maj 27 17:17:12 hostname kernel: snd_hda_intel :00:03.0: Cannot turn
on display power on i915
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https://bugs.launchpad.net/bugs/1459310
Title:
[HDA-Intel - HDA
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=1d04c9de5c76df113e4af7120feb53c628b5efcc
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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post the output of alsa-info.sh
https://wiki.ubuntu.com/PulseAudio/Log
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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May 26 14:37:03 username-MS-7850 kernel: Could not create debugfs 'switch_mm'
directory
May 26 14:37:03 username-MS-7850 kernel: Could not create debugfs
'i915_context_free' directory
May 26 14:37:03 username-MS-7850 kernel: Could not create debugfs
'i915_context_create' directory
May 26
in addition to comment the volume entry of [Element Master] , you may
need to comment volume entry of [Element PCM] in analog-
output.conf.common
the dB range of three paths in pulseaudio log should be the same (i.e.
-91.5 dB +3 dB)
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=3abb4f4d0e7aaad0d12004b5057f4486a688752b
this patch change PCM playback Volume to Line Out Playback Volume
softvol plugin create a user control PCM playback Volume with dB range
-51 dB
control.XX {
As third use case, sometimes you might want to have simultaneous
output to Line Out and Headphones output, but then you may want to
change their volumes independently.
https://bugs.freedesktop.org/show_bug.cgi?id=90496
** Bug watch added: freedesktop.org Bugzilla #90496
Simple mixer control 'Mic',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 23 [74%] [0.00dB] [on]
Front Right: Playback 23 [74%] [0.00dB] [on]
your hear mic signal because mic playback switch is ON
Node
recording from audio mixer node 0x0b is disabled on notebook because of
auto mic switch
Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In
Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80]
[0x80 0x80]
http://lists.freedesktop.org/archives/pulseaudio-
discuss/2015-May/023902.html
you need this patch since [Element Line Out] is not defined in lineout.
ong
https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio
open upstream bug report if it still not work
** Changed in: pulseaudio
mag 26 13:35:14 hostname kernel: sound hdaudioC0D0: The codec is being used,
can't reconfigure.
mag 26 13:35:32 hostname kernel: sound hdaudioC0D0: The codec is being used,
can't reconfigure.
mag 26 13:36:20 hostname kernel: sound hdaudioC0D0: The codec is being used,
can't reconfigure.
**
WARNING: CPU: 0 PID: 0 at /build/buildd/linux-3.16.0/drivers/iommu/dmar.c:760
warn_invalid_dmar+0x6d/0x85()
mag 26 12:57:26 hostname kernel: Your BIOS is broken; DMAR reported at address
fed9 returns all ones!
BIOS vendor: American Megatrends Inc.; Ver: 2101
Default sink name: alsa_output.pci-_01_00.1.hdmi-stereo
Default source name: alsa_output.pci-_01_00.1.hdmi-stereo.monitor
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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Sysfs Files
!!---
/sys/class/sound/hwC0D0/init_pin_configs:
0x24 0x01014010
0x25 0x01011012
0x26 0x01016011
0x27 0x01012014
0x28 0x0221401f
0x29 0x02a19037
0x2a 0x0181303e
0x2b 0x01a19036
0x2c 0x90331138
0x2d 0x185600f0
0x2e 0x074411f0
/sys/class/sound/hwC0D0/driver_pin_configs:
https://wiki.ubuntu.com/PulseAudio/Log
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https://bugs.launchpad.net/bugs/1458849
Title:
[ Asus P7P55D LE , VIA VT1828S, Green Line Out, Rear] Pulseaudio fails
to
