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[Alsa-user] alsa 0.9 with a cs4237b -> 2.4.19 kernel freeze

[alsa user]

nt Left - Front Right  Limits: Playback 0 - 63 
Front Left: Playback 63 [100%] [on]  Front Right: Playback 63 [100%] 
[on] Simple mixer control 'DSP',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right  Limits: Playback 0 - 63 
Front Left: Playback 63 [100%] [on]  Front Right: Playback 63 [100%] 
[on] Simple mixer control 'Line',0
Capabilities: volume pswitch cswitch
Playback channels: Front Left - Front Right  Capture channels: Front 
Left - Front Right  Limits: 0 - 31
Front Left: 31 [100%] Playback [on] Capture [off]  Front Right: 31 
[100%] Playback [on] Capture [off] Simple mixer control 'Line Capture 
Bypass',0
Capabilities: pswitch
Playback channels: Front Left - Front Right  Mono:  Front Left: 
Playback [on]  Front Right: Playback [on] Simple mixer control 'CD',0
Capabilities: volume pswitch cswitch
Playback channels: Front Left - Front Right  Capture channels: Front 
Left - Front Right  Limits: 0 - 31
Front Left: 31 [100%] Playback [on] Capture [off]  Front Right: 31 
[100%] Playback [on] Capture [off] Simple mixer control 'Mic',0
Capabilities: volume pswitch cswitch
Playback channels: Front Left - Front Right  Capture channels: Front 
Left - Front Right  Limits: 0 - 31
Front Left: 12 [39%] Playback [off] Capture [on]  Front Right: 12 [39%] 
Playback [off] Capture [on] Simple mixer control 'Mic Playback Boost',0
Capabilities: pswitch
Playback channels: Front Left - Front Right  Mono:  Front Left: 
Playback [off]  Front Right: Playback [off] Simple mixer control 'IEC958 
Output CSBR',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'IEC958 Output Channel Status 
High',0
Capabilities: volume volume-joined
Playback channels: Mono
Limits: 0 - 255
Mono: 0 [0%] Simple mixer control 'IEC958 Output Channel Status Low',0
Capabilities: volume volume-joined
Playback channels: Mono
Limits: 0 - 127
Mono: 64 [50%] Simple mixer control 'IEC958 Output Enable',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'IEC958 Output User',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'IEC958 Output Validity',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'Mono',0
Capabilities: pvolume pvolume-joined pswitch
Playback channels: Front Left - Front Right  Limits: Playback 0 - 15 
Front Left: Playback 0 [0%] [off]  Front Right: Playback 0 [0%] [off] 
Simple mixer control 'Mono Output',0
Capabilities: pswitch
Playback channels: Front Left - Front Right  Front Left: Playback [off] 
Front Right: Playback [off] Simple mixer control 'Mono Playback Bypass',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'Capture',0
Capabilities: cvolume
Capture channels: Front Left - Front Right  Limits: Capture 0 - 15
Front Left: Capture 15 [100%]  Front Right: Capture 15 [100%] Simple 
mixer control 'Capture Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right  Limits: 0 - 3
Mono: 3 [100%]  Front Left:  Front Right: Simple mixer control 'Analog 
Loopback',0
Capabilities: cswitch
Capture channels: Front Left - Front Right  Front Left: Capture [off] 
Front Right: Capture [off] Simple mixer control 'Digital Loopback',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right  Limits: Playback 0 - 63 
Front Left: Playback 0 [0%] [off]  Front Right: Playback 0 [0%] [off] 
[root@alsatest /root]#




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Re: [Alsa-user] alsa 0.9 with a cs4237b -> 2.4.19 kernel freeze

[alsa user]

Takashi Iwai wrote:


At Fri, 29 Nov 2002 01:12:28 -0600,
alsa user wrote:



Using a Crystal cs4237B device with ALSA drivers from either
the stable release or the latest development release.
When opening and reading from the audio device (/dev/audio
or /dev/dsp0) the system hangs. [Opening the device for write
only is no problem -- and playing music works great.]

With latest 0.9 release things are definitely better (than with 0.5),
but reading /dev/audio or /dev/dsp0 still causes same lockup.

What works:
It is possible to record audio using "arecord" or "record" and at the
same time play some tune in a different Unix process using "aplay"
(and ALSA playing quality is outstanding!). It is also possible
to do conference calls with gnomemeeting, however once in a while
the kernel will still freeze.

What never works: kernel will always freeze within a second
of reading from /dev/dsp0 or /dev/audio (e.g. cat /dev/dsp0).
Detailed configurations attached.



it sounds like a problem on OSS emulation.
could you try to playback and capture at the same time _via OSS
emulation_ ?  it doesn't matter whether it's a single process or not.
just run e.g. "play" and "rec" (both from sox) at the same time.


using rec to record audio freezes the computer even without running
play in the background. Apparently only "arecord" and "record" are
stable way to record in this configuration. Anything else I should try?
I have turned debugging on the kernel but not sure what to look for
after a reboot in this case.




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Re: [Alsa-user] alsa 0.9 with a cs4237b -> 2.4.19 kernel freeze

[alsa user]

Takashi Iwai wrote:


At Sat, 30 Nov 2002 10:17:49 -0600,
alsa user wrote:


it sounds like a problem on OSS emulation.
could you try to playback and capture at the same time _via OSS
emulation_ ?  it doesn't matter whether it's a single process or not.
just run e.g. "play" and "rec" (both from sox) at the same time.


using rec to record audio freezes the computer even without running
play in the background. Apparently only "arecord" and "record" are
stable way to record in this configuration. Anything else I should try?
I have turned debugging on the kernel but not sure what to look for
after a reboot in this case.



ok, then please tell me the detail you're using.
which kernel, which alsa version, which soundcard (or laptop)?

do you mean "freeze" as a complete lock-up (typically, blinking LEDs)?
or did you get any kernel oops or panic messages?


1) Using a Crystal cs4237B device with ALSA drivers
on Dell CPI laptop -- verified this card accurate from DELL
technical spec. It is also the card correctly loaded by alsa.
All options from alsamixer (2 dozens) are functional.
2) ALSA Versions for this test -- the latest dev versions at the time:
alsa-driver-0.9.0rc5 alsa-lib-0.9.0rc5 alsa-utils-0.9.0rc5
3) Kernel is generic 2.4.19 compiled with gcc-3.0 (same
for alsa)

Freeze symptoms: Screen and cpu freeze, tty access
frozen (cannot switch to another tty screen)
and netwok unreachable as well.
Do not see any oops or panic message.

3 LEDS are in ON OFF OFF state
(CPU, HD, Battery)
The network LED is still on, but it is NOT possible
to access laptop remotely.
There is NO hard-disk activity.

Original message config details follow:

[ using generic kernel.org 2.4.19 build with gnu gcc 3.0.
Alsa 0.9 built with same compiler.
Independent builds on each machine.]

Versions for this test -- the latest dev versions:
alsa-driver-0.9.0rc5 alsa-lib-0.9.0rc5 alsa-utils-0.9.0rc5
(loaded driver without the "-card-").

Versions used using "snd-card-" prefix instead of "snd-"
(also using "index" and adding "snd_").
alsa-driver-0.5.12a alsa-utils-0.5.10 alsa-lib-0.5.10b
Same lockup results. (Just verifying this is not something
that was introduced with new development version...)

