** Changed in: alsa-driver
Status: Confirmed => Unknown
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Title:
[CA0106 - CA0106, playback] Playback
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Four Stereo ADC Inputs with Analogue Gain Adjust from +24dB to –21dB in 0.5dB
Steps
Digital Gain Adjust from -21.5dB to -103dB.
table 10 of wm8775 mention that 0.5 dB per steps and min 0 is mute
As max 255 is +24 dB , 0xcf is 0 dB ,
step 1 is -103 dB
Source #1
State: SUSPENDED
seem
chmap -Dsurround41:CARD=CA0106 query
fail but
chmap -Dsurround41:CARD=CA0106 get
return channel map
in ca0106.conf
there is no definition of ca0106.pcm.surround41
just
confdir:pcm/surround21.conf
confdir:pcm/surround41.conf
confdir:pcm/surround50.conf
Do surround21 ,
static unsigned int i2c_adc_init[][2] = {
{ 0x17, 0x00 }, /* Reset */
{ 0x07, 0x00 }, /* Timeout */
{ 0x0b, 0x22 }, /* Interface control */
{ 0x0c, 0x22 }, /* Master mode control */
{ 0x0d, 0x08 }, /* Powerdown control */
{ 0x0e, 0xcf }, /*
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_mixer.c?id=fff36e472b4315df77513f4339c5c199c6aad28b
static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
the scale only define step 1 is -103.50 dB and step size 0.5dB
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Tried with a stereo 48k file (converted
/usr/share/sounds/KDE-Im-Cant-Connect.ogg to wav, tested with ffprobe that it's
truly 48k/2chan) and 64 bytes buffer size:
# aplay -D plughw:CARD=CA0106 -v --buffer-size=64 /tmp/kde.wav
Playing WAVE '/tmp/kde.wav' : Signed 16 bit Little Endian, Rate 48000
Interesting. I've converted that file to 44100 Hz and now it reveals a
strange behavior. When I play it with buffer size of 64 it sounds like
the pitch is lower than it should be (even if it doesn't overrun though
it does from time to time). When the size is 128 it sounds fine.
# aplay -D
This command doesn't work, it says aplay: set_params:1239: Channels
count non available. The -D default:CARD=CA0106 variant works and it
does not overrun. I only had overruns on recording (at least, explicit).
That said, I've made a huge improvement in my system, I compiled a pf-
kernel and set
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:
(In reply to Raymond from comment #57)
Do surround21 , surround41 appear in
aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default:CARD=CA0106
CA0106, CA0106
Default Audio Device
sysdefault:CARD=CA0106
CA0106,
if you cannot find stereo 48000Hz wav file
aplay -D plughw:CARD=CA0106 -v --buffer-size=1024 any.wav
try to find minimum buffer size 1024,512, 256,128,64,... without
underrun
1024 frames = 21.33 ms x 48000Hz
4096 = 85.33 ms x 48000Hz
if request tlength 85.33ms is buffer time and 21.33ms is
how about CD audio (44100Hz stereo) using plughw?
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Title:
[CA0106 - CA0106, playback] Playback problem -
# aplay -D plughw:CARD=CA0106 -v --buffer-size=128
/usr/share/sounds/alsa/Side_Left.wav
Playing WAVE '/usr/share/sounds/alsa/Side_Left.wav' : Signed 16 bit Little
Endian, Rate 48000 Hz, Mono
Plug PCM: Route conversion PCM (sformat=S16_LE)
Transformation table:
0 - 0
1 - 0
Its setup is:
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
analog source seem supported by pulseaudio
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/mixer/paths
/analog-input.conf.common?id=67e3925795ad939f2c9acb3aa93122a18672fade
pactl list
do pulseaudio show four input ports : phone, mic, line in and aux
SImple mixer
Created attachment 111662
Pulse log
(In reply to Raymond from comment #35)
can you specify device 0
+ capture.pcm {
+type hw
+card $CARD
+ device 0
+
Added this, here's the startup log.
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CA0106 use specific name, you may need to add them to
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
/analog-output.conf
Simple mixer control 'Analog Center/LFE',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback 0 -
control.29 {
iface MIXER
name 'Phone Capture Volume'
value.0 0
value.1 0
comment {
access 'read write'
type INTEGER
count 2
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/sound/pci/ca0106/ca0106_main.c
GENERAL INFO:
*Model: SB0410
*P17 Chip: CA0106-DAT
*AC97 Codec: None
*ADC: WM8775EDS (4 Channel)
*DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD
(In reply to Raymond from comment #49)
did your sb0410 have these two chips ?