is there any point to change line out playback volume to minimum when speaker
profile is selected
you have to follow up with the author if his patch still not fix your problem
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf
the balance slider in sound preference is useless when volume slider is
using with the mono virtual master playback volume
Card hw:0 'Intel'/'HDA Intel at 0xfe9fc000 irq 44'
Mixer name: 'Analog Devices AD1983'
Components: 'HDA:11d41983,102801da,00100400'
Controls : 28
Simple
in theory ,you just need pulseaudio to mute line out playback switch
when headphone is plugged
and pulseaudio use line out playback volume when line out profile is
selected
index b6ee70b..a7a2140 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf
+++
http://colin.guthr.ie/2010/09/compiling-and-running-pulseaudio-from-git/
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https://bugs.launchpad.net/bugs/1436269
Title:
When you select Analog Output or Speakers as
http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grepq=line+out
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https://bugs.launchpad.net/bugs/1436269
Title:
When you select Analog Output or Speakers as
mag 26 12:57:52 hostname pulseaudio[719]: [pulseaudio] server-lookup.c: Unable
to contact D-Bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch
a dbus-daemon without a $DISPLAY for X11
mag 26 12:57:52 hostname pulseaudio[719]: [pulseaudio] main.c: Unable to
contact D-Bus:
mag 26 12:57:51 hostname pulseaudio[719]: [pulseaudio] authkey.c: Failed to
open cookie file '/var/run/speech-dispatcher/.config/pulse/cookie': File o
directory non esistente
mag 26 12:57:51 hostname pulseaudio[719]: [pulseaudio] authkey.c: Failed to
load authentication key
you have to post pulseaudio verbose log
state.Intel {
control.1 {
iface MIXER
name 'Line Out Playback Volume'
value.0 61
value.1 61
comment {
access 'read write'
** Bug watch added: Linux Kernel Bug Tracker #98921
http://bugzilla.kernel.org/show_bug.cgi?id=98921
** Also affects: alsa-driver via
http://bugzilla.kernel.org/show_bug.cgi?id=98921
Importance: Unknown
Status: Unknown
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[5.165709] [drm] Initialized i915 1.6.0 20141121 for :00:02.0 on minor 0
[5.190378] fbcon: inteldrmfb (fb0) is primary device
[5.190441] Console: switching to colour frame buffer device 170x48
[5.190462] i915 :00:02.0: fb0: inteldrmfb frame buffer device
[5.190464] i915
control.16 {
iface CARD
name 'Internal Mic Phantom Jack'
value true
comment {
access read
type BOOLEAN
count 1
}
}
control.17 {
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/patch_realtek.c?qt=grepq=gigabyte
all gigabyte motherboard use same pci system id, this mean that any pci
quirk of the driver apply to all realtek alc88x codesc
you need to ask the author to use an unique id (e.g.
it should be the responsibility of the browser (pulseaudio client) to
assign diffferent name if the application using pa_stream_new to connect
the the server multiple time
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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5) I suspend and resume the PC. The headphones stop working, and they
are gone from the sound settings GUI.
diff -u alsa_ifo_before_suspend alsa_info_after_resuume
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1) I disconnected my speakers from the rear jack, and plugged my
headphones into the front jack. After rebooting, the headphones are
detected and the sound works - everything is perfect. I opened the Sound
Settings dialog, and Headphones are present, Line Out is absent.
