Public bug reported:
The Ubuntu 16.04.1 LTS upgrade from kernel 4.4.0-21 to 4.4.0-34
introduced the following periodic error message sequence reported in
dmesg/syslog:
...
sd 4:0:0:0: [sdc] tag#0 FAILED Result: hostbyte=DID_ERROR
driverbyte=DRIVER_SENSE
sd 4:0:0:0: [sdc] tag#0 Sense Key : Hardwa
Joseph, we just verified that this bug manifests itself in the current
upstream kernel (4.4.19). I added the tag "kernel-bug-exists-upstream"
to the report. Thanks!
** Tags added: kernel-bug-exists-upstream
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I can't find an edit button; the command I quoted above is incorrect. The
corrected one is:
rrogers@rrogers-desktop:~/tmp$ apt-cache policy libgl1-mesa-dr1*
libgl1-mesa-dri-lts-utopic-dbg:
Installed: (none)
Candidate: 10.3.2-0ubuntu1~trusty2
Version table:
10.3.2-0ubuntu1~trusty2 0
Public bug reported:
This resulted from trying to install libgli from
http://packages.ubuntu.com/trusty/i386/libgl1-mesa-dri/download
Presumably, it's the wrong package?
Following the instructions below:
ubuntu-bug mesa
failed reporting that it wasn't installed. Well that was because the
insta
Seem similar oops
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1561449
8] WARNING: CPU: 1 PID: 313 at /build/linux-
lAMkDx/linux-4.4.0/drivers/i2c/busses/i2c-designware-baytrail.c:112
baytrail_i2c_acquire+0x139/0x1e0 [i2c_designware_platform]() [ 4.811122]
Modules linked in: mac8021
[ 4.588798] WARNING: CPU: 3 PID: 272 at
/build/linux-_Kv5oI/linux-4.2.0/drivers/i2c/busses/i2c-designware-baytrail.c:112
baytrail_i2c_acquire+0x139/0x1e0 [i2c_designware_platform]()
[ 4.588801] Modules linked in: v4l2_common(+) videodev btusb btrtl snd_pcm
media btbcm btintel bluetooth snd_seq_
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=d08c5ef2a039393eaf2ab2152db5f07790fa0f40
You have to retask line out to speaker if node 0x0f is speaker
Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x00
!!Advanced information - PCI Vendor/Device/Subsystem ID's
!!---
00:1b.0 0403: 8086:2668 (rev 03)
Subsystem: 104d:81cd
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snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC260: line_outs=1
(0x10/0x0/0x0/0x0/0x0) type:hp
snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0)
snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0
sn
Seem same pci ssid
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=5e1b1518a53fc62d9f39a13819c849336c6d8dd4
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=c3c2c9e7ff3e38bd9ff5b721b6ae8634fce42802
Alc260 basic model was removed, seem use 0x15 as line out and use mono
pin
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- /* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* enable HP */
-
You have to ask the reporter of VGN-A790 laptop which has same pci ssid
104d:81cd as your VGN-A497XP
http://git.alsa-project.org/?p=alsa-
tools.git;a=blob;f=hdajackretask/README;hb=HEAD
You need to find out whether line out node is internal speaker and which
mode is your mic and internal mic
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/hda/alc260_quirks.c?id=c29b3f6dd7798964d77199af4925be72a3a48349
This is a regression of removing static model of alc260, you have to
file upstream bug report at bugzilla.kernel.org
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aplay -Dhw:0 --dump-hw-params anystereo.wav
should show channel max is 4 when you need software low pass filter but
only 2 wheb your laptop have hardware low pass filter
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How did you test ?
When there is hardware low pass filter and downmix widget,
Converter: stream=1, channel=0
Subwoofer channel. Is same as speaker
When there is no hardware low pass filter, channel=1
and you need pulseaudio to provide software lowpass filter and downmixing
Node 0x02 [Audio
Which sst platform are you using?
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/soc/intel/Kconfig
** Changed in: alsa-driver (Ubuntu)
Status: Confirmed => Incomplete
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( 5.156| 0.000) I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH
flag is set
( 5.156| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 1981 ms
( 5.158| 0.001) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE
failed (-32)
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You have to post output of alsa-info.sh
Which card is your default sink?
Seem fail with surround21 but does your card suppport multichannel?
