1f support 16bits 16000Hz
and 32000 Hz but not 20bit , 24bit or 19200Hz as the other node 0x12 and
0x20
The driver just query node 0x12 instead of 0x1f for the formats and
rates
arecord: pcm_read:1692: read error: Input/output error
you have to ask the alsa developer for this issue since 44100 Hz ,
16bits and 2 channels is still supported by 0x1f , 0x12 and 0x20, I
have no idea why
However many users expect they can use the internal mic by default
Headset jack has
I have an old Turtle Beach Santa Cruz sound card using the CS46xx
driver and was having exactly these symptoms after upgrading to Karmic
(audio with pulseaudio was working perfectly in Jaunty).
What are the differences of pulseaudio in Jaunty and 0.9.16 ?
e.g. set the stop threshold to boundary
The point is only spec-adc_nids[0] is used in via_build_pcms
1953 static int via_build_pcms(struct hda_codec *codec)
1954 {
1955 struct via_spec *spec = codec-spec;
1956 struct hda_pcm *info = spec-pcm_rec;
1957
1958 codec-num_pcms = 1;
1959 codec-pcm_info =
The solution proposed by Szabó could be a temporal workaround, but
not a fix as pulseaudio is an essential component of ubuntu.
I have not yet tried to recompile wine from source, but made a test
with WINEDEBUG=warn+all and attached the output logs to this bug in
case that someone finds
-41.dB is quite low
Simple mixer control 'Speaker',0
Capabilities: pvolume pswitch pswitch-joined penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 44
Mono:
Front Left: Playback 0 [0%] [-41.00dB] [on]
Front Right: Playback 0 [0%] [-41.00dB] [on]
does PA find the
You need it in both the hda controller and codec (aka routing).
if the hda controller does not have enough dma for two capture stream,
the driver should not expose the device to use
The problem is how can the application know which Input source control
should be used for which capture device
your cs46xx have dual ac97 codecs CS4297A and CS4294
There are two Master Playback Volume controls, I guess PA mis
calculate the dB range -129.00 dB to 12.00 dB.
I: sink.c: device.description = CS 4614/22/24/30 [CrystalClear SoundFusion
Audio Accelerator] Analog Stereo
I: sink.c:
Aibara Iduas wrote on 2010-12-06: #89
* pulseverbose.log (151.1 KiB, text/plain)
( 10.906| 0.000) D: source.c: Processing rewind...
( 12.662| 1.755) W: ratelimit.c: 16 events suppressed
( 12.662| 0.000) I: alsa-sink.c: Underrun!
( 13.279| 0.616) D: sink-input.c: Requesting
On the latency webpage, did you also read the section about bypassing
pulseaudio/dmix (-o audio.alsa.device=plughw:0)? Or do you perhaps do
not use pulseaudio or dmix at all?
it also depend on the sample rate of the soundfonts in sf2 match with
the sample rate of audio.alsa device
if the
This mean that this is a bug of xfce4-mixer which cannot grey out those
inactive control
Please try alsamixer to find out those inactive control
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https://bugs.launchpad.net/bugs/669261
can you post the output of amixer and lsusb - of da40 and da45 ?
do these device also have a mono playback volume control?
you have to follow up your case in pulseaudio-discuss mailing list about the
clipping of the mono playback,
the clipping may also ocurr with recording mono with a
there are two ext mic nodes and one int mic node
Node 0x1a [Pin Complex] wcaps 0x400481: Stereo
Control: name=Mic Jack, index=0, device=0
Pincap 0x1324: IN Detect
Vref caps: HIZ 50 80
Pin Default 0x03a1103e: [Jack] Mic at Ext Left
Conn = 1/8, Color = Black
DefAssociation =
you have to follow up the case upstream if you think this is a
regression
git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=ee0eb25119c2c948b5d30da1a62f6d44020cb636;hp=12837c983dc5ec56155b1e95a6fa9a74e4da381f
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select 6ch channel mode retask the pink and blue jack as output
rear mic and line in are used as surround and clfe
Simple mixer control 'Channel Mode',0
Capabilities: enum
Items: '2ch' '4ch' '6ch'
Item0: '6ch'
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*** This bug is a duplicate of bug 986710 ***
https://bugs.launchpad.net/bugs/986710
Simple mixer control 'PCM',0
Capabilities: pvolume pswitch pswitch-joined penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 0 [0%] [-34.50dB] [off]
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
/analog-output-lfe-on-mono.conf
; Intended for usage in laptops that have a separate LFE speaker
; connected to the Master mono connector
this meam that it is bug of pulseaudio since this is not a laptop
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