[root@alsatest 2.4.19]# lsmod
Module Size Used by
snd-cs4236 5552 0 (unused)
snd-cs4236-lib 10352 0 [snd-cs4236]
snd-opl3-lib 5408 0 [snd-cs4236]
snd-hwdep 3408 0 [snd-opl3-lib]
snd-mpu401-uart 2800 0 [snd-cs4236]
snd-rawmidi 12256 0 [snd-mpu401-uart]
snd-seq-device 3728 0 [snd-opl3-lib snd-rawmidi]
snd-cs4231-lib 13776 0 [snd-cs4236 snd-cs4236-lib]
snd-pcm 56864 0 [snd-cs4236-lib snd-cs4231-lib]
snd-timer 9824 0 [snd-opl3-lib snd-cs4231-lib snd-pcm]
snd 24096 0 [snd-cs4236 snd-cs4236-lib snd-opl3-lib snd-hwdep 
snd-mpu401-uart snd-rawmidi snd-seq-device snd-cs4231-lib snd-pcm snd-timer]
soundcore 3536 3 [snd]
xirc2ps_cs 12064 1 [root@alsatest 2.4.19]# arecord -l
card 0: Crystal_Audio [CS4237B], device 0: CS4231 [CS4237B]
Subdevices: 1/1
Subdevice #0: subdevice #0
[root@alsatest 2.4.19]# [root@alsatest /root]# amixer
Simple mixer control 'Master Digital',0
Capabilities: volume pswitch cswitch
Playback channels: Front Left - Front Right Capture channels: Front Left 
- Front Right Limits: 0 - 71
Front Left: 71 [100%] Playback [on] Capture [off] Front Right: 71 [100%] 
Playback [on] Capture [off] Simple mixer control '3D Control - Center',0
Capabilities: volume volume-joined
Playback channels: Mono
Limits: 0 - 15
Mono: 0 [0%] Simple mixer control '3D Control - IEC958',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control '3D Control - Mono',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control '3D Control - Space',0
Capabilities: volume volume-joined
Playback channels: Mono
Limits: 0 - 15
Mono: 0 [0%] Simple mixer control '3D Control - Switch',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off] Simple mixer control 'PCM',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right Limits: Playback 0 - 63 
Front Left: Playback 63 [100%] [on] Front Right: Playback 63 [100%] [on] 
Simple mixer control 'Synth',0
Capabilities: volume pswitch cswitch
Playback channels: Front Left - Front Right Capture channels: Front Left 
- Front Right Limits: 0 - 31
Front Left: 31 [100%] Playback [on] Capture [off] Front Right: 31 [100%] 
Playback [on] Capture [off] Simple mixer control 'Synth Capture Bypass',0
Capabilities: pswitch
Playback channels: Front Left - Front Right Mono: Front Left: Playback 
[on] Front Right: Playback [on] Simple mixer control 'FM',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right

Re: [Alsa-user] alsa 0.9 with a cs4237b -> 2.4.19 kernel freeze

[alsa user]

Takashi Iwai

Any news or suggestion on this issue? Do I need to resubmit
this problem to the development list as well? A developper
from Alsa project is welcome to visit us and investigate the
problem on site -- we can only offer room and board but that
could be worth the trip itself...

Best Regards,

[EMAIL PROTECTED]

http://well2u.com

At Sat, 30 Nov 2002 10:17:49 -0600,
alsa user wrote: [ snip ]
This previous message tried to provide answers to remaining questions.







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rc6 FIX -- was Re: [Alsa-user] alsa 0.9 with a cs4237b -> 2.4.19 kernel freeze

[alsa user]

Takashi Iwai wrote:



well, iirc, there was a bug regarding the capture on oss emulation.
could you try the latest cvs version whether this problem still
exists?


YES, alsa 09 rc6 fixes the problems reported a few weeks ago
(that existed since alsa 05), it is now possible to use "rec"
without freezing the kernel, it is also possible to use "rec" while
"play" is working. It is possible (as expected) to open /dev/dsp
and /dev/audio for read without freezing.

Great thanks!
[EMAIL PROTECTED]
http://well2u.com



if the problem still remains, then please repost it to alsa-devel, so
that more people can take a look at the problem.


thanks,

Takashi






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[Alsa-user] MIDI keyboard attached through serial port

[alsa-user]

 Hi,

   I have Mandrake 9.0, a serial port and a MIDI keyboard w/serial port
DIN. I can see the keyboard sending carrier & MIDI data under minicom at
34800 8N1 w/HW flow control.

   Mandrake 9.0 uses alsa-utils-0.9.0-0.6rc2mdk and libalsa2-0.9.

   How do I record from the keyboard ? I guess this divides:
1. How do I connect the alsa/MIDI subsystem to the serial port device, 
and 2. How do I set things up then / what (simple) tool do I use then to 
record something from the keyboard (and play it back) ?

   Any pointers are appreciated (I've done google, but nothing 
straightforward came out.)


Thanks a bunch,

   John


-- 
-- Gospel of Jesus' kingdom = saving power of God for all who believe --
 ## To some, nothing is impossible. ##
   http://Honza.Vicherek.com/



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[Alsa-user] Alsa+Mplayer outstanding on P 266MHz

[alsa user]

Well this may sound obvious to everyone here,
but I am thrilled, I can't believe how fairly good
dvds are playing on a refurbished 266MHz P
laptop. After replacing the oss drivers with alsa
an using vesa, the audio is undistorted and the
image frame rate is acceptable with:

mplayer -fs -zoom -framedrop -nodouble -vo vesa -dvd 1

Only on Linux! Every one else wants > 500MHz I hear.
http://well2u.com/olivier/class/linux/mplayer/mpvesa
Just push the limit...



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[Alsa-user] Support for Gigabyte GA-7N400-L1 mobo with Realtek ALC650?

[alsa-user]

Hello,

I own a Gigabyte GA-7N400-L1 AMD motherboard
(home page at:
http://www.giga-byte.com/MotherBoard/Products/Products_GA-7N400-L1.htm
detailed specs at:
http://www.giga-byte.com/MotherBoard/Products/Products_Spec_GA-7N400-L1.htm
full manual (PDF) available from:
http://www.giga-byte.com/Motherboard/Support/Manual/Manual_GA-7N400-L1.htm)
According to the specs (I'm trying to dig what I suspect to be
the important parts, I'm a total newbie when it comes to such level
of hardware):
1. Chipset is nVidia nForce2 SPP Memory/AGP/PCI controller (PAC)
2. nVidia nForce2 MCP Integrated Peripheral Controller (PSIPC)
3. On-Board sound:
   3.1 Realtek ALC650 CODEC
   3.2 Line Out/2 front speaker
   3.3 Line In/2 rear speaker (by s/w switch)
   3.4 Mic In/center & subwoofer (by s/w switch)
   3.5 SPDIF In/Out
   3.6 CD In/AUC In/Game port
(I'm just copying from the specs, 95% of the above doesn't make any
sense to me).
I tried to install alsa-source 0.9.6 on debian unstable (because that's
what's available as a debian package). kernel modules compilation works
fine (with "ALSA_CARDS=all" for a start) "alsa start" recognizes the
sound card as requiring the intel8x00 driver.
The output of "lsmod" is:
-
Module  Size  Used byNot tainted
snd-seq-oss29632   0  (unused)
snd-seq-midi-event  3584   0  [snd-seq-oss]
snd-seq38384   2  [snd-seq-oss snd-seq-midi-event]
snd-pcm-oss38436   0  (unused)
snd-mixer-oss  13304   0  [snd-pcm-oss]
snd-intel8x0   18916   0
snd-pcm60868   0  [snd-pcm-oss snd-intel8x0]
snd-timer  15140   0  [snd-seq snd-pcm]
snd-ac97-codec 40888   0  [snd-intel8x0]
snd-page-alloc  6164   0  [snd-intel8x0 snd-pcm]
snd-mpu401-uart 3600   0  [snd-intel8x0]
snd-rawmidi14080   0  [snd-mpu401-uart]
snd-seq-device  4224   0  [snd-seq-oss snd-seq snd-rawmidi]
snd30212   0  [snd-seq-oss snd-seq-midi-event 
snd-seq snd-pcm-oss snd-mixer-oss snd-intel8x0 snd-pcm snd-timer 
snd-ac97-codec snd-mpu401-uart snd-rawmidi snd-seq-device]


relevant parts of "lspci -vvv" are:


00:01.0 ISA bridge: nVidia Corporation nForce2 ISA Bridge (rev a4)
Subsystem: Giga-byte Technology: Unknown device 0c11
Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- 
ParErr- Stepping- SERR
- FastB2B-
Status: Cap+ 66Mhz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- 

SERR- 
Latency: 0
Capabilities: [48] #08 [01e1]

00:01.1 SMBus: nVidia Corporation nForce2 SMBus (MCP) (rev a2)
Subsystem: Giga-byte Technology: Unknown device 0c11
Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- 
ParErr- Stepping- SERR
- FastB2B-
Status: Cap+ 66Mhz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- 

SERR- 
Interrupt: pin A routed to IRQ 5
Region 0: I/O ports at c000 [size=32]
Capabilities: [44] Power Management version 2
Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA 
PME(D0-,D1-,D2-,D3hot+,D3cold+)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-

00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97 
Audio Controler (MCP)
 (rev a1)
Subsystem: nVidia Corporation: Unknown device 4144
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- 
ParErr- Stepping- SERR
- FastB2B-
Status: Cap+ 66Mhz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- 

SERR- 
Latency: 0 (500ns min, 1250ns max)
Interrupt: pin A routed to IRQ 11
Region 0: I/O ports at b000 [size=256]
Region 1: I/O ports at b400 [size=128]
Region 2: Memory at e0001000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [44] Power Management version 2
Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA 
PME(D0-,D1-,D2-,D3hot-,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-

---

simply running "aplay" with this setup or any other sound program
results in the program playing as if everything is ok but no sound
comes out.
I ran alsamixer and unmuted and raised the volume for all channels.
What else can I do?  Is this hardware supposed to be supported?