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_conexant.c?id=6ffc0898b29a2811a6c0569c5dd9b581980110df
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Tit
You have to post result of hdajacksensetest when you plug and unplug
headphone jack and mic jack
http://git.alsa-project.org/?p=alsa-
tools.git;a=tree;f=hdajacksensetest;hb=HEAD
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https://
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/soc/intel?id=bd01fdc3aa63b7ba0b035f9196d80551ad03f5d4
But there is no 0x808622a8 in your system log
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The auto parser seem ignore the line out pin when default association is
zero
Default 0x01014000: [Jack] Line Out at Ext Rear
Conn = 1/8, Color = Green
DefAssociation = 0x0, Sequence = 0x0
/sys/class/sound/hwC0D0/init_pin_configs:
0x0f 0x01014000
0x10 0x02214000
0x11 0x50171000
0x12 0x0
Check your system log, seem kernel oops
- [26825.955291] WARNING: CPU: 0 PID: 16771 at
/build/linux-_Kv5oI/linux-4.2.0/fs/block_dev.c:57 __blkdev_put+0xbd/0x280()
[26825.955293] Modules linked in: nls_iso8859_1 uas usb_storage binfmt_misc
ax88179_178a usbnet uvcvideo videobuf2_vmalloc videobuf
| 0.000) W: [pulseaudio] alsa-mixer.c: Volume element Speaker has 8
channels. That's too much! I can't handle that! ( 5.131| 0.000) D:
[pulseaudio] alsa-mixer.c: Probe of element 'Speaker' failed.
Pulseaudio not support volume control with more than two channels
Card hw:4 'Device'/'USB Sound Devi
5.039| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:5 ( 5.039|
0.000) I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH
flag is set ( 5.039| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw
buffer size is 11888 ms ( 5.041| 0.001) D: [pulseaudio] alsa-util.c: Set
buffer size fir
If your card 2 only support stereo, it is strange that surround21 can be
opened
155| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on
Analog Surround 2.1 (analog-surround-21) ( 5.155| 0.000) D: [pulseaudio]
alsa-util.c: Trying surround21:2 with SND_PCM_NO_AUTO_FORMAT ... (
5.156| 0.00
The major problem is why you override pcm.default with dmix whenn you
running pulseaudio
!!User specific config file (~/.asoundrc)
pcm.!default { type plug slave { pcm "dmix:Device" } }
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!
The driver sort the speaker pin using sequence number, the subwoofer should has
sequence number higher than the internal speaker
!--- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x90a60140
0x14 0x90170110
0x17 0x4000
0x18 0x41f0
0x19 0x04a11030
0x1a 0x41f0
0x1b 0x41
You have to post output of alssa-info.sh
pactl list sinks
xrandr --verbose
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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https://bugs.launchpad.net/bugs/1563917
https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/7.0/
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Title:
Lenovo Y700-17ISK subwoofer doesn't work
To manage notifications about t
The objective of using early patching or hdajackretask is fixup the missing
subwoofer so that is appear as the second entry after the node 0x14
snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC233: line_outs=1
(0x14/0x0/0x0/0x0/0x0) type:speaker
snd_hda_codec_realtek hdaudioC0D0: speaker_ou
Can the other confirm that 0x17 is the subwoofer imstead of 0x1b?
This mean you can mute can change the volume of subwoofer by using
alsamixer
Do you hear high frequency from the subwoofer when playing stereo or 4
channel playback?
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Try specify model=dell-headset-multi
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/HD-Audio-Models.txt
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/HD-Audio-DP-MST-audio.txt
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Title:
change in monitor r
[ 18.227418] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC668:
line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker
[ 18.227421] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=0
(0x0/0x0/0x0/0x0/0x0)
[ 18.227422] snd_hda_codec_realtek hdaudioC1D0: hp_outs=1
(0x15/0x0/0x0/0x0/0x0)
[ 18.227
Can you post the output of alsa-info.sh when specify model ?
The auto mic selection is not enabled, you have to manally selected
headset mic as capture source
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/sys/class/sound/hwC0D0/init_pin_configs: 0x0f 0x02014110 0x10
0x21011120 0x11 0x41f0 0x12 0x02a15910 0x13 0x41f0 0x14
0x21845120 0x15 0x41f0 0x16 0x41f0 0x17 0x99831140 0x18
0x21451130 0x19 0x41f0
/sys/class/sound/hwC0D0/driver_pin_configs: 0x0f 0x01211020 0x10
0x0001003f 0x11
Both headphone and line out are not available
active profile: sinks:
alsa_output.pci-_00_1b.0.analog-stereo/#0: Built-in Audio Analog
Stereo sources: alsa_output.pci-_00_1b.0.analog-stereo.monitor/#0:
Monitor of Built-in Audio Analog Stereo alsa_input.pci-_00_1b.0
.analog-stereo/#1: B
Your original pin defaults are line out and dock line out but changed to hp and
line out
Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control:
name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3,
dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00,
stepsize=0x00,
control.14 { iface CARD name 'Line Out Jack' value false comment { access read
type BOOLEAN count 1 } }
control.15 { iface CARD name 'Headphone Jack' value false comment { access read
type BOOLEAN count 1 } }
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You have to file upstream bug report at bugzilla.kernel.org
-110 connection time out
-71 protocol error
-19 no device
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Title:
[USB-Audio - VX2,
Do you need the fixup of the external subwoofer?