2) I plugged my speakers
!PCI Soundcards installed in the system
!!--
00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev 09)
00:1b.0 Audio device: Intel Corporation Lynx Point-LP HD Audio Controller (rev
04)
0.294826] pci :00:03.0: [8086:0a0c] type
** Package changed: linux (Ubuntu) = alsa-driver (Ubuntu)
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https://bugs.launchpad.net/bugs/1457476
Title:
Realtek ALC887-VD speakers don't mute when headphones are
post pulseaudio verbose log
https://wiki.ubuntu.com/PulseAudio/Log
are there any error message in pulseaudio verbose log
when the stream tag of analog convertor and digital convertor are same vlaue 8
this imply driver output signal to both analog and digital
the driver should return EBUSY
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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https://bugs.launchpad.net/bugs/1454828
Title:
PCI/internal sound card not detected
Status in
https://bugs.launchpad.net/ubuntu/+source/linux-lts-saucy/+bug/1288161
snd-hda-intel seem loaded in 14.04 LTS
This look like regression and you can compare bootdmesg and yours
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http://www.via.com.tw/en/products/audio/codecs/vt1718s/
2 independent 16/24-bit S/PDIF TX Outputs supporting 16/20/24-bit,
48K/96K/44.1K/88.2KHz sample rate
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APLAY
List of PLAYBACK Hardware Devices
card 0: MID [HDA Intel MID], device 0: VT1718S Analog [VT1718S Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 2: VT1718S Alt Analog [VT1718S Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
http://www.asrock.com/mb/Intel/H55M%20Pro/
there is optical spdif at the rear panel and internal HDMI SPDIF
connector
seem bios did not setup pin default correctly
Node 0x2e [Pin Complex] wcaps 0x400701: Stereo Digital
Pincap 0x0010: OUT
Pin Default 0x474411f0: [N/A] SPDIF Out at Ext
it is driver bug if all jack detect kctls return false
control.38 {
iface CARD
name 'Line Out Front Jack'
value false
comment {
access read
type BOOLEAN
count 1
http://git.alsa-project.org/?p=alsa-
tools.git;a=tree;f=hdajacksensetest;hb=HEAD
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https://bugs.launchpad.net/bugs/1453282
Title:
[Neither headphones nor speakers are
try ubuntu 14.10 or 14.04 LTS LiveCD and check whether those iommu oops
occur or not
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https://bugs.launchpad.net/bugs/1453282
Title:
[Neither headphones nor speakers
your system have many problems
[0.00] WARNING: CPU: 0 PID: 0 at
/build/buildd/linux-3.19.0/drivers/iommu/dmar.c:829
warn_invalid_dmar+0x7c/0x90()
[0.00] Your BIOS is broken; DMAR reported at address fed91000 returns
all ones!
BIOS vendor: American Megatrends Inc.; Ver: P2.20;
18.950222] ACPI Warning: SystemIO range 0x0828-0x082f
conflicts with OpRegion 0x0800-0x084f (\PMRG)
(20141107/utaddress-258)
[ 18.950228] ACPI: If an ACPI driver is available for this device, you should
use it instead of the native driver
[
diff -u alsa_info_before_suspend alsa_info_after_resume
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Title:
[Neither headphones nor speakers are displayed under All
Default sink name: alsa_output.pci-_00_1b.0.hdmi-stereo
Default source name:
alsa_input.usb-Auvitek_Auvitek_001-01-U0x5e10x400.iec958-stereo
** Changed in: pulseaudio (Ubuntu)
Status: Confirmed = Incomplete
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Node 0x1e [Audio Selector] wcaps 0x300501: Stereo
Power states: D0 D1 D2 D3
Power: setting=D0, actual=D0
Connection: 5
0x2b 0x2a 0x29 0x21 0x30*
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[4.671751] autoconfig: line_outs=1 (0x24/0x0/0x0/0x0/0x0) type:speaker
[4.671756]speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[4.671758]hp_outs=1 (0x25/0x0/0x0/0x0/0x0)
[4.671760]mono: mono_out=0x0
[4.671761]inputs:
[4.671768] Mic=0x30
Node 0x2d [Pin Complex] wcaps 0x400701: Stereo Digital
Control: name=HDMI Phantom Jack, index=0, device=0
Pincap 0x0010: OUT
Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Pin-ctls: 0x40: OUT
Power
Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out
Control: name=PCM Playback Volume, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name=PCM Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name=Line Out Jack, index=0,
seem connect to incorrect audio output
Node 0x0f [Pin Complex] wcaps 0x400181: Stereo
Control: name=Headphone Jack, index=0, device=0
Pincap 0x0014: OUT Detect
Pin Default 0x40f000f0: [N/A] Other at Ext N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
disable Auto mute mode mean driver don't mute the speaker when headphone
or line out is plugged
pulseaudio 's module-switch-on-available-port use alsa jack detection
kctls and playback switch to implement pulseaudio 's auto mute
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[ 15.459357] sound hdaudioC0D0: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0)
type:line
[ 15.459362] sound hdaudioC0D0:speaker_outs=1 (0xd/0x0/0x0/0x0/0x0)
[ 15.459364] sound hdaudioC0D0:hp_outs=1 (0xa/0x0/0x0/0x0/0x0)
[ 15.459365] sound hdaudioC0D0:mono: mono_out=0x0
[
/sys/class/sound/hwC0D0/init_pin_configs:
0x0a 0x2121101f
0x0b 0x03a1102e
0x0c 0x90a70120
0x0d 0x90170110
0x0e 0x21811040
0x0f 0x03211030
0x14 0x40f000f0
0x18 0x40f000f0
0x19 0x40f000f0
0x1e 0x40f000f0
0x1f 0x40f000f0
0x20 0x40f000f0
/sys/class/sound/hwC0D0/driver_pin_configs:
ay 02 19:56:26 hostname kernel: WARNING: CPU: 0 PID: 0 at
/build/buildd/linux-3.19.0/drivers/iommu/dmar.c:829
warn_invalid_dmar+0x7c/0x90()
May 02 19:56:26 hostname kernel: Your BIOS is broken; DMAR reported at address
0!
