Sound.git is unstable and not gauarnteed to compile at any time
The safest way is to compile debug kernel/driver at stable ubuntu
release
Tiwai expect you to apply the patch and report back whether it has no side
effect
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/hda_codec.c?id=4b3acaf5b56f53ef259a6ddf8e17dcb529631d9b
It seem that you are using copy front when channel tag of two audio
output are same
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Converter: stream=1, channel=0
The other difference is channel, seem still using two channels when
channels tag of two audio output are same
speaker-test -c4 -t wav -D hw:0,0
only need two channel if there is low pass filter with mono pin complex
and driver don't support 4 channels
For
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=f5914908a5b7b2338f210e56827a1ef35585dc6d
To debug the ptr position at period interrupt, you need alsa driver
build with debug which allow you to enable trace point to log the ptr
position in system
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In theory, you will like to have
Speaker volume and Subwoofer volume
Headphone share with one of them
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Title:
Lenovo Y700-17ISK subwoofer
You better post the outout of alsa-info.sh
Since you need to check the amp out and mute of the path from audio
output, audio mixer and pin complex
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Only one pin complex can be retasked at each test since there are only
two audio output/volume control but there are three output pin
complexes, this mean one of the volume control must be shared by two pin
complex
The pin-ctls, power setting of the subwoofer should be same as the
speaker node
The difference between 0x17 mono and 0x1b stereo pin complex
You need to find out whether there is low pass filter when you get sound
from subwoofer and whether both channel affect the subwoofer if the
subwoofer is at the stereo pin complex 0x1b
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ice1712/ice1712.c?id=4f8e940095536bc002a81666a4107a581c84e9b9
The fix for the other driver in the original bug report
It look like au88x0 pointer callback does not return zero when interrupt occur,
just 2 or 3 frames
dmesg | grep -i "au88"
Since pcm_avail.c only write buffer size once, it is unable to trigger
the bug which ptr > buffer size
so we need to modify the logic to loop more data
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Do the number of BUG messsges in system log still 2 frames larger than
buffer size
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer
Can you change
periods from 4 to 16
and period_size from 1024 to 256
post the output of pcm_avail.c
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Title:
snd_au8820 :01:04.0: BUG: ,
Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps:
ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00]
Connection: 1 0x0d
This seem to be stereo to mono down mix widget since node 0x0d is stereo
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Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right Limits: Playback 0 - 87
Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback
0 [0%] [-65.25dB] [off]
Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch
The other output capable pin is node 0x1b, do the driver connected this
node to node 0x02 or 0x03?
Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In
caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00
0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00,
Node 0x0f is connected to 0x03 through 0x0d
You have to check whether those amp out and mute switfch of the path
from node 3 to node 0x17 is not muted by pulseaudio (e.g. headphone
plasyback volume )
Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker
Playback Volume",
s. 31 16:05:42 hostname pulseaudio[1728]: [alsa-source-USB Audio] alsa-
source.c: ALSA vekket oss for å lese nye data fra enheten, men vi fant
ingen data! des. 31 16:05:42 hostname pulseaudio[1728]: [alsa-source-USB
Audio] alsa-source.c: Dette er høyst sannsynligvis en programfeil i
ALSA-driveren
As there is no datasheet, we can only follow the patent description
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0/au8820.h#n26
FIG. 15c is a diagrammatic block diagram of a mode register 382 of sub-
buffer table 380(3) of FIG. 15b. Mode register 382 includes
Only two of the remain unconnected pins pin-cap support OUT
.0x17 and 0x1b
You can use hdajackretask set one of them as subwoofer, reboot
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https://bugs.freedesktop.org/show_bug.cgi?id=84585#c60
Modified version
You can compare the result
** Bug watch added: freedesktop.org Bugzilla #84585
https://bugs.freedesktop.org/show_bug.cgi?id=84585
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Time diffenece between successive output and frames difference will tell
you whether the card is accurate enoung for pulseaudio timer scheduling
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Can you post the output of pcm_avail.c when using rate 44100 Hz instead
of 48000Hz?