Thanks very much,

--Amos



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Re: [Alsa-user] Support for Gigabyte GA-7N400-L1 mobo with Realtek ALC650?

[alsa-user]

Grégoire wrote:
Le dim 26/10/2003 à 18:45, [EMAIL PROTECTED] a écrit :

Hello,


1. Chipset is nVidia nForce2 SPP Memory/AGP/PCI controller (PAC)
2. nVidia nForce2 MCP Integrated Peripheral Controller (PSIPC)


What else can I do?  Is this hardware supposed to be supported?


I've the same chipset and the same problem.
I found that such software as xmms alsaplay are more or less able to
unmute the volume
Thanks.

Which cards do you have? Which modules are loaded?
Have you tried with aplay?
Anything special you had to do to make xmms alsaplay work?
Thanks,

--Amos



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Re: [Alsa-user] Support for Gigabyte GA-7N400-L1 mobo with Realtek ALC650?

[alsa-user]

Grégoire wrote:

I see I have the soundcore & not u.. this is a module I got with the
kernel
Right, but from what I see in 
kernel-source/2.4.22/Documentation/Configure.help this module is enabled
with CONFIG_SOUND and with ISA PnP cards, which I don't have.
I tried enabling ISA PnP compile but it doesn't fit on my machine.


Have you tried with aplay?


I don't have any wav one my machine & i don't konw if it knows mp3 (i
doubt)
Ah right, I testted it with common sounds from KDE.



Anything special you had to do to make xmms alsaplay work?
run xmms
or run alsaplay..
Tried these, didn' help :-(

Thanks,

--Amos



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Re: [Alsa-user] ALSA configuration for Realtek ALC 655

[alsa-user]

On Wed, Oct 29, 2003 at 09:31:29AM +0100, Frans Ketelaars wrote:
> On Tuesday 28 October 2003 13:45, Rafael Paoliello Guimaraes wrote:
> > Hello Folks,
> >
> > I own a Gigabyte Motherboard that came with the Realtek ALC 655 sound
> > chipset. I have installed the alsa drivers (version 0.9.8) but I
> > still don't know how I should configure my modules.conf? Anybody can
> > help me???
> >
> > Best regards,
> 
> You can check with 'lspci' which Multimedia audio controller you have. 
> If it's VIA8235 or something like that instructions are here:
> http://www.alsa-project.org/alsa-doc/doc-php/template.php?company=VIA&card=&chip=via8235&module=via82xx

At least in the output of "lspci -vvv" as root, there is no "via" mentioned
anywere.

Here are three lines which might be relevant:

00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio Contr
oler (MCP) (rev a1)
Subsystem: nVidia Corporation: Unknown device 4144
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Step
ping- SERR- FastB2B-

Could the secret lay behind the "Unknown device 4144"?

> 
> HTH,
> 
> -Frans
> 

Thanks,
--Amos


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Re: [Alsa-user] ALSA configuration for Realtek ALC 655

[alsa-user]

Frans Ketelaars wrote:


Ok, then here are the instructions to set up ALSA:
http://www.alsa-project.org/alsa-doc/doc-php/template.php?company=Nvidia&card=nForce&chip=NM2360&module=intel8x0
Hi,

I followed this page but I keep failing to get a "soundcore" module
file.
I have CONFIG_SOUND=y in my .config, and I see "LD sound/soundcore.o"
in the output of "make-kpkg".
I suspect that this "LD" means that soundcore is compiled into the
kernel, is this enough?  If not then how can I cause "soundcore.o"
(or "soundcore.ko", since I just switched to 2.6.0test9) to be
included?
I already have:

# OSS/Free portion
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
in my modules.conf.

Thanks,

--Amos



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Re: [Alsa-user] ALSA configuration for Realtek ALC 655

[alsa-user]

Frans Ketelaars wrote:
AFAIK CONFIG_SOUND=y means that soundcore is compiled into the kernel 
and everything should be ok.
That's what I suspected.
So I have it in my kernel but still it doesn't help.
Anything else I can try?

Thanks,

--Amos



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[Alsa-user] VX Pocket V1 soundcard

[alsa-user]

Title: Message



Hello 
All,
 
I've been trying to 
get my Digigram VX Pocket soundcard working with ALSA on Mandrake 9.1 with not 
much success. I know that a version 2 card works but mine is the older 
version 1. Has anyone managed to get this version to work?
 
I compiled and 
installed the drivers with no problem, following the instructions from http://www.alsa-project.org/alsa-doc/doc-php/template.php?company=Digigram&card=VXpocket+V2&chip=&module=vxpocket. 
However when I come to start alsamixer, or any other mixer, it tells me that no 
card has been detected. Neither does it appear in the /proc files or /dev. 
A similar thing happens when I try to run vxloader. 
 
Cheers!
 
Alistair 
Matthews 

RE: [Alsa-user] VX Pocket V1 soundcard

[alsa-user]

Hi Alain,

Thanks very much for this ~ it worked like a charm! I had messed around
with the /etc/pcmcia files but obviously messed it up.

Cheers!

Alistair.


-Original Message-
From: Alain Crétet [mailto:[EMAIL PROTECTED] 
Sent: 07 January 2004 15:27
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: RE : [Alsa-user] VX Pocket V1 soundcard



Hi Alistair Matthews,

Apparently VXpocket V1 and VXpocket V2 have the same firmware and CSID.
Your VXpocket V1 could work with the ALSA driver, but I have not tested
it personnaly.

Did you copy /usr/src/alsa-driver-XXX/pcmcia/vx/vxpocket.conf (or
vxp440.conf) to /etc/pcmcia, and restart pcmcia service?

Regards,

Alain

> --
> De :  [EMAIL PROTECTED]
> Envoyé :  Wednesday, January 7, 2004 3:51 PM
> À :   [EMAIL PROTECTED]
> Objet :   [Alsa-user] VX Pocket V1 soundcard
> 
> Hello All,
>  
> I've been trying to get my Digigram VX Pocket soundcard working with 
> ALSA on Mandrake 9.1 with not much success. I know that a version 2
card works but mine is the older version 1. Has anyone managed to get
this version to work?
>  
> I compiled and installed the drivers with no problem, following the 
> instructions from
http://www.alsa-project.org/alsa-doc/doc-php/template.php?company=Digigr
am&card=VXpocket+V2&chip=&module=vxpocket. However when I come to start
alsamixer, or any other mixer, it tells me that no card has been
detected. Neither does it appear in the /proc files or /dev. A similar
thing happens when I try to run vxloader.
>  
> Cheers!
>  
> Alistair Matthews
> 





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Re: [Alsa-user] User lists v forums

[alsa-user-admin]

On Mon, May 22, 2006 at 03:27:54PM +1000, Terry North wrote:
> Is there any support for a forum as an alternative to
> the user list, which is such a clumsy, fragmented and
> disorganised format?

It may seem like that to you, but mailing lists have a 
long history, with excellent tools exist to manage groups
of mail messages. Mailing lists are structured enough
that they are the preferred medium for many software
developers, including the linux kernel.

Online you can find wikis with distilled wisdom from 
the mailing lists, however I think many here will agree 
that beer is healthier than spirits; one can drink 
much more in a day, with fewer ill effects.

See http://www.alsa-project.org/ for wikis and 
bug-reporting tools.

> To whom should one address a proposal for a forum?

His Highness Higram Bumfeld III, King of the Alsa Project :-p

If you have a specific problem you may like to 
check the mailing list archives before posting 
to the list.

