Default sink name: alsa_output.pci-_00_1b.0.analog-stereo
Default source name:
alsa_input.usb-Sony_Computer_Entertainment_Wireless_Stereo_Headset-00-Headset.analog-mono
in theory , it should switch to hdni or usb
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_ca0132.c?id=fe14f39e88c8ac16d0a051f8444a2294f8cb358c
ca0132 only support chromebook and alienware 15
you may need to find out hp pin by
hdajacksensetest -a
+ spec->num_outputs = 2;
+
!Sysfs Files
!!---
/sys/class/sound/hwC0D2/init_pin_configs:
0x0b 0x010140f0
0x0c 0x014520f0
0x0d 0x414570f0
0x0e 0x41c510f0
0x0f 0x412170f0
0x10 0x022140f0
0x11 0x018110f0
0x12 0x01a190f0
0x13 0x50d000f0
0x18 0x50f0
try
hdajacksensetest -a
when you plug and unplug headphone
--
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Default sink name: alsa_output.pci-_00_1b.0.analog-stereo.equalizer
Default source name: alsa_input.pci-_00_1b.0.analog-stereo
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Title:
[L
aug 25 09:49:54 hostname pulseaudio[1544]: [pulseaudio]
module-equalizer-sink.c: module-equalizer-sink is currently unsupported, and
can sometimes cause PulseAudio crashes, increased latency or audible artifacts.
aug 25 09:49:54 hostname pulseaudio[1544]: [pulseaudio]
module-equalizer-sink.c: If
ports:
analog-output-speaker: Luidsprekers (priority 1, latency
offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-speakers"
analog-output-headphones: Analoge koptelefoon (priority 900
control.22 {
iface CARD
name 'Headset Mic Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
control.23 {
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Title:
[Latitude E5530 non-vPro, IDT 92HD93BXX, Speaker, Internal] No
aut
http://git.alsa-project.org/?p=alsa-
tools.git;a=commit;h=7a7d94a031b6b3c605c47de2779b0e03442a8d30
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https://bugs.launchpad.net/bugs/1487861
Title:
Problems with Headphone Jack Z87 Sting
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1315769
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Title:
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda?qt=grep&q=vt1708
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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https://bugs.launchpa
** Changed in: virtualbox (Ubuntu)
Status: Invalid => Incomplete
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https://bugs.launchpad.net/bugs/1487168
Title:
[Emulated AD1980 AC97 inside Virualbox 5.0, playback] volume slid
the count of master mono playback volume control is not stereo but mono
this is emulation bug of virtual box
control.3 {
iface MIXER
name 'Master Mono Playback Switch'
value.0 false
value.1 false
comment {
” Mono Volume Register (Index 06h) Reg No.Name
D15D14D13D12D11D10D9D8D7D6D5*D4D3D2D1D0
Default 06h
Mono Volume
MVM X X X X X X X X X MV5 MV4 MV3 MV2 MV1 MV0 8000h
*Refer to Table II for examples. This register controls the Mono output
volume and mute bit. The volume register contains five bits,
0:06 = bfbf
a real ad1980 won't return this value since only bit 15 and (bit 5 to bit 0)
are defined
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Title:
[Emulated AD1980 AC97 inside Viru
static DECLCALLBACK(int) ichac97IOPortNAMWrite(PPDMDEVINS pDevIns,
void *pvUser, RTIOPORT
Port, uint32_t u32, unsigned cb)
{
PAC97STATE pThis = (PAC97STATE)pvUser;
case 2:
{
[ 15.709705] hda-intel :80:01.0: Using VIACOMBO position fix
[ 15.709798] snd_hda_intel :80:01.0: PCI: Disallowing DAC for device
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Titl
https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers
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Title:
[, Analog Devices AD1984A, Green Speaker, Internal] Playback problem
no speak
you need virtualbox verbose log which provide debug messages printed by
LogFlowFunc()
the linux driver check those ad specific registers for multi codecs ,
and ADI compatiable mode
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/ac97/ac97_patch.c
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the reason for no master playback volume/switch is
master rename as master surround but no headphone playback volume/switch
state.I82801AAICH {
control.1 {
iface MIXER
name 'Master Surround Playback Switch'
value.0 true
value