BIOS vendor: American Megatrends Inc.; Ver:
do you mean specify software_mute=N does not works as fallback?
you need to send email to the author of the patch, acpi-devel and alsa-
devel mailing list
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** Summary changed:
- [HDA-Intel - HDA Intel, playback] volume slider problem
+ [ThinkPad X60 - HDA Intel AD1981, playback] Volume Mute, Up, Down keys have
no effect
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!!Module: thinkpad_acpi
brightness_enable : 2
brightness_mode : 4
dbg_bluetoothemul : 0
dbg_uwbemul : 0
dbg_wlswemul : 0
dbg_wwanemul : 0
enable : Y
experimental : 0
fan_control : N
force_load : N
id :
9.559421] thinkpad_acpi: ThinkPad ACPI Extras v0.25
[9.559426] thinkpad_acpi: http://ibm-acpi.sf.net/
[9.559428] thinkpad_acpi: ThinkPad BIOS 7BETD7WW (2.18 ), EC 7BHT40WW-1.13
[9.559430] thinkpad_acpi: Lenovo ThinkPad X60, model 170686U
[9.562798] thinkpad_acpi: detected a 8-level
do you mean regression of these patches ?
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/thinkpad_helper.c?id=b317b032d2dcb5e518cc9630cc6f1c7c24afedfc
y 01 18:57:15 hostname pulseaudio[2071]: [pulseaudio] rtsp_client.c: failed to
connect to server 'fe80::ba27:ebff:febb:21e8:3'
May 01 18:57:15 hostname pulseaudio[2071]: [pulseaudio] module-raop-sink.c:
Failed to connect to server.
May 01 18:57:15 hostname pulseaudio[2071]: [pulseaudio]
20:19:13 david-XPS13-9333 pulseaudio[7977]: [pulseaudio] pid.c: Daemon already
running.
Jan 6 20:19:13 david-XPS13-9333 pulseaudio[7980]: [pulseaudio] pid.c: Daemon
already running.
Jan 6 20:21:48 david-XPS13-9333 pulseaudio[8578]: [pulseaudio] pid.c: Daemon
already running.
Jan 6 20:58:09
control.6 {
iface MIXER
name 'Capture Source'
value 'Headphone Mic'
comment {
access 'read write'
type ENUMERATED
count 1
item.0
seem missing this message in log
pa_log_debug( lfe filter activated (LR4 type)
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=979f19a434733afba0480e2ba45698362e05
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Soundcards recognised by ALSA
!!-
0 [HDMI_1 ]: HDA-Intel - HDA Intel HDMI
HDA Intel HDMI at 0xd083 irq 31
1 [PCH]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xd0834000 irq 28
2 [HDMI ]:
do you mean hp is node 0x1a instead of node 0x15 ?
you also need to change pin default of node 0x15 to [N/A]
Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Control: name=Headphone Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control:
control.44 {
iface CARD
name 'Line Out Front Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
control.45 {
pr 29 21:58:00 david-XPS13-9333 pulseaudio[2211]: [pulseaudio] pid.c: Daemon
already running.