print the time at the same time to check whether the ptr is monotically
increasing as expected rate
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=0ab1ace856205d10cbc1924b2d931c01ffd216a6
This patch only prevent kerneloops pop up, your system log still
flooding with invalid position message every 0.371 second (assume once
per buffer time) when playing audio
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2013-February/059559.html
You need to install many packages in order to build alsa driver
git, autoconf, kernel-header,
You need to rebuild the alsa driver whenever kernel change
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no need to perform git bisect since you already know the reason of
kernel ops is due to "BUG:"
Do ubuntu allow user just build alsa driver as module instead of
recompile the whole kernel
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-build.git/
The oscillator is the silver metal component just above the au8820 chip
of your montego card
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Title:
snd_au8820 :01:04.0: BUG: , pos =
You may need to dump mmio VORTEX_SMP_TIME which is tick at 48000 * 256
Hz (what is the clock printed at the oscillator on your au8820 sound
card) to find out the elapsed time between periods is correct or not
- return (bytes_to_frames(substream->runtime, current_ptr));
+ current_ptr =
the differnce between successive output are almost 5 frames
The only bad thing
ptr is not always at period boundary 1024, 2048, 3072 and 4096
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Card numer can be specified by module parameter "index"
Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj :
32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1
beep_mode :
N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable :
Run two alsa applications which using hw device, hwmixvolume allow you
to change the subdevice PCM volume controls which dB range allow gain in
addition to attenuation
For any pulse application, it just control by the subdevice opened by
pulseaudio server
The hardware mixed signal are pass
Each subdevice has its own volume control
control.59 { iface PCM name 'PCM Playback Volume' value.0 0 value.1 0
comment { access 'read write inactive' type INTEGER count 2 range '-128
- 32' dbmin -9600 dbmax 2400 dbvalue.0 0 dbvalue.1 0 } }
control.60 { iface PCM subdevice 1 name 'PCM Playback
There will be volume slider(hardware per stream volume control) for each
application if the application use hw instead of pulse
pavucontrol provide sofware per stream volume control
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His program hardcode to use card 0, you have to change the card number
if ypur au8820 is card 1
or change the order of your two cards by adding specify card number for
the two drivers in /etc/modprobe.d/alsa-base.conf
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You have to send email to the author and commiter since flooding the
system log with message is useless
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Title:
snd_au8820 :01:04.0: BUG: ,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=00d90154404ae6218730068d25bf2faad3186631
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Title:
Are you the kernel oops is related to au8820 since it just flooding the
system log with BUGS message but no seg fault or dump
it is strange that your bug report did not have pulselist or pulseaudio
log
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Did his program use #include similar to aplay.c?
http://git.alsa-project.org/?p=alsa-
utils.git;a=blob;f=aplay/aplay.c;hb=HEAD
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Title:
http://packages.ubuntu.com/search?lang=en=libasound2-dev
For pcm_avail.c
You need to install libasound2-dev and gcc
sudo apt-get install libasound2-dev
gcc pcm_avail.c -o pcm_avail -l asound
./pcm_avail
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Can you post output for your au8820 pci ssid 12eb:0001
do it support D3?
lspci -
[ 0.244875] pci :01:04.0: [12eb:0001] type 00 class 0x040100
[ 0.244898] pci :01:04.0: reg 0x10: [mem 0xfb10-0xfb11]
[ 0.244908] pci :01:04.0: reg 0x14: [io 0x2000-0x2007]
[ 0.244917] pci
[11.080876] snd_au8820 :01:04.0: init started snd_au8820 :01:04.0:
init done.