See http://www.alsa-project.org/mailing-lists.php

-- 
Joel Roth


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[Alsa-user] Sound problems in PCCHIPS A13G+

[alsa-user . 20 . pris]

[Athlon64/Opteron] DRAM Controller
00:18.3 Host bridge: Advanced Micro Devices [AMD] K8
[Athlon64/Opteron] Miscellaneous Control

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[Alsa-user] volume mix several apps in software?

[akb+lists . alsa-user]

Hi--

I have several different programs outputting sound at the same time.
I believe my sound card does not do hardware mixing. 

How do I make say, a game quieter 
and a music player louder at the same time, 
if the apps do not have internal volume controls?


Configs Info:

*  debian Linux
*  debian alsa-base package version 1.0.10-3
*  my own 2.6.16 kernel
*  driver version 1.0.11rc2 (Wed Jan 04 08:57:20 2006 UTC)
*  there is no /etc/asound.conf, and no ~/.asoundrc


*  /proc/asound/cards:
   0 [CK8]: NFORCE - NVidia CK8
NVidia CK8 with ALC658D at 0xee005000, irq 19

*  /proc/asound/devices:
0: [ 0]   : control
   16: [ 0- 0]: digital audio playback
   18: [ 0- 2]: digital audio playback   (digital out? pcm says IEC958)
   24: [ 0- 0]: digital audio capture
   25: [ 0- 1]: digital audio capture
   33:: timer

*  lsmod:
snd_intel8x0   34332  7 
snd_ac97_codec 96736  1 snd_intel8x0
snd_ac97_bus2368  1 snd_ac97_codec
snd_pcm93000  4 snd_intel8x0,snd_ac97_codec
snd_timer  25860  4 snd_pcm
snd56420  13 snd_intel8x0,snd_ac97_codec,
  snd_pcm,snd_timer
soundcore  10528  1 snd
snd_page_alloc 11272  2 snd_intel8x0,snd_pcm


*  fuser -v /dev/snd/* says:

   USERPID ACCESS COMMAND
  /dev/snd/controlC0   jones  4690 f  gkrellm
   jones 20115 f  firefox-bin
  /dev/snd/pcmC0D0cjones 17717 f...m  firefox-bin
   jones 20115 f...m  firefox-bin
  /dev/snd/pcmC0D0pmusic  4368 f...m  esd
   music  4373 f...m  esd
   jones 17717 m  firefox-bin
   jones 20115 f...m  firefox-bin
  /dev/snd/timer   music  4368 f  esd
   jones 20115 f  firefox-bin


  --thanks!
  --akb



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[Alsa-user] config questions, definitions

[akb+lists . alsa-user]

Hi--

I was looking over the config data I just sent,
and I'd love some explanations.

(1) what do soundcore and snd do? 

I assume one provides the ALSA api, but what about the other?

(2) why do some apps grab a control channel and others not?
(2A) what's a control channel control, anyway?

Example:

I have fuser(1) output that includes: 
   USERPID ACCESS COMMAND 
  /dev/snd/controlC0   jones  4690 f  gkrellm 
   jones 20115 f  firefox-bin 
  /dev/snd/pcmC0D0pmusic  4368 f...m  esd 
   music  4373 f...m  esd 
   jones 17717 m  firefox-bin 

now, gkrellm adjusts volume, so I assume that's one thing you need
a control channel for (unless you are going to drop the volume within
the pcm stream). 

why doesn't esd have a control channel?

do control channels just control out of band info like volume?
do control channels have anything to do with 
  what encoding/sample rate the chipset expects? 
do control channels control hw mixing? 

  --thanks more
  --akb



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[Alsa-user] dmix diagnostics

[akb+lists . alsa-user]

if I can issue two aplay's simultaneously,
and hear both sounds playing,
does that mean dmix is working correctly?

system has cheap motherboard sound (nforce2), 
and thus no hardware mixer. 


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[Alsa-user] jackd and artsd questions

[akb+lists . alsa-user]

When I try to start jackd ("jackd -d alsa"), I get:

  loading driver ..
  creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
  control device hw:0
  the playback device "hw:0" is already in use. Please stop the
application using it and run JACK again

Why isn't it using dmix? 

Do I need to feed it the dmix device name with 
  "jackd -d alsa -P " or something similar?

If so, what is the device name? 

And can I feed the same device name to artsd?


Configs:

I have alsa 1.0.11rc2, no asoundrc's,
nforce2 motherboard sound, snd_intel8x0 driver, 
debian linux (mixed sarge/etch) on intel, 2.6.16 kernel, 

aplay -L says:
   List of PLAYBACK Hardware Devices 
  card 0: CK8 [NVidia CK8], device 0: Intel ICH [NVidia CK8]
Subdevices: 0/1
Subdevice #0: subdevice #0
  card 0: CK8 [NVidia CK8], device 2: Intel ICH - IEC958 [NVidia CK8 - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0
  
aplay -l output is 165 lines long, so I won't include it here. 

  --thanks for any clues...
  --akb


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[Alsa-user] Boss BR-80

[Benjamin via Alsa-user]

Hi everyone!

Has anybody managed to get this device working as a Linux soundcard? 
It's being recognised and displayed but neither jack nor ALSA seem to be 
able to use it.


alsamixer doesn't show any controls for the device and jack won't run, I 
only get the following response:


Hi everyone!

Has anybody managed to get this device working as a Linux soundcard? 
It's being recognised and displayed but neither jack nor ALSA seem to be 
able to use it.


|jackd -d alsa --device hw:1
jackdmp 1.9.11
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2016 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
no message buffer overruns
no message buffer overruns
no message buffer overruns
JACK server starting in realtime mode with priority 10
self-connect-mode is "Don't restrict self connect requests"
audio_reservation_init
Acquire audio card Audio1
creating alsa driver ... hw:1|hw:1|1024|2|48000|0|0|nomon|swmeter|-|32bit
Using ALSA driver USB-Audio running on card 1 - Roland BR-80(AUDIO) at 
usb-:00:1d.7-2, high speed

configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods
ALSA: final selected sample format for capture: 32bit integer little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit integer little-endian
ALSA: use 2 periods for playback
ALSA: poll time out, polled for 34828045 usecs
JackAudioDriver::ProcessAsync: read error, stopping...
^@^CJack main caught signal 2
Released audio card Audio1
audio_reservation_finish


Thank you for any wisdom here!

Ben
|
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Re: [Alsa-user] Boss BR-80

[Benjamin via Alsa-user]

Hi Clemens,

thanks for your reply!

aplay or arecord doesn't work. (See code below)

Cheers,


Ben




benjamin@microlab:~$ aplay -D hw:CARD=BR80AUDIO,DEV=0 pinknoise.wav
Wiedergabe: WAVE 'pinknoise.wav' : Signed 16 bit Little Endian, Rate: 
44100 Hz, mono

aplay: set_params:1299: Sample-Format nicht unterstützt
Available formats:
- S32_LE
benjamin@microlab:~$ arecord -D hw:CARD=BR80AUDIO,DEV=0 bluenoise.wav
Aufnahme: WAVE 'bluenoise.wav' : Unsigned 8 bit, Rate: 8000 Hz, mono
arecord: set_params:1299: Sample-Format nicht unterstützt
Available formats:
- S32_LE
benjamin@microlab:~$


On 03.07.2017 10:08, Clemens Ladisch wrote:

Benjamin via Alsa-user wrote:

ALSA: poll time out, polled for 34828045 usecs

Apparently, this device is not strictly compatible with the USB Audio
specification and requires some vendor-specific command(s) to work.

Does aplay or arecord alone work?


Regards,
Clemens

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Re: [Alsa-user] Boss BR-80

[Benjamin via Alsa-user]

to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.267521] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.267562] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.267597] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.267986] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.269842] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.270781] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.270873] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.270957] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.271384] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.271790] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.274328] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.274468] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.274594] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.277596] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.278289] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.279531] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.279572] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.279608] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.281504] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.282009] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.283156] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.283197] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.283232] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.283624] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.283986] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.287035] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.287129] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.287213] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.287715] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.289404] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.290670] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.290808] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.290935] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.291433] usb 3-1: Unable to 
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.291910] usb 3-1: Unable to 
change format on ep #8e: already in use



On 08.07.2017 18:33, Stefan Sauer wrote:

tail -f /var/log/messages



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Re: [Alsa-user] Boss BR-80

[Benjamin via Alsa-user]

alright! Thank you!


