0.00] WARNING: CPU: 0 PID: 0 at
/build/buildd/linux-3.11.0/drivers/iommu/dmar.c:488
warn_invalid_dmar+0x8d/0xa0()
[0.00] Your BIOS is broken; DMAR reported at address fed92000 returns
all ones!
[0.00] BIOS vendor: Hewlett-Packard; Ver: 786F1 v01.32; Product
Version:
[
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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https://bugs.launchpad.net/bugs/1343187
Title:
sound automatically switch on and
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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Title:
[Amilo Si 1520, Conexant CX20549
active profile: output:hdmi-stereo
sinks:
alsa_output.pci-_00_1b.0.hdmi-stereo/#0: Built-in Audio
Digital Stereo (HDMI)
sources:
alsa_output.pci-_00_1b.0.hdmi-stereo.monitor/#0: Monitor of
Built-in Audio Digital Stereo (HDMI)
ports:
Codec: Realtek ALC282
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0282
Subsystem Id: 0x103c2170
Revision Id: 0x13
No Modem Function Group found
Default PCM:
N/A
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
State of AFG node 0x01:
Power: setting=UNKNOWN, actual=UNKNOWN,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/?qt=grepq=alc282
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Title:
[HP Pavilion 15
seem missing speaker node and driver ignore node 0x0f
16.049837] autoconfig: line_outs=1 (0x10/0x0/0x0/0x0/0x0) type:hp
[ 16.049843]speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 16.049845]hp_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 16.049846]mono: mono_out=0x0
[ 16.049848]
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=d08c5ef2a039393eaf2ab2152db5f07790fa0f40
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/hda_auto_parser.c?id=9b7564a64999597844513604df4a206fa4da3b69
seem driver ignore line out when Default association is zero
case AC_JACK_LINE_OUT:
seq =
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[Vaio VGN-A417S] XUbuntu 14.04 NO
you have to use hda Jack retask to Change the node 0x0d from speaker to
line out
http://voices.canonical.com/david.henningsson/2011/11/29/turn-your-mic-
jack-into-a-headphone-jack/
https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers
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13.068894] autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker
[ 13.068901]speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 13.068904]hp_outs=1 (0xa/0x0/0x0/0x0/0x0)
[ 13.068906]mono: mono_out=0x0
[ 13.068908]dig-out=0x10/0x0
[ 13.068910]inputs:
[ 13.068914] Front
post pulseaudio verbose log
seem pulseaudio remove line out / speaker path when there is only one
pcm playback volume control
Node 0x0d [Pin Complex] wcaps 0x400181: Stereo
Control: name=Speaker Jack, index=0, device=0
Pincap 0x173f: IN OUT HP Detect Trigger ImpSense
Vref caps: HIZ
should be Jack line out instead of Jack speaker if it is a desktop
you need to file an upstream bug report
12.982457] snd_hda_intel :00:1b.0: irq 40 for MSI/MSI-X
[ 13.068894] autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker
[ 13.068901]speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
THE LOCKING CTL BUG can be easily reproduced by open two terminals
aplay -D front:AUDIGY2 -f DAT Foo1.wav
arecord -D front:AUDIGY2 -f DAT Foo2.wav
you need to change /usr/share/alsa/conf/cards/AUDIGY2.conf
so that the slave capture device has no locks or just define playback
slave only
xrandr -x verborse
does output show audio is enabled at different graphic modes ?
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Title:
Only audio over HDMI
you have to ask the graphic driver developer since the graphic driver
have the abillity todisable hdmi audio when bandwidth is not enough
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xrandr --verbose
does output show audio is enabled at different graphic modes ?
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Title:
Only audio over HDMI
+ type asym
+ playback.pcm {
type hooks
slave.pcm {
type hw
card $CARD
}
hooks.0 {
type ctl_elems
hook_args [
+ }
+ capture.pcm {
+ type hw
+ card $CARD
+}
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ports:
analog-input-microphone: Микрофон (priority 8700, latency
offset 0 usec, available: unknown)
properties:
device.icon_name = audio-input-microphone
analog-input-linein: Линейный вход (priority 8100,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/oxygen?id=e96f38f732d24515792296b3738842934c985539
seem just using AC97 mic1 mic2 but Change the name of the control
you have to use Mic Source
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths/analog-input.conf.common
however front panel seem not a defined name for input
and you just have three capture switches instead of capture source
control for selecting Mic, line and
how do this driver allow you to select capture source?
do the control allow you to turn ON more than one input capture switch
at same time ?