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/patch_realtek.c?qt=grep&q=asus+n
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** Also affects: alsa-driver via
http://bugzilla.kernel.org/show_bug.cgi?id=115481
Importance: Unknown
Status: Unknown
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Title:
No so
This is the drawback of using dell-headset-multi which allow combo jack
to support headphone, headset and microphone
You can try the patch in alsa devel mailing list archive for asus
n750jk which support headset only
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As the codec cannot differentitate headphone, headset and microphone,
you need to switch capture source to determine the jack type
it is just a simpifed dell-headset-multi by removing the support of
headphone and microphone jack, assuming user only use headset
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sys/class/sound/hwC1D0/init_pin_configs: 0x12 0x90a60130 0x14 0x90170110
0x15 0x04211020 0x16 0x4000 0x18 0x41f0 0x19 0x41f0 0x1a
0x41f0 0x1b 0x41f0 0x1d 0x40c6832d 0x1e 0x41f0 0x1f
0x41f0
/sys/class/sound/hwC1D0/driver_pin_configs:
0x19 0x03a1913d
0x1b 0x03a1113c
/
Should remove 0x19
Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out
Control: name="Headphone Mic Boost Volume", index=0, device=0
ControlAmp: chs=3, dir=In, idx=0, ofs=0
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nst
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
** Summary changed:
- [hostname, Realtek ALC668, Mic, Internal] No autoswitch
+ [Asus N551JM, Realtek ALC668, Mic, Internal] No autoswitch
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Christopher, we (Andrew and Raymond) just verified that this bug still
manifests itself in the 4.8-rc7 kernel (4.8.0-040800rc7-generic). Please
let us know if you need any further information.
Thanks!
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Do your DSDT contain HID ? 10ec
** Changed in: alsa-driver (Ubuntu)
Status: Confirmed => Incomplete
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https://bugzilla.kernel.org/show_bug.cgi?id=115531#c1
** Bug watch added: Linux Kernel Bug Tracker #115531
http://bugzilla.kernel.org/show_bug.cgi?id=115531
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** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Title:
Thinkpad T460 & Ultra Dock - No audio output on dock plug
To manage
What happen when you
speaker-test -c4 -t wav -D hw:0
Which speaker did your hear front left, front right, rear left and rear
right?
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Title:
Len
Seem power still in D3
Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Mic Boost
Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps:
ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap
0x0020: IN Pin Default 0x90a6013
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/hda_generic.c?id=e7fdd52779a6c2b49d457f452296a77c8cffef6a
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Title:
Subsystem: FIRST INTERNATIONAL Computer Inc Device [1509:801b]
if your ion603 does not have use alc203's mono pin and pci subsytem id
is unique , the easy way is use snd_ctl_remove_id() to remove "Master
Mono Playback Volume" and "Master Mono Playback Switch" in cs5535audio.c
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Take a look the analog mixer block of alc203 datasheet
http://www.realtek.com/products/productsView.aspx?Langid=2&PNid=23&PFid=29&Level=5&Conn=4&ProdID=54
The audio path from
DAC output to headphone is MX18 + MX04 (i.e. PCM + Headphone)
DAC output to line-out is MX18 + MX02 (i.e. PCM + Master
>> This is the output of the alsa-info.sh command that was saved in the
/tmp directory. (14.7 KiB, text/plain)
This is not a real ens1371, just the emulated ens1371 sound card inside a vm
Manufacturer: VMware, Inc.
Product Name: VMware Virtual Platform
Product Version: None
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It can be reproduced with the emulated intel8x0 (stac97xx) inside
virtual box
Does it mean that when there is no "Front" or "Headphone" volume control
, sound-preference list "lfe-on-mono" port for the stereo internal audio
?
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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 m
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 48
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 s
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]
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>> 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
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>> Does this give you any clue?
you have to post the output of pulseaudio server
pulseaudio -k;pulseaudio -v
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you can use "strace alsactl store" to find out why "X11 connection
rejected"
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https://bugs.launchpad.net/bugs/557016
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ub
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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>> 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
>> Subdevic
-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: a
>> 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 sound
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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T
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, hdmi
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 speak
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 subs
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, plus
(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 pl
(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 downloade
(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 dr
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
subd
(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
f
(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 you
(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
> wo
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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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
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error message when gnome-alsamixer is launching
https://bugs.launchpad.net/bugs/106903
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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%
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Changing volume to 0% breaks channel ballance
https://bugs.launchpad.net/bugs/672420
You received this bug notificatio
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
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 Anal
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 m
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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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
At
>> 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 h
>> 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 boundar
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 =
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