[ 11.837183] gameport gameport0: AU88x0 Gameport is pci:01:04.0/gameport0,
speed 1087kHz
in theory, you should look for au8820 in system log since those dev_info
also print the dev name
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-September/081501.html
you can try Alexander's pcm_avail.c and post the output using your
au8820
It is rather strange that only 2 frames since dma transfer in brust
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dmesg is the command which show the system log similar to
https://launchpadlibrarian.net/243848211/CurrentDmesg.txt
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Title:
snd_au8820
You have to recompile alsa driver by adding
if (current_ptr > runtime->buffer_size)
current_ptr = runtime->buffer_size;
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Title:
snd_au8820
The dirty hack is to ensure current_ptr is less than buffer size in the
above routine
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0/au88x0_pcm.c
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Title:
snd_au8820 :01:04.0: BUG: , pos =
/* pointer callback */
static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream
*substream)
{
vortex_t *chip = snd_pcm_substream_chip(substream);
stream_t *stream = (stream_t *) substream->runtime->private_data;
int dma = stream->dma;
snd_pcm_uframes_t current_ptr = 0;
System log
dmesg
Look for "virt" or "delta" when you use aplay with small period size,
the sound card has four subbuffer for each playback stream
http://www.google.com/patents/US6167465
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The wavetable is not implememted since dma since not start for au8830
card 1: au8820 [Aureal Vortex au8820], device 3: Aureal Vortex WT
[au8820 wt] Subdevices: 32/32 Subdevice #0: subdevice #0 Subdevice #1:
subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3
Subdevice #4: subdevice
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0/au88x0_core.c
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Title:
snd_au8820 :01:04.0: BUG: , pos =
dev_info(vortex->card->dev, "%d virt=%d, real=%d, delta=%d\n", adbdma,
dma->period_virt, dma->period_real, delta);
Do you find the above info in system log when you specify small fragment
--period-size=16 when using aplay
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How often do the message appear in system log? (BUG 16.04 or XRUN before
14.04)
it mean the value return is larger than buffer size or the interrupt is
not handled properly
static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int
adbdma) { stream_t *dma = >dma_adb[adbdma]; int temp,
http://git.alsa-project.org/?p=alsa-
tools.git;a=blob;f=hwmixvolume/README;hb=HEAD
You can use hwmixvolume to change the playback volume of each
application which use hw device of au8820
It allow up to 8 stereo streams or 16 mono streams
card 1: au8820 [Aureal Vortex au8820], device 0:
Should use au8820 instead of AU8820 when using aplay and speaker-test
Do device 3 of au8820 work or not? never work on au8830
hw:CARD=au8820,3
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Do the message appear in every application or occur everytime when
playing sound?
aplay -D hw:CARD=AU8820 anystereo.wav
speaker-test -c2 -t wav -D hw:CARD=AU8820
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core?id=d507941beb1ef98c19e2902007aee4faf36f854f
ALSA: pcm: Correct PCM BUG error message
While converting to dev_*(), the message showing the invalid PCM
position was wrongly tagged as if an XRUN although it's actually a
It is not related to the above two patch if your au8820 still work
without the error on ubuntu 14.4 live cd
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Title:
snd_au8820 :01:04.0:
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/au88x0
Your problem is pointer callback return a value which is larger than buffer
size
If this is regression , you may need to perform git bisect to find out
the related patch
or
revert this patches
** Package changed: linux (Ubuntu) => alsa-driver (Ubuntu)
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size = 16384,
period
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size = 16384, period
size = 1024 snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size =
16384, period size = 1024
Any steps to reprodcue these message
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/au88x0
This
introduced in 2.16 that caused spurious failures under Python 2.
Error messages for command and function syntax parse failures have improved.
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Public bug reported:
A "remote magtape protocol module? On PCs? In 2016?
The are potential users for this thing, maybe, on Z-series mainframes,
but having it be autoinstalled on vanilla desktop machines, or even
ordinary servers, is deeply silly.