SImple mixer control 'Line',0
Capabilities: cswitch cswitch-joined penum
Capture channels: Mono
Mono: Capture [off]
Simple mixer control 'Mic',0
the following are normal AC97 capture controls , the driver can be
customised to remove some unused input sources
control.20 {
iface MIXER
name 'Capture Source'
value.0 Mic
value.1 Mic
comment {
access 'read write'
type ENUMERATED
count 2
item.0 Mic
item.1 CD
pactl list of sound card with AC97 codec
alsa.components = AC97a:41445374
module-udev-detect.discovered = 1
device.icon_name = audio-card-pci
ports:
analog-input-microphone;input-microphone-1: Microphone /
Microphone 1 (priority
[Element Mic Source]
enumeration = select
[Option Mic Source:Mic]
name = input-microphone
priority = 20
[Option Mic Source:Front Panel]
name = input-microphone
priority = 19
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Simple mixer control 'Mic Boost (+20dB)',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [off]
it is a bug of alsa lib/driver to regard this switch as pswitch since
this affect both playback and capture
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/HD-Audio-Models.txt
name Description
-----
ALC880
==
3stack3-jack in back and a headphone out
3stack-digout 3-jack in back, a HP out and a SPDIF out
5stack
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[, Realtek ALC880, Line Out, N/A]
Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out
Control: name=Front Playback Volume, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x00 0x00]
Amp-Out caps: ofs=0x1f,
you have to fix the pin by hda jack retask
try hda jack sense to find the nodes of HP and Mic jack by plug and
unplug
remove redundant pin e.g. CD
if the internal speaker and Mic is not correct, you have to retask the
remaining nodes one at a time
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post output of
xrandr --verbose
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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https://bugs.launchpad.net/bugs/1394711
Title:
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda?qt=grepq=alc668
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Title:
[Dell
post system log
serm you have an oops
088789] Modules linked in: xt_conntrack ipt_REJECT pci_stub vboxpci(OX)
vboxnetadp(OX) vboxnetflt(OX) vboxdrv(OX) ip6table_filter ip6_tables
ebtable_nat ebtables bbswitch(OX) xt_CHECKSUM iptable_mangle ipt_MASQUERADE
iptable_nat nf_conntrack_ipv4
no mic jack
only intermal mic, headphone and speaker
control.10 {
iface CARD
name 'Mic Phantom Jack'
value true
comment {
access read
type BOOLEAN
count 1
seem only three pin complex
you have to use hda jack sense test to find out the missing mic jack
sys/class/sound/hwC2D0/init_pin_configs:
0x12 0x90a60130
0x14 0x90170110
0x15 0x0321101f
0x16 0x4000
0x18 0x41f0
0x19 0x41f0
0x1a 0x41f0
0x1b 0x41f0
0x1d 0x40d6832d
0x1e
Node 0x12 [Pin Complex] wcaps 0x400401: Stereo
Control: name=Mic Phantom Jack, index=0, device=0
Pincap 0x0020: IN
Pin Default 0x90a60130: [Fixed] Mic at Int N/A
Conn = Digital, Color = Unknown
DefAssociation = 0x3, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls: 0x20: IN
Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out
Control: name=Speaker Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name=Speaker Phantom Jack, index=0, device=0
Amp-In caps: N/A
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00,
http://voices.canonical.com/david.henningsson/2014/03/07/headset-jacks-
on-newer-laptops/
do you computer support headset, headphone and conventional mic?
try mute right channel or record in stereo to check the signal of left and
right are inverted
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http://bazaar.launchpad.net/~unity-settings-daemon-team/unity-settings-
daemon/trunk/view/head:/plugins/media-keys/what-did-you-plug-in/pa-
backend.c
the combo jack is setuo to support headphone, you need to change it to
headset or mic by the popup dialog
post the output of
pactl list source
check wthether the internal mic and headset mic are available
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Title:
refer to pa_backend.c
did you find headset mic jack or headset mic phantom jack in codec info
using dell headset model?