ADC: WM8775EDS (4 Channel)
DAC: CS4382
My card looks exactly like this:
http://www.ixbt.com/multimedia/creative-live!24bit/card-big.jpg
Probably, it has all these chips.
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arecord -v -d 1 -D iec958:CARD=CA0106 -f dat -t wav --dump-hw-params
test.wav
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Title:
[CA0106 - CA0106,
It does:
arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
overrun!!! (at least 0.084 ms long)
^CAborted by signal Interrupt...
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does analog capture also overrun ?
arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/ca0106/ca0106_mixer.c
if your sb0410 only have four jacks and support 7.1
it need snd_ca0106_capture_line_in_side_out which
(In reply to rkfg from comment #43)
It does:
arecord -f dat -D hw:CARD=CA0106 -t wav -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz,
Stereo
overrun!!! (at least 0.084 ms long)
^CAborted by signal Interrupt...
http://www.ixbt.com/multimedia/creative-live!24bit.shtml
seem need specific jack to support 7.1
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Title:
[CA0106
It happens more likely with small buffers and sometimes doesn't happen with
larger buffer. However, this behavior is not consistent:
%[homecomp]:[/tmp/test] arecord -f dat -D hw:CARD=CA0106 -t wav -d 5
--buffer-size=2048 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate
Created attachment 111663
pactl list log
Log of pactl list.
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Title:
[CA0106 - CA0106, playback] Playback
did your sb0410 have these two chips ?
ADC: WM8775EDS (4 Channel)
DAC: CS4382
if you look at cs4382 datasheet , it has three modes which can double or quad
the speed
for wm8775 which does not have any boost for mic, you have to ask the author
who add support of mic line in switch since
device.description = CA0106 Soundblaster (SB0410 SBLive! 24-bit) Analog Stereo
alsa.mixer_name = CA0106
module-udev-detect.discovered = 1
device.icon_name = audio-card-pci
Ports:
analog-input-mic: Microphone (priority: 8700)
http://www.alsa-project.org/main/index.php/XRUN_Debug
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Title:
[CA0106 - CA0106, playback] Playback problem -
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
HW Params of device iec958:CARD=CA0106:
ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
FORMAT: S16_LE S32_LE
SUBFORMAT: STD
SAMPLE_BITS: [16 32]
FRAME_BITS: [32 64]
CHANNELS: 2
RATE: [48000 192000]
http://git.alsa-project.org/?p=alsa-
lib.git;a=commitdiff;h=1af088e39b75a0a0897c7036487b143e983cd423;hp=57b5076c30b3453ee843912c0aeb3df8dbee3f68
it is strange the pulseaudio no longer probe surround40 , 51, and 71
your log have no traces of surround21 too
BTW, my mic is connected to the blue jack, not the usual pink one. This
is the only configuration I've found to be working, others result in
noise or silence. The mic is a bit more quiet (not _almost silent_ as if
it's connected to a Line-In jack, just not comfortable enough) than it
was on the
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_mixer.c?id=6129daaa0d2b84c0e376b6b17b3d3740c4d1d1ca
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Raymond tends to be side-tracked from time to time and ask questions
which are related to your original issue. Also apologies for not having
read the entire thread through.
PulseAudio closes the audio stream five seconds after no applications
are active, so that's probably when the issue goes
Jan 01 18:52:27 homecomp pulseaudio[14807]: [pulseaudio] alsa-mixer.c:
Skipping profile output:iec958-stereo+input:iec958-stereo - will not be
able to open input:iec958-stereo
seem cannot opem iec958 for input
can you specify device 0
+ capture.pcm {
+type hw
+
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer
/profile-sets/default.conf
[Mapping iec958-stereo]
device-strings = iec958:%f
channel-map = left,right
paths-input = iec958-stereo-input
paths-output = iec958-stereo-output
priority = 5
as pulseaudio already define
Created attachment 111585
ALSA test log
I've applied those changes for the ALSA config. I ran the test as
./a.out front:0 0 4 alsa-test.log and here's the output. I stopped it
with Ctrl-C. When I set the fillrate less than 4, it stops with the
assertion failure like before. Setting the fillrate
(In reply to Raymond from comment #25)
seem default sample 16 bits has no effect on format , pulseaudio prefer 32
bits
No, it does. I have these options uncommented in my daemon.conf:
default-sample-format = s32le
default-sample-rate = 48000
I have them enabled for some time and PA startup
CA0106.pcm.iec958.0 {
@args [ CARD AES0 AES1 AES2 AES3 ]
@args.CARD {
type string
}
@args.AES0 {
type integer
}
@args.AES1 {
type integer
}
@args.AES2 {
type integer
http://mailman.alsa-project.org/pipermail/alsa-
devel/2008-October/011913.html
: module-alsa-sink.c: Using 4 fragments of size 8816 bytes, buffer time is
99.95ms
D: module-alsa-sink.c: hwbuf_unused_frames=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to
do the error
alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77)
appear when you fix the error cannot lock ctl elem as pulseaudio
start the probing of those surround40, 51 and71 ?