only virtualbox debug log can provide debug info about why emulated ich
controller support multi channel
case GLOB_CNT:
1470/* Global Control */
1471*pu32 = pThis->glob_cnt;
1472LogFlowFunc(("glob_cnt -> %#x\n"
if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_PCM_SLEFT)) {
chip->multi4 = 1;
if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE)) {
chip->multi6 = 1;
if (chip->ac97[0]->flags & AC97_HAS_8CH)
control.33 {
iface CARD
name 'Front Mic Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
control.34 {
https://wiki.ubuntu.com/PulseAudio/Log
you have to post the pulseaudio verbose log in virtualbox 5.0 ad1980
find the dB range of analog outout path
since "master surround playback volume" is not defined in pulseaudio
analog-output-*.conf
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/sr
https://www.virtualbox.org/changeset/55272/vbox/trunk/src/VBox/Devices/Audio/DevIchAc97.cpp
since it hardcode ad1980 on a dell machine, you may need patch in alsa
driver to prevent the swap of the master playback volume/switch and
headphone playback volume/switch
if (fChipAD1980)
{
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1325308
this is dell dimension 8400 with ad1980 which virtualbox 5.0 try to
emulate
the difference of alsa-info between your and the real dell dimension
8400
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aplay -D hw:0,0 --dump-hw-params stereo.wav
this dump the channel max of the sound cardc driver
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Title:
[Emulated AD1980 AC97 inside Virualbox
HW Params of device "hw:0,0":
ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT: S16_LE
SUBFORMAT: STD
SAMPLE_BITS: 16
FRAME_BITS: [32 96]
CHANNELS: [2 6]
RATE: 48000
the emulated intel8x0 claim to support 6 channels
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> I don't understand this response.
I am not an experienced programmer with respect to these platforms and have
no control what Virtual Box presents back to the OS.
I will say that this problem effects multiple distributions.
You should study the ad1980 datasheet
take a look at the functional
** Bug watch added: Virtualbox Trac #14508
http://www.virtualbox.org/ticket/14508
** Also affects: virtualbox via
http://www.virtualbox.org/ticket/14508
Importance: Unknown
Status: Unknown
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static void snd_intel8x0_codec_write(struct snd_ac97 *ac97,
unsigned short reg,
unsigned short val)
{
struct intel8x0 *chip = ac97->private_data;
+ dev_info(chip->card->dev, "ac97 write %d - %.2x : %.4x\n", ac97->nu
as you should notice that driver use ac97->num to access different
codecs
iaputword(chip, reg + ac97->num * 0x80, val);
res = iagetword(chip, reg + ac97->num * 0x80);
but virutalbox does not response correctly when driver try to accesss
secodary ac97 codecs when it only emulate single prima
:06 username pulseaudio[684]: [autospawn] core-util.c: Home directory not
accessible: Permission denied
Aug 29 06:03:06 username pulseaudio[684]: [autospawn] lock-autospawn.c: Cannot
access autospawn lock.
Aug 29 06:03:06 username pulseaudio[684]: [pulseaudio] main.c: Failed to
acquire autospawn
http://git.alsa-project.org/?p=alsa-
kmirror.git;a=commitdiff_plain;h=bb9a16dc52281d98dfb525552d29d8939845d8e2
it was this patch change master mono from mono to stereo
/* build master mono controls */
if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_MONO)) {
-
** Summary changed:
- [ENS1371 - Ensoniq AudioPCI, playback] Sound volume set to 0 when booting
from install CD
+ [Enumertaed ENS1371 inside VMWare, playback] Sound volume set to 0 when
booting from install CD
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[3.154949] ioremap: invalid physical address f000fea5f000312c
[3.154952] [ cut here ]
[3.154957] WARNING: CPU: 7 PID: 322 at
/build/linux-OEJm1p/linux-4.1.0/arch/x86/mm/ioremap.c:104
__ioremap_caller+0x32e/0x360()
[3.154958] Modules linked in: mac_hid tpm_c
the driver and firmwarewas designed for chromebook which support stereo
only
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_ca0132.c?id=fe14f39e88c8ac16d0a051f8444a2294f8cb358c
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Added jdstrand's usr.bin.evince fix above to evince 3.10.3 in Ubuntu
14.04 trusty release and it seems to fix it.