In PulseAudio ports will show up with the following names:
Headphones - analog-output-headphones
Headset mic - analog-input-microphone-headset
Jack in mic-in mode -
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name: auto_null
Default source name: auto_null.monitor
145984.025831] WARNING: CPU: 0 PID: 3405 at
/build/buildd/linux-3.16.0/drivers/gpu/drm/i915/intel_pm.c:6001
intel_display_power_put+0xf9/0x160
index: 8
name: alsa_card.usb-Plantronics_Plantronics_C420-00-C420
driver: module-alsa-card.c
owner module: 31
properties:
alsa.card = 2
alsa.card_name = Plantronics C420
alsa.long_card_name = Plantronics Plantronics
https://bugzilla.kernel.org/show_bug.cgi?id=88251
** Bug watch added: Linux Kernel Bug Tracker #88251
http://bugzilla.kernel.org/show_bug.cgi?id=88251
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if your computer chassis do not have hda front audio panel and the
front mic jack always return false
pulseaudio treat the front mic jack is not plugged
you need to disable the creation of jack detection control by set bit 8
of pin default
Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo
** Package changed: pulseaudio (Ubuntu) = alsa-driver (Ubuntu)
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Title:
speakers audio won't work - pavucontrol
your system log have lots of
201.019841] iwl3945 :02:00.0: Radio disabled by HW RF Kill switch
[ 201.020002] iwl3945 :02:00.0: Radio disabled by HW RF Kill switch
[ 201.021485] iwl3945 :02:00.0: Radio disabled by HW RF Kill switch
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_analog.c?id=0f7dbda0ec3bc4d778d7acf741b220fbf4318a2
you have to post output of alsa-info.sh
seem pro52h not in the above asus notebooks which use ad1986a
you need to report upstream pci developer
00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio
Controller (rev 02)
0.874389] pci :00:1b.0: [8086:27d8] type 00 class 0x040300
[0.874418] pci :00:1b.0: reg 0x10: [mem 0xd044-0xd0443fff 64bit]
[
CA0106.pcm.iec958.0 {
@args [ CARD AES0 AES1 AES2 AES3 ]
@args.CARD {
type string
}
@args.AES0 {
type integer
}
@args.AES1 {
type integer
}
@args.AES2 {
type integer
http://mailman.alsa-project.org/pipermail/alsa-
devel/2008-October/011913.html
: module-alsa-sink.c: Using 4 fragments of size 8816 bytes, buffer time is
99.95ms
D: module-alsa-sink.c: hwbuf_unused_frames=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to
do the error
alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77)
appear when you fix the error cannot lock ctl elem as pulseaudio
start the probing of those surround40, 51 and71 ?
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those locks in hook plugin prevent you running
aplay -D iec958:CARD=CA0106 stereo48000.wav
and
arecord -D iec958:CARD=CA0106 test.wav
at same time
those message Cannot lock ctl elem should disaapear when you can playback and
capture iec958 at same time ?
post pulseaudio verbose log
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(In reply to rkfg from comment #26)
Created attachment 111615 [details]
Fixed PA log
It now doesn't report Cannot lock ctl elem.
it is strange that pulseaudio still don't probe surround profiles
pactl list should list those surround profiles similar to
seem default sample 16 bits has no effect on format , pulseaudio prefer
32 bits
Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] card.c: Created 2
alsa_card.pci-_06_00.0
Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: cannot
disable ALSA period wakeups
Dec 30
for alsa-time-test.c
dev = argc 1 ? argv[1] : front:0;
cap = argc 2 ? atoi(argv[2]) : 0;
fillrate = argc 3 ? atoi(argv[3]) : 1;
there are three parameter which allow you to change
device
capture or playback
fillrate
most sound cards cannot use default fill rate 1
for hda-intel,
can you post the output of alsa-info.sh ?
did it support 7.1 since there are only 4 jacks ?
speaker-test -c 4 -t wav -D surround40:CARD=CA0106
speaker-test -c 6 -t wav -D surround51:CARD=CA0106
speaker-test -c 8 -t wav -D surround71:CARD=CA0106
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(In reply to rkfg from comment #27)
Created attachment 111616 [details]
pcm_avail log
Here's the output of pcm_avail.c program.
avail increment at 8 frames which is less than min period size 16 and
period size 1024
this mean some tsched feature are supported
hwbuf_unused can be set to non
/usr/share/alsa/cards/CA0106.conf
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Title:
[CA0106 - CA0106, playback] Playback problem - Surround dies, no valid
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
pcm.name {
type asym # Asym PCM
playback STR# Playback slave name
# or
playback { # Playback slave definition
pcm STR # Slave PCM name
http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-September/081501.html
you can try Alexander's pcm_avail.c to find out whether your ca0106 can
report DMA_RESIDUE_GRANULARITY_BURST ?
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http://support.lenovo.com/us/en/products/laptops-and-netbooks/thinkpad-x
-series-laptops/thinkpad-x240/documents/PD029567?tabName=ProductParts
The combo audio connector does not support a conventional microphone.
Combo audio connector
Connect headphones or a headset that has a 3.5-mm
08:47:21 work pulseaudio[2333]: Requested tlength=34.83 ms, minreq=5.80 ms
Feb 02 08:47:21 work pulseaudio[2333]: Adjust latency mode enabled, configuring
sink latency to half of overall latency.