Recommendation: split the package. Yes, tar is
[ 3.212954] NVRM: corruption and stability problems, and is not
supported. [ 3.307199] snd_hda_codec_hdmi hdaudioC1D0: HDMI: invalid ELD
data byte 4
Your HDMI are plugged too
control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value true comment {
access read type BOOLEAN count 1 } } control.8 {
*** This bug is a duplicate of bug 1539209 ***
https://bugs.launchpad.net/bugs/1539209
mrt 06 20:17:03 username-EasyNote-TJ77 pulseaudio[2191]: [alsa-source-
ALC272X Analog] alsa-mixer.c: Assertion
'pa_cvolume_compatible_with_channel_map(v, cm)' failed at modules/alsa
/alsa-mixer.c:1222,
Pin-ctls: 0x00: --- >
Pin-ctls: 0x40: OUT
Driver mute speaker pin complex by set pin ctls to 0 instead of OUT
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Title:
No sound from internal
** Package changed: linux (Ubuntu) => alsa-driver (Ubuntu)
** Changed in: alsa-driver (Ubuntu)
Status: Confirmed => Incomplete
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Title:
Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps:
ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00]
Connection: 1 0x0d
Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps:
ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap
Take a diff of output of alsa-info.sh when work and non working
Which node is headphone or dock headphone
Strange that eapd(external amplifier power down) is enabled
Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker
Playback Switch", index=0, device=0 ControlAmp:
Did not find any patch related to your hda codec
?
https://wiki.ubuntu.com/TrustyTahr/ReleaseNotes/ChangeSummary/14.04.1
https://wiki.ubuntu.com/TrustyTahr/ReleaseNotes/ChangeSummary/14.04.2
https://wiki.ubuntu.com/TrustyTahr/ReleaseNotes/ChangeSummary/14.04.3
Linux 3.19.0-47-generic x86_64
Seem different from 16.04.4 kernel 4.x
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Title:
[Intel DP35DP, SigmaTel STAC9271D, Green Speaker, Rear]
https://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa?id=527078523815587bfd021bc1d10782403e9b3e84
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Title:
No sound on ASUS
by passing `jack_detect = no` string
via sysfs or a patch file, you can disable the jack detection, thus
the codec parser will skip the features like auto-mute or mic
auto-switch.
The driver create headphone phantom jack and lineout phantom jack
instead of headphone jack and lineout jack
Codec: Conexant CX20751/2 Address: 0 AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x14f1510f Subsystem Id: 0x17aa3805 Revision Id: 0x100100 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,
You have to file upstream bug report
https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers
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Title:
[80E3, Conexant CX20751/2, Speaker,
If your acer aspire 7540 have line in, mic and internal mic, do alsa
driver disable auto mic selection by create capture source control for
selecting input
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*** This bug is a duplicate of bug 1539209 ***
https://bugs.launchpad.net/bugs/1539209
It seem bug of virtual box 5.0 in emulated three ac97 codecs on ich
80281ab controller which does not support multi channel /codecs
0-0/0: Analog Devices AD1980 PCI Subsys Vendor: 0x1028 PCI Subsys
*** This bug is a duplicate of bug 1539209 ***
https://bugs.launchpad.net/bugs/1539209
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1065812
The emulated ac97 codec was used in dell dimension 8400
0-0/0: Analog Devices AD1980 PCI Subsys Vendor: 0x1028 PCI Subsys Device: 0x0177
Usb audio still not implemented ice958 control to set non audio aes bits
for ac3/dts passthrough
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Title:
Asus Xonar U7 S/PDIF output not
ports: analog-input-front-mic: Front Microphone (priority 8500, latency
offset 0 usec, available: no) properties: device.icon_name = "audio-
input-microphone" analog-input-rear-mic: Rear Microphone (priority 8200,
latency offset 0 usec, available: no) properties: device.icon_name =
Active Profile: output:analog-stereo+input:iec958-stereo
Do you really change since capture device 1 not exist?