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those locks in hook plugin prevent you running
aplay -D iec958:CARD=CA0106 stereo48000.wav
and
arecord -D iec958:CARD=CA0106 test.wav
at same time
those message Cannot lock ctl elem should disaapear when you can playback and
capture iec958 at same time ?
post pulseaudio verbose log
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Created attachment 111616
pcm_avail log
Here's the output of pcm_avail.c program.
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Title:
[CA0106 - CA0106,
Created attachment 111615
Fixed PA log
It now doesn't report Cannot lock ctl elem.
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Title:
[CA0106 - CA0106,
(In reply to rkfg from comment #26)
Created attachment 111615 [details]
Fixed PA log
It now doesn't report Cannot lock ctl elem.
it is strange that pulseaudio still don't probe surround profiles
pactl list should list those surround profiles similar to
seem default sample 16 bits has no effect on format , pulseaudio prefer
32 bits
Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] card.c: Created 2
alsa_card.pci-_06_00.0
Dec 30 22:56:29 homecomp pulseaudio[1103]: [pulseaudio] alsa-util.c: cannot
disable ALSA period wakeups
Dec 30
for alsa-time-test.c
dev = argc 1 ? argv[1] : front:0;
cap = argc 2 ? atoi(argv[2]) : 0;
fillrate = argc 3 ? atoi(argv[3]) : 1;
there are three parameter which allow you to change
device
capture or playback
fillrate
most sound cards cannot use default fill rate 1
for hda-intel,
can you post the output of alsa-info.sh ?
did it support 7.1 since there are only 4 jacks ?
speaker-test -c 4 -t wav -D surround40:CARD=CA0106
speaker-test -c 6 -t wav -D surround51:CARD=CA0106
speaker-test -c 8 -t wav -D surround71:CARD=CA0106
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(In reply to rkfg from comment #27)
Created attachment 111616 [details]
pcm_avail log
Here's the output of pcm_avail.c program.
avail increment at 8 frames which is less than min period size 16 and
period size 1024
this mean some tsched feature are supported
hwbuf_unused can be set to non
(In reply to Raymond from comment #14)
you need to put the current ca0106 iec958 slave pcm and hook into playback
slave and create capture slave of asym plugin
Sorry, I don't understand. I'm not that familiar with low-level ALSA configs, I
only did some basic operations with ~/.asoundrc
What
/usr/share/alsa/cards/CA0106.conf
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Title:
[CA0106 - CA0106, playback] Playback problem - Surround dies, no valid
Good, the file is there and it has those lines you mentioned in #10. Now
how should I modify them? Could you provide a diff to apply? I don't
really get what to put the current ca0106 iec958 slave pcm and hook
into playback slave and create capture slave of asym plugin means. If
the alternative
(In reply to Raymond from comment #8)
how did you run the program ?
the program is hardcoded to use 44100Hz , it should run continously without
any error if fillrate is equal to period size
if sound card can report hw_ptr with better grannularity, the program can
still run continously
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
pcm.name {
type asym # Asym PCM
playback STR# Playback slave name
# or
playback { # Playback slave definition
pcm STR # Slave PCM name
http://mailman.alsa-project.org/pipermail/alsa-
devel/2014-September/081501.html
you can try Alexander's pcm_avail.c to find out whether your ca0106 can
report DMA_RESIDUE_GRANULARITY_BURST ?
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Created attachment 111649
ALSA info
Here's the output of alsa-info.sh. The speaker test works, all three
commands without errors. However, I only have 2 speakers so I hear
front left and front right.
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** Changed in: alsa-driver (Ubuntu)
Status: Expired = Incomplete
** Bug watch added: freedesktop.org Bugzilla #87713
https://bugs.freedesktop.org/show_bug.cgi?id=87713
** Also affects: alsa-driver via
https://bugs.freedesktop.org/show_bug.cgi?id=87713
Importance: Unknown
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https://bugs.freedesktop.org/show_bug.cgi?id=87713.
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