Feb 02 08:47:21 work pulseaudio[2333]: Requested latency=11.61 ms, Received
latency=11.61 ms
Feb 02
(In reply to rkfg from comment #9)
LMMS uses 64 frames, 1.5 ms but actually it has latency of 30-35 ms as
pacmd list-sink-inputs says. Setting its output to ALSA results in
extremely garbled sound, the only two options that actually produce
reasonable sound is PulseAudio (bad latency) and SDL.
refer to src/core/Audio/AudioPulseAudio.cpp,
buffer_attr.maxlength = (uint32_t)(-1);
// play silence in case of buffer underun instead of
using default rewind
buffer_attr.prebuf = 0;
which motherboard are you using ?
do this motherboard support internal spdif connector to graphic card ?
it is rare that it only has internal spdif connector but not
optical/rca digital output
You can also connect the onboard header to an S/PDIF input header
available on some graphics cards
(In reply to rkfg from comment #7)
Created attachment 113083 [details]
Underruns with prealloc=2048
I was able to leverage the Underrun! issue by setting prealloc to 2048 in
kernel options (recompiled it). It can also be achieved via echo 2048
/proc/asound/card0/pcm0p/sub0/prealloc without
https://bugzilla.kernel.org/show_bug.cgi?id=24052
you have to follow up the up stream bug report
** Bug watch added: Linux Kernel Bug Tracker #24052
http://bugzilla.kernel.org/show_bug.cgi?id=24052
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Simple mixer control 'Speaker',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 74
Mono:
Front Left: Playback 0 [0%] [-9.99dB] [off]
Front Right: Playback 0 [0%] [-9.99dB] [off]
control.21 {
iface CARD
Hrm, that is actually a good question. In theory, I would expect
module-switch-on-port-available to switch profiles between 2.0 and 2.1
as headphones are plugged in and out, but in practice,
- I'm not 100% sure if our don't switch to HDMI might prevent
switching from 2.1 to 2.0 when
Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In
Control: name=Mic Boost Volume, index=0, device=0
ControlAmp: chs=3, dir=In, idx=0, ofs=0
Control: name=Mic Jack, index=0, device=0
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Pincap
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=73bdd597823e2231dc882577dbbaf8df92fe1775
you need to use debug version of alsa driver to check the system log to
find out whether your codec alc292 can automatcally differentiate
headset and
http://bazaar.launchpad.net/~unity-settings-daemon-team/unity-settings-
daemon/trunk/view/head:/plugins/media-keys/what-did-you-plug-in/pa-
backend.c
Headphone Mic Jack - indicates headphone and mic-in mode share the same jack,
i e, not two separate jacks. Hardware cannot distinguish
try record stereo instead of mono using audacity and check whether left
and right channel out of phase,
try mute one channel using alsa mixer or hda analyzer
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how about arecord ?
arecord -D plughw:0 -c 1 -r 48000 -f S16_LE -v --buffer-time=5000
test.wav
arecord -D plughw:0 -c 1 -r 48000 -f S16_LE -v test.wav
Feb 3 23:07:17 kaa-W740SU pulseaudio[2423]: ( 24.746| 0.000) [pulseaudio]
resampler.c: Channel matrix:
Feb 3 23:07:17 kaa-W740SU
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[W740SU, VIA VT1802, Mic, Internal]
Do you mean loopback mixing wasnot workas expected
since only output of node 0x13 was passthrough audio mixer 0x1b to
headphone and internal spesker ?