Apr 29 22:04:37 david-XPS13-9333 dbus[736]: [system] Rejected send message, 10
matched rules; type=error, sender=:1.106 (uid=112 pid=2168
comm=/usr/bin/pulseaudio --start --log-target=syslog )
device.profile.name = analog-stereo
device.profile.description = Analog Stereo
device.description = Built-in Audio Analog Stereo
alsa.mixer_name = Realtek ALC882
alsa.components = HDA:10ec0882,1043e601,00100101
Codec: Realtek ALC3661
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0668
Subsystem Id: 0x1028060a
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://bugs.freedesktop.org/show_bug.cgi?id=84983
** Bug watch added: freedesktop.org Bugzilla #84983
https://bugs.freedesktop.org/show_bug.cgi?id=84983
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I just tried Ubuntu 15.04 which embeeds pulse audio 6.0.
In the sound settings panel, I can switch to 2.1 profile, and within test
panel, everything works just fine : Left, right and subwoofer.
*But* when I launch any sound application, there is no sound through
subwoofer.
** Summary changed:
- Stale PID file, overwriting.
+ main.c: Daemon startup without any loaded modules, refusing to work.
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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post the output
lsusb -
it is strange that there is no device zero
have you try device 1 ?
ARECORD
List of CAPTURE Hardware Devices
card 1: Ed [Xonar U7 Echelon Ed.], device 1: USB Audio [USB Audio #1]
Subdevices: 1/1
Subdevice #0: subdevice #0
Simple mixer control
you have to follow up with the alsa developer as he require event trace
log
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https://bugs.launchpad.net/bugs/1445596
Title:
lenovo t410 [2537BF9, Conexant CX20585,
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
Headset Mic Phantom Jack - indicates headset jack where hardware can not
distinguish between headphones and headsets
Headset Mic Jack -
no popup because bios setup internal Mic only
[7.890594] sound hdaudioC0D0: autoconfig: line_outs=1
(0x14/0x0/0x0/0x0/0x0) type:speaker
[7.890601] sound hdaudioC0D0:speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[7.890605] sound hdaudioC0D0:hp_outs=1 (0x21/0x0/0x0/0x0/0x0)
[
[ 3504.156337] ALSA: PCM: [Q] Lost interrupts?: (stream=0, delta=72,
new_hw_ptr=1489, old_hw_ptr=1417)
do you still get those ALSA: PCM: [Q] Lost interrupts?: when you specify
align_buffer_size=1 ?
MODULE_PARM_DESC(align_buffer_size,
Force buffer and period sizes to be
(In reply to rkfg from comment #35)
When I start PA without tsched=0, it's ALSA setup is weird:
Apr 10 09:46:52 work pulseaudio[5161]: [pulseaudio] alsa-util.c: rate
: 44100
Apr 10 09:46:52 work pulseaudio[5161]: [pulseaudio] alsa-util.c: exact
rate : 44100 (44100/1)
Apr 10
pr 14 13:51:06 work pulseaudio[11059]: [pulseaudio] alsa-util.c: buffer_size
: 192
Apr 14 13:51:06 work pulseaudio[11059]: [pulseaudio] alsa-util.c: period_size
: 32
Apr 14 13:51:06 work pulseaudio[11059]: [pulseaudio] alsa-util.c: period_time
: 725
the minimum period size of stereo
https://www.google.com/patents/US20100131783
FIG. 2 illustrates an exemplary FIFO which has a FIFO size of 192 bytes, and a
threshold value of 128 bytes.
Taking 48 kHz sample rate, 2 channels each having 16 bits (or 2 bytes)
for example, each frame thus contains 4 bytes of data, wherein each
Also, after I do:
echo 1 /proc/asound/card0/pcm0p/xrun_debug
1 is not enough
http://www.alsa-project.org/main/index.php/XRUN_Debug
you need 11 to find out whether hardware pointer are correct on period
update and how often lost interrupt occur compared with period update ?