On 10.07.2017 21:53, Stefan Sauer wrote:

On 07/09/2017 12:49 PM, Benjamin via Alsa-user wrote:

Thank you Stefan!


When I plug it in, I get the following:

Jul  9 12:43:57 microlab kernel: [26915.544039] usb 3-1: new
high-speed USB device number 2 using ehci-pci
Jul  9 12:43:57 microlab kernel: [26915.695264] usb 3-1: New USB
device found, idVendor=0582, idProduct=0130
Jul  9 12:43:57 microlab kernel: [26915.695267] usb 3-1: New USB
device strings: Mfr=1, Product=2, SerialNumber=0
Jul  9 12:43:57 microlab kernel: [26915.695269] usb 3-1: Product:
BR-80(AUDIO)
Jul  9 12:43:57 microlab kernel: [26915.695270] usb 3-1: Manufacturer:
Roland
Jul  9 12:43:57 microlab mtp-probe: checking bus 3, device 2:
"/sys/devices/pci:00/:00:1d.7/usb3/3-1"
Jul  9 12:43:57 microlab mtp-probe: bus: 3, device: 2 was not an MTP
device
Jul  9 12:43:57 microlab kernel: [26916.206685] usbcore: registered
new interface driver snd-usb-audio
Jul  9 12:43:58 microlab kernel: [26917.143903] usb 3-1: Unable to
change format on ep #8e: already in use

Yes it is those. I get them too and spent once a night to understand
what is going on. Unfortunately one seems to know a lot about usb to
make any sense out of it (ep is "end-point").

Stefan


Jul  9 12:43:58 microlab kernel: [26917.143986] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.144080] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.144620] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.145000] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.146531] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.146576] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.146634] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.147069] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.147485] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.150170] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.150268] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.150353] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.150905] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.151337] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.156123] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.156282] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.157840] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.158384] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.158936] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.161557] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.161621] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.161659] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.162069] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.162447] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.168128] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.168185] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.168221] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.168639] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.169006] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.174697] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.174835] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.174923] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.175421] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.177743] usb 3-1: Unable to
change format on ep #8e: already in use
Jul  9 12:43:58 microlab kernel: [26917.179765] usb 3-1: Unable to
change forma

Re: [Alsa-user] Boss BR-80

[Benjamin via Alsa-user]

Hi Clemens hi everyone,

the guys over at linux musicians forum mentioned that it used to work in 
older kernels and then not any more:


https://linuxmusicians.com/viewtopic.php?p=83159#p83159

Does anyone know about this?

Thanks, Ben


On 07.07.2017 09:12, Benjamin wrote:

Hi Clemens,

thanks for your reply!

aplay or arecord doesn't work. (See code below)

Cheers,


Ben




benjamin@microlab:~$ aplay -D hw:CARD=BR80AUDIO,DEV=0 pinknoise.wav
Wiedergabe: WAVE 'pinknoise.wav' : Signed 16 bit Little Endian, Rate: 
44100 Hz, mono

aplay: set_params:1299: Sample-Format nicht unterstützt
Available formats:
- S32_LE
benjamin@microlab:~$ arecord -D hw:CARD=BR80AUDIO,DEV=0 bluenoise.wav
Aufnahme: WAVE 'bluenoise.wav' : Unsigned 8 bit, Rate: 8000 Hz, mono
arecord: set_params:1299: Sample-Format nicht unterstützt
Available formats:
- S32_LE
benjamin@microlab:~$


On 03.07.2017 10:08, Clemens Ladisch wrote:

Benjamin via Alsa-user wrote:

ALSA: poll time out, polled for 34828045 usecs

Apparently, this device is not strictly compatible with the USB Audio
specification and requires some vendor-specific command(s) to work.

Does aplay or arecord alone work?


Regards,
Clemens

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Re: [Alsa-user] Fwd: RE: RE: inventory

[Benjamin via Alsa-user]

Oh sorry that last email was a mistake! Could you please remove that! 
Thank you!


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[Alsa-user] boss problems solved

[Benjamin via Alsa-user]

Hi everyone,


it seems that the problems for BOSS RC-300 and BR-80 have been solved. 
https://linuxmusicians.com/viewtopic.php?f=6&t=17180&p=86347&hilit=boss#p86347


How can these fixes be implemented into the official ALSA? Would be 
great if users didn't have to make their own kernel modules..


Thanks for any insights here!


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[Alsa-user] Tascam US-800

[Benjamin via Alsa-user]

Hi there,

has anybody had any luck recently getting the said interface running? 
I'm trying in AV-Linux 2018 but I only get errors when I try to use it.


It's said on forums to have been working for others in Ubuntu but this 
is a few years ago,


Thanks a lot for any insights,

Ben


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[Alsa-user] Tascam US-800 Problem

[Benjamin via Alsa-user]

9.063033] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x801, wIndex = 0x1f00, type = 4
[11389.063041] usb 3-1: 31:0: cannot get min/max values for control 8 
(id 31)
[11389.221248] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x901, wIndex = 0x1f00, type = 4
[11389.221255] usb 3-1: 31:0: cannot get min/max values for control 9 
(id 31)
[11389.372035] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0xa01, wIndex = 0x1f00, type = 4
[11389.372044] usb 3-1: 31:0: cannot get min/max values for control 10 
(id 31)
[11389.501544] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x101, wIndex = 0x2000, type = 4
[11389.501554] usb 3-1: 32:0: cannot get min/max values for control 1 
(id 32)
[11389.636184] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x201, wIndex = 0x2000, type = 4
[11389.636194] usb 3-1: 32:0: cannot get min/max values for control 2 
(id 32)
[11389.748948] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x301, wIndex = 0x2000, type = 4
[11389.748957] usb 3-1: 32:0: cannot get min/max values for control 3 
(id 32)
[11389.867806] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x401, wIndex = 0x2000, type = 4
[11389.867816] usb 3-1: 32:0: cannot get min/max values for control 4 
(id 32)
[11389.963344] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x501, wIndex = 0x2000, type = 4
[11389.963354] usb 3-1: 32:0: cannot get min/max values for control 5 
(id 32)
[11390.059938] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x601, wIndex = 0x2000, type = 4
[11390.059946] usb 3-1: 32:0: cannot get min/max values for control 6 
(id 32)
[11390.175805] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x701, wIndex = 0x2000, type = 4
[11390.175813] usb 3-1: 32:0: cannot get min/max values for control 7 
(id 32)
[11390.265410] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x801, wIndex = 0x2000, type = 4
[11390.265417] usb 3-1: 32:0: cannot get min/max values for control 8 
(id 32)
[11390.362211] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x901, wIndex = 0x2000, type = 4
[11390.362220] usb 3-1: 32:0: cannot get min/max values for control 9 
(id 32)
[11390.465310] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0xa01, wIndex = 0x2000, type = 4
[11390.465320] usb 3-1: 32:0: cannot get min/max values for control 10 
(id 32)
[11391.410036] usb 3-1: cannot get ctl value: req = 0x83, wValue = 
0x701, wIndex = 0x1e00, type = 4
[11391.410044] usb 3-1: 30:0: cannot get min/max values for control 7 
(id 30)
[11391.545018] usb 3-1: cannot get ctl value: req = 0x81, wValue = 
0x701, wIndex = 0x1e00, type = 4

[11391.677362] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.681750] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.686703] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.691946] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.697321] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.703325] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.707567] usb 3-1: 1:1: usb_set_interface failed (-71)
[11391.711825] usb 3-1: 1:1: usb_set_interface failed (-71)


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[Alsa-user] MOTU Microbook iic support

[jidder--- via Alsa-user]

Hi!

Anyone that could help getting the "Class compliant" Motu Microbook iic to run 
under Linux?