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Title:
Apparmor denial when vi
https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers
Sound(ALSA)
you have to file upstream bug
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Title:
No sound at all snd-hda-intel / ALC
spdif out does not has any volume control
this look like pulseaudio bug
** Package changed: alsa-driver (Ubuntu) => pulseaudio (Ubuntu)
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Title:
Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital
Converter: stream=8, channel=0
Digital: Enabled GenLevel
Digital category: 0x2
IEC Coding Type: 0x0
PCM:
rates [0x5f0]: 32000 44100 48000 88200 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Power states: D0 D1 D2 D
http://git.alsa-project.org/?p=alsa-tools.git;a=tree
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Title:
[HP EliteBook 840 G2, Realtek ALC3228, Black Headphone Out, Right]
Headphone out o
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=b4b33f9d64c3edcdcbea874acdc1e9626fd961f1
00:1b.0 0403: 8086:9ca0 (rev 03)
Subsystem: 103c:2216
try add pci quirk for your laptop if those missing dock pins are the same as
yours
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> Should I file a bug report somewhere else for that?
https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio
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https://bugs.launchpad.net/bugs/1040873
Title:
[N56VZ, Realtek ALC6
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1302090
try hdajackretask to change node 0x1a to speaker
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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try hdajackretask to change node 0x1a as speaker
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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post output of
hdajacksensetest -c1 -a
before you dock, plug and unplug of dock line out and line in jacks
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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https://
> changing "switch" from "mute" to "on" under [Element PCM] in
/usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf.common (see
below) did have an effect -- pressing the "mute" button now only mutes
Master and Headphone (not PCM anymore), pressing it again only unmutes
Master. Changing the sam
you should follow up the bug report vt82xx and alc662
https://bugzilla.kernel.org/show_bug.cgi?id=18512
** Bug watch added: Linux Kernel Bug Tracker #18512
http://bugzilla.kernel.org/show_bug.cgi?id=18512
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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do you mean the module-switch-on-port-available change the default sink
or pulseaudio did not remember your default setting after reboot ?
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Title:
you have to post output of
pactl list
and pulseaudio verbose log
in theory, here should be three output ports
dock line out, headphone and speaker
internal speaker is selected when both line out and headphone are not
plugged
control.21 {
iface CARD
name 'Dock L
http://www.idt.com/products/audio-products/high-definition-hd-audio-
codecs/92hd71b-four-channel-hd-audio-codec
Codec: IDT 92HD71B7X does not have any mute switch at the pin complex
the driver just set pin ctl to 0 to mute the speaker when headphone is
plugged
this mean that pulseaudio cannot f
do your ASRock A790GXH/128M have fornt audio panel connectors , it is
strange that the codec info have no pin complex for the headphone and
front mic ?
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pcm playback volume of the analog mixer have 12dB gain which is
different from those front, ... , side playback volume
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Mic' 'Line' 'CD' 'Stereo Mixer'
Item0: 'Mic'
you can select stereo mixer to record what is playing
Nod
:~$ amixer sset Master toggle
Simple mixer control 'Master',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum
Playback channels: Mono
Limits: Playback 0 - 31
Mono: Playback 16 [52%] [-22.50dB] [off]
this means that ctl.default is not set to ctl.pulse as your master
Refer to ASRock A790GXH/128M user manual
To enable Multi-Streaming function, you need to connect a front panel audio
cable to the front
panel audio header. After restarting your computer, you will find “VIA HD Audio
Deck” tool on
your system. Please follow below
instructions according to the OS
http://voices.canonical.com/david.henningsson/2012/07/13/top-five-wrong-
ways-to-fix-your-audio/
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Title:
[p6654y, Realtek ID 888, Green Line Out,
https://wiki.ubuntu.com/Audio/AlsaInfo
Node 0x1a [Pin Complex] wcaps 0x400501: Stereo
Pincap 0x00010010: OUT EAPD
EAPD 0x2: EAPD
Pin Default 0x901701f0: [Fixed] Speaker at Int N/A
Conn = Analog, Color = Unknown
DefAssociation = 0xf, Sequence = 0x0
Misc = NO_PRESENCE
Pin-ctls:
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/ALSA-Configuration.txt
have you changed power_save or power_save_controller setting ?