# Enable basic
if the result if alexander pcm_avail.c show available increment by
period size
this means you need to use more than two fragments and increase
rewind_safeguard
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=4df443bbe682055a41e7c2248877dcc7682a69b8
16 19:16:17 homecomp
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/ca0106/ca0106_main.c
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) 0)
return err;
if ((err = snd_pcm_hw_constraint_step(runtime, 0,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa/Procfile.txt
http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-November/083495.html
you need to ask the author how to debug your case Lost Interrupt!
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Enable basic debugging, do jiffies check and dump position on each period and
hardware pointer update calls
# Usefull when the lowlevel (specific) hardware driver is somehow broken
echo 29 /proc/asound/card0/pcm0p/xrun_debug
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-September/081501.html
you need to modify Alexander's pcm_avail.c to find out how accurate
hwptr position can be reported by your ca0106 hw device since his
program hardcode to use 44100Hz which is not supported by your ca0106
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You
as pulseaudio use snd_pcm_rewind() which require the sound card report
accurate hwptr position than those sound card which report hwptr
position at period bounary when interrupt occur
the difference between hda-intel and your ca0106 is the granularity of the
pointer callback
hardware pointer should be increased by one period on every
period_update, the driver report lost interrupt when the elasped time is
more than 1.5 period time
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/core/pcm_lib.c
no_jiffies_check:
if (delta
seem dump stack is not useful
just need 1+4+8 = 13
1 Basic debugging - show xruns in ksyslog interface
2 Dump stack - dump stack for basic debugging
4 Jiffies check - compare the position with kernel jiffies (a sort of
in-kernel monotonic clock),
show what's changed when basic
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=73bdd597823e2231dc882577dbbaf8df92fe1775
seem different model
1) combo jack which support headset, headphone and mic
2) headset and dock mic
3) headset mode
4) headset mode no hp mic
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda?qt=grepq=alc283
still no acer aspire using alc283
do dell multi headset multi allow you to select hp mic ?
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51.449| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms
( 51.449| 0.000) I: [pulseaudio] alsa-util.c:
snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
( 51.449| 0.000) I: [pulseaudio] alsa-util.c:
snd_pcm_hw_params_set_period_size_near() failed: Invalid
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=732348c9d9d97b0820caca0c7aad67a2fd83c8dd
do you mean this patch still not fix your problem ?
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http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grepq=lfe
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https://bugs.launchpad.net/bugs/1445861
Title:
Subwoofer drops out when playback ends
Status
does the user manual mention that it also support headphone ?
Dell headset multi is used for those combo jack which support headset,
headphone and coventional Mic jack
if headset Mic work , this mean you have to remove the headphone Mic pin
fixup from Dell fixup
and let driver switch to headset
as you are not using 48000Hz 6 channels
it depend on your HDA controller don't has 128 bytes alignment in period
bytes restriction
( 2.009| 0.000) I: [pulseaudio] sink.c: device.vendor.id = 8086
( 2.009| 0.000) I: [pulseaudio] sink.c: device.vendor.name = Intel
Corporation
(
51.446| 2.302) D: [pulseaudio] module-suspend-on-idle.c: Sink
alsa_output.pci-_00_1b.0.analog-stereo becomes idle, timeout in 5 seconds.
( 51.446| 0.000) D: [alsa-sink-ALC1150 Analog] sink-input.c: Have to rewind
15840 bytes on render memblockq.
( 51.446| 0.000) D: [alsa-sink-ALC1150
you were recording at 64.50dB
Simple mixer control 'Front Mic Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 3 [100%] [30.00dB]
Front Right: 3 [100%] [30.00dB]
Simple mixer control
seem your BENQ display are connected DVI-I-1 connected primary
not sure why default sink change to non available HDMI when you have
both USB audio and onboard analog device available
https://wiki.ubuntu.com/PulseAudio/Log
did you customise default.pa to select any specified sink ?
control.37
https://lkml.org/lkml/2013/1/13/11
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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https://bugs.launchpad.net/bugs/1445596
Title:
lenovo t410
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