/d
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Re: [Alsa-user] MOTU Microbook iic support

[jidder--- via Alsa-user]

Hi Clemens,

See here:
http://www.alsa-project.org/db/?f=46cd3f0c6ecf553222ac2bcb3e4d536dab48f613

Alsa-mixer is empty,
on macOS (without any drivers installed), I can see the number of inputs and 
outputs 
incl. channel names. Input: (Mic, Guitar, Line 1-2)
Output: (Main 1-2, Line 1-2, Phones 1-2)


best David

> On 28 Jul 2018, at 22:12, Clemens Ladisch via Alsa-user 
>  wrote:
> 
> jidder--- wrote:
>> Anyone that could help getting the "Class compliant" Motu Microbook iic to 
>> run under Linux?
> 
> In theory: plug it in.
> 
> If something different happens in practice, you have to tell us about it.
> 
> 
> Regards,
> Clemens
> 
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Re: [Alsa-user] MOTU Microbook iic support

[jidder--- via Alsa-user]

Anyone that's willing to help out here to make a patch getting this soundcard 
running under linux.

> On 30 Jul 2018, at 11:42, jidder--- via Alsa-user 
>  wrote:
> 
> Hi Clemens,
> 
> See here:
> http://www.alsa-project.org/db/?f=46cd3f0c6ecf553222ac2bcb3e4d536dab48f613
> 
> Alsa-mixer is empty,
> on macOS (without any drivers installed), I can see the number of inputs and 
> outputs 
> incl. channel names. Input: (Mic, Guitar, Line 1-2)
> Output: (Main 1-2, Line 1-2, Phones 1-2)
> 
> 
> best David
> 
>> On 28 Jul 2018, at 22:12, Clemens Ladisch via Alsa-user 
>>  wrote:
>> 
>> jidder--- wrote:
>>> Anyone that could help getting the "Class compliant" Motu Microbook iic to 
>>> run under Linux?
>> 
>> In theory: plug it in.
>> 
>> If something different happens in practice, you have to tell us about it.
>> 
>> 
>> Regards,
>> Clemens
>> 
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Re: [Alsa-user] MOTU Microbook iic support

[jidder--- via Alsa-user]

Any clues on how to get this up&running on linux?

http://www.alsa-project.org/db/?f=46cd3f0c6ecf553222ac2bcb3e4d536dab48f613

Alsa-mixer is empty,
on macOS (without any drivers installed), I can see the number of inputs and 
outputs 
incl. channel names. Input: (Mic, Guitar, Line 1-2)
Output: (Main 1-2, Line 1-2, Phones 1-2)


best David

> On 28 Jul 2018, at 22:12, Clemens Ladisch via Alsa-user 
>  wrote:
> 
> jidder--- wrote:
>> Anyone that could help getting the "Class compliant" Motu Microbook iic to 
>> run under Linux?
> 
> In theory: plug it in.
> 
> If something different happens in practice, you have to tell us about it.
> 
> 
> Regards,
> Clemens
> 
> --
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> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
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[Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Paul via Alsa-user]

Hi,

I have Ryzen 5 2400G CPU with built-in Vega 11 GPU - checking out 
available codecs on Fedora 29 (kernel 4.19.10) I get the following:


Codec: ATI R6xx HDMI
Address: 0
AFG Function Id: 0x1 (unsol 0)
Vendor Id: 0x1002aa01
Subsystem Id: 0x00aa0100
Revision Id: 0x100700
No Modem Function Group found
Default PCM:
    rates [0x70]: 32000 44100 48000
    bits [0x2]: 16
    formats [0x5]: PCM AC3


When will support for 24/32 bit and 96/192KHz audio playback be 
available for this GPU?



Thanks.



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[Alsa-user] Can't play mono file using HDMI device explicitly, but can do it if HDMI is default

[curious_lurker via Alsa-user]

Hello. I want to be able to play mono sound using HDMI device. Strange thing is 
I can play mono file when HDMI is selected as default in .asoundrc, but I can 
not do it if I use device explicitly in aplay.

I play file like this:

localhost ~ # aplay -D hdmi:CARD=HDMI,DEV=2 testmono.wav
Playing WAVE 'testmono.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
aplay: set_params:1345: Channels count non available

localhost ~ # aplay -D default testmono.wav
Playing WAVE 'testmono.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono

My .asoundrc looks like this:
defaults.pcm.card 0
defaults.pcm.device 8

Output of aplay -L and -l:

localhost ~ # aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
hdmi:CARD=HDMI,DEV=0
HDA Intel HDMI, HDMI 0
HDMI Audio Output
hdmi:CARD=HDMI,DEV=1
HDA Intel HDMI, HDMI 1
HDMI Audio Output
hdmi:CARD=HDMI,DEV=2
HDA Intel HDMI, HDMI 2
HDMI Audio Output
hdmi:CARD=HDMI,DEV=3
HDA Intel HDMI, HDMI 3
HDMI Audio Output
hdmi:CARD=HDMI,DEV=4
HDA Intel HDMI, HDMI 4
HDMI Audio Output
default:CARD=PCH
HDA Intel PCH, ALC662 rev1 Analog
Default Audio Device
sysdefault:CARD=PCH
HDA Intel PCH, ALC662 rev1 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
Front speakers
surround21:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, ALC662 rev1 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers

localhost ~ # aplay -l
 List of PLAYBACK Hardware Devices 
card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA Intel HDMI], device 10: HDMI 4 [HDMI 4]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

What am I doing wrong? Can I configure ALSA so it can play mono file with 
'hdmi:CARD=HDMI,DEV=2'?
If there is not enough information here, please tell me what else is needed.

Thanks!


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[Alsa-user] Crackling audio during acquisition

[rodomar705--- via Alsa-user]

I have audio stutters with an ALC892 Realtek Chipset on a B450 GAMING ELITE 
board from Gigabyte. No issues with Windows.

After opening an issue and testing with a PulseAudio developer, he has 
determined that the issue is not within PulseAudio, but on ALSA.

There are already two issues about that on the bug tracker on Linux, namely 
this two:

ALC892: https://bugzilla.kernel.org/show_bug.cgi?id=203351
ALC1220: https://bugzilla.kernel.org/show_bug.cgi?id=195303

To me both appears to be identical, for this reason even if I have a ALC892 
I’ve posted on the second one, just because it had more posts.

The particular thing is that acquiring with PulseAudio on Audacity show the 
problem, however using ALSA directly the issue is not present. Trying using 
arecord still present the issue from PulseAudio.

Tried to play with the parameters /sys/devices/audio to no avail. Compared the 
pinconfigs from Windows, they are identical.

Can someone try to help me fix the issue?

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[Alsa-user] Support for Pioneer DJ DJM-900NXS2 over USB

[Pander via Alsa-user]

Hi all,

With which version of Linux kernel and ALSA is Pioneer DJ DJM-900NXS2
over USB fully supported?

I am not sure in which release is:

https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/tree/sound/usb/mixer_quirks.c#n2825

and

https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/tree/sound/usb/quirks-table.h#n3646

which offer to 10 playback channels and 12 capture channels.

I'm using Ubuntu 20.10 which lists it as USB device with playback and
capture. Both aplay -l -L and alsamixer reports this. However, I can't
seem to use it. A black screen without controls is shown in alsamixer.
Sending or receiving audio is not possible when selecting the device in
gnome-control. Only the device is listed, not the individual channels.

The versions for 20.10 are:

Linux 5.8.0-7630-generic
alsa-base 1.0.25+dfsg-0ubuntu5
alsa-topology-conf 1.2.3-1
alsa-ucm-conf 1.2.2-1ubuntu5.1
alsa-utils 1.2.3-1ubuntu1

For Ubuntu 21.04, these version are expected:

Linux 5.11.0-11.12 /  5.11.0-13.14
alsa-base 1.0.25+dfsg-0ubuntu7
alsa-topology-conf  1.2.4-1
alsa-ucm-conf 1.2.4-2ubuntu1
alsa-utils 1.2.4-1ubuntu2

Will that solve it?

thanks,

Pander



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[Alsa-user] "alsactl restore" does not load the driver state

[YuGiOhJCJ via Alsa-user]

Hello,

When I type "alsactl restore", it does not load the driver state.

Steps to reproduce:
1. Open a first terminal dedicated to alsa-mixer
2. Open a second terminal where you will type the following commands
3. In the first terminal, press F4, set the audio volume for capture to 40
4. In the second terminal, type "sudo alsactl store" in order to save the 
driver state to /var/lib/alsa/asound.state
5. In the first terminal, set the audio volume for capture to 45
6. In the second terminal, type "sudo alsactl restore" in order to load the 
driver state to /var/lib/alsa/asound.state

Expected results: I should see in the first terminal that the audio volume for 
capture is restored to 40.