Node 0x0b [Pin Complex] wcaps 0x400581: Stereo
Control: name=Front Headphone Jack, index=0, device=0
Pincap 0x0001001c: OUT HP EAPD Detect
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[HDA-Intel - HDA Intel, playback]
you can try hda-analyzer to change
Amp-In vals: [0x80 0x80] [0x4a 0x4a] [0x80 0x80] [0x4a 0x4a]
and
Connection: 4
0x17* 0x18 0x23 0x24
as node 0x24 is connected to audio output, this allow you record what is
playing
Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In
Amp-In caps:
*** This bug is a duplicate of bug 1302090 ***
https://bugs.launchpad.net/bugs/1302090
** This bug has been marked a duplicate of bug 1302090
Dell Alienware 14, Speaker sound output is mono until a headphone jack is
plugged
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You need to file an upstream bug report
EAPD 0x3: BALANCED EAPD
Node 0x24 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out
Control: name=Speaker Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control:
seem jack detection problem as line out is not plugged
control.33 {
iface CARD
name 'Front Mic Jack'
value true
comment {
access read
type BOOLEAN
count 1
Created attachment 113564
add SNDRV_PCM_INFO_BATCH when precise_ptr is not enabled
rme9652 driver need to specify SNDRV_PCM_INFO_BATCH when precise_ptr is
not enabled
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refer rme9652_hw_pointer() to rme9652.c
if (!rme9652-precise_ptr)
return (status RME9652_buffer_id) ? period_size : 0;
if precise_ptr is not enabled by module parameter, it just return
pointer value as period_size or zero, this mean time scheduling should
be disabled since
http://lists.freedesktop.org/archives/pulseaudio-
discuss/2015-January/023042.html
you need a low pass filter for the subwoofet
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Default sink name: alsa_output.pci-_00_1b.0.iec958-stereo
Default source name: alsa_input.pci-_00_1b.0.analog-stereo
the default sink was changed to digital ouutput instead of analog
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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https://bugs.freedesktop.org/show_bug.cgi?id=84983
bug#96539
both are related to xonar sound card which have a multi channl volume
control which is not supported by pulseaudio
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Four Stereo ADC Inputs with Analogue Gain Adjust from +24dB to –21dB in 0.5dB
Steps
Digital Gain Adjust from -21.5dB to -103dB.
table 10 of wm8775 mention that 0.5 dB per steps and min 0 is mute
As max 255 is +24 dB , 0xcf is 0 dB ,
step 1 is -103 dB
Source #1
State: SUSPENDED
seem
chmap -Dsurround41:CARD=CA0106 query
fail but
chmap -Dsurround41:CARD=CA0106 get
return channel map
in ca0106.conf
there is no definition of ca0106.pcm.surround41
just
confdir:pcm/surround21.conf
confdir:pcm/surround41.conf
confdir:pcm/surround50.conf
Do surround21 ,
static unsigned int i2c_adc_init[][2] = {
{ 0x17, 0x00 }, /* Reset */
{ 0x07, 0x00 }, /* Timeout */
{ 0x0b, 0x22 }, /* Interface control */
{ 0x0c, 0x22 }, /* Master mode control */
{ 0x0d, 0x08 }, /* Powerdown control */
{ 0x0e, 0xcf }, /*
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_mixer.c?id=fff36e472b4315df77513f4339c5c199c6aad28b
static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
the scale only define step 1 is -103.50 dB and step size 0.5dB
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what is the smallest buffer size which aplay without underrun ?
1024,512,256,128,64,...
aplay -D front:CARD=CA0106 -v --buffer-size=1024 stereo48000.wav
19:29:42 homecomp pulseaudio[1911]: Protocol version: remote 26, local 29
Dec 25 19:29:42 homecomp pulseaudio[1911]: Got credentials:
if you cannot find stereo 48000Hz wav file
aplay -D plughw:CARD=CA0106 -v --buffer-size=1024 any.wav
try to find minimum buffer size 1024,512, 256,128,64,... without
underrun
1024 frames = 21.33 ms x 48000Hz
4096 = 85.33 ms x 48000Hz
if request tlength 85.33ms is buffer time and 21.33ms is
how about CD audio (44100Hz stereo) using plughw?
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https://bugs.launchpad.net/bugs/1315212
Title:
[CA0106 - CA0106, playback] Playback problem -
for unknown reason which pulseaudio did not probe surround40
pulseaudio still using timer base scheduling in your last log
but it was strange tradtional mode was used in your first log
20:25:49 homecomp pulseaudio[11971]: [pulseaudio] sink.c: alsa.name =
CA0106
Jan 02 20:25:49 homecomp
17.936573] snd_hda_intel :00:1b.0: irq 18 for MSI/MSI-X
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[ 18.948045] snd_hda_intel :00:1b.0: Codec #0 probe error; disabling it...
--
[ 19.956047] snd_hda_intel :00:1b.0: Codec #1 probe error;
disabling it...
[ 24.996042] snd_hda_intel :00:1b.0: no AFG or MFG node
did this happen when there are two or more user login since first user
's pulseaudio accquire the sound card and second user ' pulseaudio
cannot open the sound card any more ?
sudo fuser /dev/snd/*
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client(s) logged in.
index: 0
driver: module-systemd-login.c
owner module: 15
properties:
application.name = Login Session c3
systemd-login.session = c3
index: 1
driver: module-systemd-login.c
owner module: 15
+if (e-n_channels 2) {
+/* FIXME: In some places code like this is used:
+ *
+ * e-masks[alsa_channel_ids[p]][e-n_channels-1]
+ *
+ * The definition of
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