power_save- Automatic power-saving timeout (in second, 0 =
disable)
power_save_controller - R
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/ALSA-Configuration.txt
The model name "generic" is treated as a special case. When this
model is given, the driver uses the generic codec parser without
"codec-patch". It's sometimes good for test
https://wiki.ubuntu.com/Audio/AlsaInfo
** Changed in: alsa-driver (Ubuntu)
Status: Confirmed => Incomplete
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Title:
Sound output device ke
http://git.alsa-project.org/?p=alsa-
lib.git;a=commitdiff;h=57b5076c30b3453ee843912c0aeb3df8dbee3f68;hp=326c6802e49e5499e16cf141e1cdb0360fce14aa
do asus_mode4 still work when surround21 use 6 channels instead of 4 ?
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do you mean you close laptop lid while the sound is playing ?
you have to provide alsa-info before closing the lid and after reopen
the lid
codec has two paths for the output pins which can either connect directly to
audio output 0x15, 0x16, 0x17 or analog mixer 0x1d through node 0x1e by
enab
Refer to Studio 1735 quick reference guide
the mic jack can be used as line in or surround line out connector
Audio Audio type 5.1 channel High Definition Audio
Audio controller IDT 92HD73C
Stereo conversion 24-bit (analog-to-digital and digital-to-analog)
Interfaces:
Internal
Intel High Definit
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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Title:
Pulseaudio crashes system
To manage notifications about this bug go
[ 1208.436593] totem-plugin-vi[5809]: segfault at 7ffe ip
7f41f7f4de0a sp 7fff98856d80 error 5 in
libglib-2.0.so.0.3200.4[7f41f7eed000+f2000]
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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** Package changed: alsa-driver (Ubuntu) => totem (Ubuntu)
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Title:
[Satellite L640, Conexant CX20585, Green Headphone Out, Right] No
sound at a
the case are different if you have laptop with different codecs,
pulseaudio require driver to provide jack detection control since ubuntu
12.04
Control: name="Headphone Jack", index=0, device=0
iface CARD
name 'Headphone Jack'
value false
comment {
access read
type BOOLEAN
cou
for laptop with docking station
both the headphone jack and dock station line out/headphone jack are
used to determine whether the speaker is mute or not
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> Which part of "pactl list" is relevant?
the relevant point is
1) how do pulseaudio know which volume control should be used when the
volume control is shared by two or more pin complexes
2) whether "PCM playback volume" and "PCM playback Switch" is slave of
virtual "Master playback volume" an
you have to follow up the pulseaudio bug upstream if your need to use
the dock station
if you Don' t need the dock station and you are using latest alsa
driver
you can use hdajackretask to change [Jack] of the dock headphone/line
out Jack to [n/a], the driver will be able to create the two volu
if you look at 92HD71b block diagram, the mute switch at dac0 does not
affect the headphone which use dac1
this mean that both pcm playback volume switch of your laptop is not
suitable defined in
.include analog-output.conf.common
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[ 13.268373] snd_hda_codec: module verification failed: signature and/or
required key missing - tainting kernel
[ 13.292488] snd_hda_intel: Unknown symbol snd_card_new (err 0)
[ 13.292793] snd_hda_intel: Unknown symbol snd_card_new (err 0)
[ 13.293120] snd_hda_intel: Unknown symbol snd_car
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/HDA-
Intel.conf;hb=HEAD
for those without hardware volume control with name "PCM", the softvol
plugin "PCM playback volume" is common to all path when the application
use "front" device
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Codec: Realtek ALC283
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0283
Subsystem Id: 0x102805f9
Revision Id: 0x13
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=4b016931a997be59a2be0da3398985a19fdfa5be
+ { 0x10ec0283, 0x1028, 0, "ALC3223" },
seem rena
how many audio jacks do your Inspiron 7537 have ?