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Title:
Asus Xonar U7 S/PDIF
card 1: U7 [Xonar U7], device 0: USB Audio [USB Audio] Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: U7 [Xonar U7], device 1: USB Audio [USB Audio #1] Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: U7 [Xonar U7], device 2: USB Audio [USB Audio #2] Subdevices: 1/1
Subdevice #0:
hda_analyzer - a tool to analyze HDA codecs widgets and connections
Usage: hda_analyzer [[codec_proc] ...]
or: hda_analyzer --monitor
codec_proc might specify multiple codec files per card:
codec_proc_file1+codec_proc_file2
or codec_proc might be a dump from
You have to specify which codec proc
/proc/asound/card0/codec#0
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Title:
[Intel DP35DP, SigmaTel STAC9271D, Green Speaker, Rear] Underruns,
[ 3.216843] pci :01:00.0: can't claim BAR 6 [mem
0xfffe-0x pref]: no compatible bridge window
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Title:
[HP ProBook 450 G3, Intel
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/oxygen?qt=grep=dg%28x%29
** Changed in: pulseaudio (Ubuntu)
Status: Confirmed => Incomplete
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http://www.intel.com/content/www/us/en/support/boards-and-kits/desktop-
boards/05512.html
Check your computer chassis front audio panel
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https://bugs.launchpad.net/bugs/1540372
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_sigmatel.c?id=eb63212868c348cc6d3ec6929d7d98f7d29493e9
ALSA: hda - Power up always when no jack detection is available
When no jack detection is available, the pins should be always
turned on since it can't
You have to find out your onboard audio still work with jack detection
and analog back at which version of alsa driver/kernel by boot from live
cd
in theory, you lost the automute feature when jack detection of hp is
not functioning but the aggressive power saving use jack detection to
power down
As your pin defaults are not correct
[ 17.585436] sound hdaudioC0D2: ignore pin 0x11 with mismatching assoc#
0xf vs 0x6
It only work when you were using static config i.e. before using
generic parser patch
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It is not in 16.04
http://bazaar.launchpad.net/~ubuntu-branches/ubuntu/wily/alsa-
lib/wily/files/head:/src/
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https://bugs.launchpad.net/bugs/1498129
Title:
Asus Xonar U7 S/PDIF output
It only work with latest alsa lib
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https://bugs.launchpad.net/bugs/1498129
Title:
Asus Xonar U7 S/PDIF output not recognized by pulseaudio
To manage notifications about this bug go
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=84f98fdf7865fbd35b312eb39ea91e5618c514c7
Do you mean headset mic not automatically switch when there are
headphone and dock headphone?
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-8.
Handle .de co and .de au in NAME sections; the groff suite needs this.
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<e...@thyrsus.com>
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Why don't you use Vendor and product id?
usb 1-1: New USB device found, idVendor=1043, idProduct=857c
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0
usb 1-1: Product: Xonar U7
usb 1-1: Manufacturer: ASUS
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Your hda controller 0x10044384 is not yet supported, you need to add and
recompile alsa dtiver
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/hda/hda_intel.c
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The codec should return 0x8000 when jack is plugged
Try execute SET_PIN_SENSE before GET_PIN_SENSE using hda verb for those
pin complex. Node
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https://bugs.launchpad.net/bugs/1540372
https://code.launchpad.net/ubuntu/+source/alsa-driver
The last modified date of trusty was 2013
Do you mean upgrade or update?
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https://bugs.launchpad.net/bugs/1540372
Title:
[Intel
You need to file upstream bug report since no io/mmio was assigned to
two hda controllers
[ 0.190122] pci :00:14.2: reg 0x10: [mem 0xfe024000-0xfe027fff 64bit] [
0.190192] pci :00:14.2: PME# supported from D0 D3hot D3cold [ 0.190229] pci
:00:14.2: System wakeup disabled by ACPI [
You have to send email to the author
define stac92xx_clfe_switch_info snd_ctl_boolean_mono_info
-
-static int stac92xx_clfe_switch_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_sigmatel.c?id=36c9db7a1aca3396b26004a3c41f27e2b735536f
Some idt codec specific feature were lost when using generic parser
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Some characters are unsafe
http://git.alsa-project.org/?p=alsa-lib.git;a=patch;h=4dc44bb34aab2b23ab45c8e61e4b17bf2cf58959
You have to send email to alsa devel mailing list how to add your sound card
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