Current results: I see in the first terminal that the audio volume for capture 
is not restored to 40. It is still set to 45!

How to fix that please?

Thank you.
Best regards.

About my system:

Packages:
- alsa-lib-1.2.9
- alsa-plugins-1.2.7.1
- alsa-utils-1.2.9

Operating system: Slackware 14.2 (64-bit)

Architecture: x86_64

Kernel: Linux 5.18.10


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[Alsa-user] No sound in headphone on Cherry trail with rt5645

[Daimon Wang via Alsa-user]

Hi,
I've got a CHT Z8350 device with a 4.4 kernel (binary). Sound do work there 
with some amixer settings. But it has other issues on display. 
For that, I'm trying to upgrade the kernel and the display issue is gone. But 
the sound is broken.

With kernel 4.11, the rt5645 is detected successful, and volume is configurable 
in pavucontrol, but the headphone is silent. I achieve this by disable the 
snd_hdmi_lpe_audio. Before that HDMI was always the default card.

I've attached alsa-info form the 2 kernel (suffix with 440 for old kernel, 411 
for new).The main difference I can find is the device node name is different, 
among may other difference in value.
Please let me know if any other info are needed.
Regards,Daimon

alsa-info.tgz
Description: Binary data
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Re: [Alsa-user] A soundcard to record IEC958_SUBFRAME from MiniDisc

[Clemens Ladisch via Alsa-user]

Mark Hills wrote:
> a soundcard that has toslink input, and supports IEC958_SUBFRAME_*
> sample formats

In practice, PCI cards with a CMI8738/CMI8768 (not CMI878x) chip.


Regards,
Clemens

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Re: [Alsa-user] Advice on snd_pcm_writei sizes

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
> If my software naturally produces samples in 64-frame chunks, but the
> device is set up to have 256-frame periods, is there any advantage to
> accumulating four chunks in my own buffer and writing entire periods,
> versus writing each chunk separately and expecting to be blocked every
> fourth time?

No.  And writing samples to the buffer earlier will slightly decrease
the risk of an underrun.

> Alternatively, would the extra overhead of reducing the period size to
> 64 (96KHz stereo 32-bit LE on a Raspberry Pi 3) be significant?

That is something you have to measure yourself ...


Regards,
Clemens

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Re: [Alsa-user] Problem reading back periods

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
> If I open my sound output device (Intel HDA on Intel mobo), set its
> parameters as follows:
>
> access = SND_PCM_ACCESS_RW_INTERLEAVED
> format = SND_PCM_FORMAT_S32_LE
> channels = 2
> rate = 48000
> periods = 4
> buffer_size = 1024
>
> it returns no errors. If I use snd_pcm_hw_params_get_periods on the
> parameters data structure, I get 4. But if I use snd_pcm_hw_params_current
> to refill it with the actual values from the device,
> snd_pcm_hw_params_get_periods returns 0.

In theory, this cannot happen.  Show the code.   Did you check for errors
with all function calls?


Regards,
Clemens

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[Alsa-user] Need help in getting ALSA source code

[Nagendra Kumar via Alsa-user]

Hi,  My project needs ALSA device driver integration. Can you please help 
in in getting:1. Demo application(may be sound recorder and player) using ALSA 
device driver.
2. in knowing function call traces about packets coming from sound card and 
being delivering to demo(during recording).3. in knowing function call traces 
about packets going from application to ALSA till sound card (during playing 
sound).

I just want to understand the function/call flow from/to application to/from 
sound card.
My project have requirement to change the packet flow and add some code and 
test in the packet flow path like delay the sound packet, modify the sound 
packet from app till sound card and vice versa.
Please kindly provide some pointers.
Thanks-Nagendra
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Re: [Alsa-user] Need help in getting ALSA source code

[Nagendra Kumar via Alsa-user]

Hi Experts,
   Any pointers ???
Thanks-Nagendra

  From: Nagendra Kumar via Alsa-user 
 To: "alsa-user@lists.sourceforge.net"  
 Sent: Thursday, 14 September 2017 8:04 PM
 Subject: [Alsa-user] Need help in getting ALSA source code
   
Hi,  My project needs ALSA device driver integration. Can you please help 
in in getting:1. Demo application(may be sound recorder and player) using ALSA 
device driver.
2. in knowing function call traces about packets coming from sound card and 
being delivering to demo(during recording).3. in knowing function call traces 
about packets going from application to ALSA till sound card (during playing 
sound).

I just want to understand the function/call flow from/to application to/from 
sound card.
My project have requirement to change the packet flow and add some code and 
test in the packet flow path like delay the sound packet, modify the sound 
packet from app till sound card and vice versa.
Please kindly provide some pointers.
Thanks-Nagendra
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Re: [Alsa-user] Slot assignment not working

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
>  options snd slots=snd-soc-hifiberry-dacplus,snd-usb-audio
>
>   midiC0D0  a USB MIDI device
>   midiC1D0  another USB MIDI device
>   pcmC2D0p and pcmC2D1p on-board crappy PWM audio
>   pcmC3D0p  Hifiberry DAC
>   pcmC4D0p  HDMI audio

Show your entire alsa.conf.


Regards,
Clemens

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Re: [Alsa-user] Slot assignment not working

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
>> Paul D. DeRocco wrote:
>>>  options snd slots=snd-soc-hifiberry-dacplus,snd-usb-audio
>>>
>>>   midiC0D0  a USB MIDI device
>>>   midiC1D0  another USB MIDI device
>>>   pcmC2D0p and pcmC2D1p on-board crappy PWM audio
>>>   pcmC3D0p  Hifiberry DAC
>>>   pcmC4D0p  HDMI audio
>>
>> Show your entire alsa.conf.
>
> I did. The one-liner you just quoted is all I used to need.

In theory, it should not be possible for snd-usb-audio to grab the first slot.
Are you sure there isn't any other "options snd" line in some .conf file?

Anyway, that slots options specifies only two slots.
If you have five devices, it's a better idea to specify five slots.


Regards,
Clemens

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Re: [Alsa-user] Slot assignment not working

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
>> Anyway, that slots options specifies only two slots.
>> If you have five devices, it's a better idea to specify five slots.
>
> What I have is three devices that always exist, one of which is the
> Hifiberry DAC that I want to use, and the others are the PWM audio and the
> HDMI audio. Then there are the unknown number of MIDI devices that someone
> might plug in. I tried specifying various cards_limit values.

cards_limit is not related with that.

> I tried the following, and nothing changed:
>
>   options snd cards_limit=8 \
> slots=snd-soc-hifiberry-dacplus,vc4,snd-bcm2835,snd-usb-audio

Are these drivers actually compiled as modules?
Are the names the same as in /proc/asound/modules?

> This is really behaving as though "snd" isn't the right device name to
> address the slots option to.

Check in /sys/module/snd/parameters/slots.


Regards,
Clemens

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Re: [Alsa-user] Slot assignment not working

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
> Can this parameter be supplied on the kernel command line as snd.slots=...?

Yes; this is possible for all parameters.


Regards,
Clemens

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Re: [Alsa-user] Slot assignment not working

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
> Does anyone know the name of the Raspberry Pi HDMI sound device? udevadm
> info shows a driver name of "vc4_hdmi" and an id attribute of "vc4hdmi".
> lsmod only lists "vc4", nothing else with "vc4" in it. alsamixer shows
> "vc4-hdmi". None of these names in the slots option made any difference.
> When I read /proc/asound/modules, I get this:
>
>  0 snd_soc_hifiberry_dacplus
>  2 snd_bcm2835
>  3 snd_usb_audio
>  4 snd_usb_audio
>  5 (null)

Looks like a bug in that driver.  (I don't know how the ASoC framework
handles this.)


Regards,
Clemens

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Re: [Alsa-user] Sound BlasterX G1, incorrect sample rate

[Clemens Ladisch via Alsa-user]

Aaron Jackson wrote:
> I bought a Sound BlasterX G1. The website says it supports 24/96
> but I can't get it to go any higher 16/44.1.

Many Creative devices require a vendor-specific command to enable
high speed.  You'd have to monitor what commands the Windows driver
sends, if you know how to do that.