do it support conventional mic and headset mic ?
alsa.mixer_name = "Realtek ALC283"
alsa.components = "HDA:10ec0283,102805f9,0013"
module-udev-detect.discovered = "1"
device.icon
[ 15.611085] autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker
[ 15.611089]speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 15.611092]hp_outs=2 (0xf/0xa/0x0/0x0/0x0)
[ 15.611093]mono: mono_out=0x0
[ 15.611095]dig-out=0x21/0x0
[ 15.611096]inputs:
[ 15.611099] In
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=5e915bb3677f1369223a87e488c340236f81bfc2
[ALSA] hda-codec - Re-add quirk support for Dell XPS 1330 and Inspiron
1420
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=2f32d909f
[ 15.624033] autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line
[ 15.624036] speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 15.624039] hp_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 15.624040] mono: mono_out=0x0
[ 15.624041] inputs:
is the null config related to HDA modem codec
Codec: Conexant ID 2c06
Address:
/sys/class/sound/hwC0D0/init_pin_configs:
0x0a 0x0221101f
0x0b 0x40f000f0
0x0c 0x40f000f1
0x0d 0x90170110
0x0e 0x02811030
0x0f 0x02011020
0x10 0x40f000f2
0x11 0x40f000f3
0x12 0x40f000f4
0x13 0x90a60040
0x14 0x40f000f5
0x21 0x034410a0
0x22 0x40f000f6
/sys/class/sound/hwC0D0/driver_pin_configs:
0x0f
ports:
analog-output-speaker: Speakers (priority 1, latency offset
0 usec, available: unknown)
properties:
device.icon_name = "audio-speakers"
analog-output-headphones: Headphones (priority 9000, latency
o
seem support surround 5.1 by retasking mic jack
http://support1.ap.dell.com/cn/zh/forum/thread.asp?fid=35&tid=132304
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Title:
[Inspiron 1420, Si
seem pulseaudio did not check "Front Headphone Surround"
https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio
> grep -ir "Headphone" *
analog-input-headphone-mic.conf:; For some ASUS netbooks that have one jack
that can be either a Headphone
analog-input-headphone-mic.conf:[Jack Head
control.19 {
iface CARD
name 'Headphone Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
** Changed in: alsa-driver (Ubun
analog-output-headphones: Cuffie analogiche (priority 9000, latency offset 0
usec, available: no)
properties:
device.icon_name = "audio-headphones"
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Bugs, which is subs
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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https://bugs.launchpad.net/bugs/1390474
Title:
"Volume element PCM has 6 channels. That's too much!" with Terratec
why do you still use 12.04 ?
try latest version of live cd ?
Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L
Control: name="Master Playback Volume", index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Device: name="CONEXANT Analog", type="Audio", device=0
Amp-Out caps:
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1402052
seem usb audio devices build the multi volume control in the same way
so you should file an up stream pulseaudio bug report for usb audio
device
have you ask Arun how to revert his patch ?
https://bugs.freedesktop.org/enter_bug
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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index: 0
driver:
owner module: 15
properties:
application.name = "Login Session c3"
systemd-login.session = "c3"
index: 1
driver:
owner module: 15
properties:
application.n
control.19 {
iface CARD
name 'Headphone Jack'
value true
comment {
access read
type BOOLEAN
count 1
}
}
Node 0x14 [Pin Complex] wcaps 0x4
+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 o
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
Do surround21 , surround41 appear in
aplay -L
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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 }, /* Attenu
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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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
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: uid=100
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
how about CD audio (44100Hz stereo) using plughw?
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Title:
[CA0106 - CA0106, playback] Playback problem - Surround dies, no valid
output
To man
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 per
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