Regards,
Clemens

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Re: [Alsa-user] Changing device while stream is running

[Clemens Ladisch via Alsa-user]

John Z. wrote:
> My general problem is: I'd like to be able to switch device that
> plays audio, while the audio stream is working - without player
> application knowing anything happened.

PulseAudio can do this.

> My current asoundrc can switch the card through environmental
> variable, and defaults to dmix (that then has hw:1,0 slave) if no
> variable is present.

There is no plugin that can do this dynamically, except "pulse".


Regards,
Clemens

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Re: [Alsa-user] Changing device while stream is running

[Clemens Ladisch via Alsa-user]

John Z. wrote:
>> There is no plugin that can do this dynamically, except "pulse".
>
> That's a bit unfortunate, as I've put a lot of effort to learn enough
> about alsa so that I can be able to remove PA from my system.

What is so bad about PA that an ALSA plugin would do better?

> I don't mind writing some code; do you think it'd be feasible to write a
> plugin to accomplish such functionality

Open both slave devices; copy around the samples appropriately.


Regards,
Clemens

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Re: [Alsa-user] aplay underrun length

[Clemens Ladisch via Alsa-user]

Samuele Carcagno wrote:
> I often get underruns

Is some other process blocking the CPU or disk I/O?

Try increasing the period size.

> a single underrun lasting about 122-ms would be negligible for my
> experiment

Please note that the stream is stopped, and restarted later.  Can you
actually live with all later samples being shifted in time?

> When aplay says the underrun is 'at least' x seconds, what does this
> mean?

It's the time between when the underrun was detected by the kernel's
interrupt handler and when this message is printed by aplay.  In other
words, aplay should have been scheduled earlier by this amount to be
able to write new samples in time.

> How large could the margin of error be in the estimate of the underrun
> duration?

You have to add the time needed to restart the stream.

> I assume this means that all underruns occuring during the playback of
> a single file will be reported separately?

Yes.


Regards,
Clemens

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Re: [Alsa-user] PCM control in alsamixer not working when using asound.conf (dmix)

[Clemens Ladisch via Alsa-user]

Tobx wrote:
> I want to set the sample rate to 44.1 kHz in dmix in order to prevent 
> upsampling.

Override defaults.pcm.dmix.rate.

> Is the Master control a hardware control?

That depends on whether your hardware has it.


Regards,
Clemens

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Re: [Alsa-user] [asoundrc] "lossless" format conversion without any rate conversion?

[Clemens Ladisch via Alsa-user]

Vincent Yu wrote:
> Will any sample rate conversions occur with this .asoundrc config?
>
> pcm_slave.force_24_bit_no_rate_convert {
> pcm "hw:0,0"
> format S24_3LE # Or other 24-bit format
> rate "unchanged" # Necessary?
> }
>
> pcm.my_new_default {
> type plug
> slave force_24_bit_no_rate_convert
> }

http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html#pcm_plugins_plug

This plugin will not do resampling.

> Specifically, does sample rate conversion still occur anywhere in the
> signal chain even though I provided the "unchanged" string for rate?

The hardware, another plugin layered on top of this, or some application
might still do resampling.  :)

> I would like to create a configuration to force digital audio output
> from my PC (Arch Linux - ALSA only, no Pulseaudio) to remain at or be
> converted to 24-bit format without any sample rate conversion.

Then try the linear plugin, which changes only the sample format:

pcm.my_new_default {
type linear
slave {
pcm "hw:0,0"
format S24_3LE
}
}

(The "plug" plugin does nothing but dynamically inserting other plugins,
such as "linear"/"route"/"rate", when required.)

> I understand that the word, "lossless," in the title may be a bit of
> a misnomer, since quantization noise from quantization errors during
> bit depth conversion will raise the noise floor of the signal by
> a small (and probably inaudible) amount

Converting between integer sample formats does not change the bits; you
get quantization errors only with volume changes.

> looking at the code and trying to grep directory trees recursively for
> the string, "unchanged", to see if I can get any leads, I got lost in
> the behemoth of a code base

The "unchanged" string is parsed in pcm.c; in pcm_plug.c, search for
"== -2".


Regards,
Clemens

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Re: [Alsa-user] can play back but can't capture

[Clemens Ladisch via Alsa-user]

Kent Tong wrote:
> Please find the aadebug.log below.

I did not find it.

> arecord captures basically silence

Probably wrong mixer settings.


Regards,
Clemens

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Re: [Alsa-user] can play back but can't capture

[Clemens Ladisch via Alsa-user]

Ralf Mardorf wrote:
> On Thu, 23 Nov 2017 08:59:44 +0100, Clemens Ladisch via Alsa-user wrote:
>>> Please find the aadebug.log below.
>>
>> I did not find it.
>
> It isn't below, it's attached.

Well, I did look below where attachments would have been shown, but didn't
see it.  Until now.  Sorry about that.

Anyway, that file does not contain the mixer settings.
Please show the output of "amixer scontents".


Regards,
Clemens

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Re: [Alsa-user] can play back but can't capture

[Clemens Ladisch via Alsa-user]

Kent Tong wrote:
> Simple mixer control 'Capture',0
>   Front Left: Capture 63 [100%] [30.00dB] [on]
>   Front Right: Capture 63 [100%] [30.00dB] [on]
> Simple mixer control 'Input Source',0
>   Items: 'Front Mic' 'Rear Mic' 'Line'
>   Item0: 'Front Mic'
> Simple mixer control 'Front Mic Boost',0
>   Front Left: 2 [67%] [20.00dB]
>   Front Right: 2 [67%] [20.00dB]

This looks OK.  Did you try to record from the front mic?

> Simple mixer control 'Capture',1
>   Front Left: Capture 0 [0%] [-17.25dB] [off]
>   Front Right: Capture 0 [0%] [-17.25dB] [off]
> Simple mixer control 'Input Source',1
>   Items: 'Front Mic' 'Rear Mic' 'Line'
>   Item0: 'Rear Mic'
> Simple mixer control 'Rear Mic Boost',0
>   Front Left: 0 [0%] [0.00dB]
>   Front Right: 0 [0%] [0.00dB]

You have a second capture device (see "arecord -l"), which is currently
configured to record silence.

Which device did you try to use?  Can you try both?

> Simple mixer control 'Line Boost',0
>   Front Left: 0 [0%] [0.00dB]
>   Front Right: 0 [0%] [0.00dB]

Does the line input work?


Regards,
Clemens

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Re: [Alsa-user] can play back but can't capture

[Clemens Ladisch via Alsa-user]

Kent Tong wrote:
> Below is the output of "arecord -l". I have no idea what that Alt device is.
>
> card 0: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog]

hw:0,0 (or better plughw:0,2 for automatic sample format conversion)

> card 0: PCH [HDA Intel PCH], device 2: Generic Alt Analog [Generic Alt Analog]

hw:0,2 (plughw:0,2)

>> Which device did you try to use?  Can you try both?
>
> I only use the front mic.

The second device can also be configured to record from the front mic.

Please try the rear mic, too.

>> Does the line input work?
>
> how to test that?

Connect anything to the line input, and set it as source.


Regards,
Clemens

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Re: [Alsa-user] ASoC: bytcr_rt5651 - sound on headphones only, no speaker no microphone

[Daimon Wang via Alsa-user]

I've ever work with an rt5645 on a Z8350, never get sound output.Since you have 
sound on headphone, maybe u can try with different kernel version? 4.10 ~ 4.14.
And check if this thread helps, 
https://bugzilla.kernel.org/show_bug.cgi?id=95681Here is a git from plbossart 
(who fix above bug), which is related to his quirks on rt serial sound cards. 
But he said it should be merged into 4.13. 
https://github.com/plbossart/sound.git
Goold luck,Daimon 

On Thursday, November 30, 2017 9:04 PM, Luke Ross  
wrote:
 

 Hi,

I've a Linx 820 tablet (cherrytrail z8350) which seems to have an
RT5651 codec. Using driver bytcr_rt5651 which is loaded automatically,
sound works fine on headphones, but the speaker output appears to play
(in that the the seconds count up and pavucontrol shows "sound" on the
little graphs) but the output is silent (I've checked it's not muted).
Also, PulseAudio denies there is any microphone on the machine and for
headphones the jack detection doesn't work so the output must be
switched manually to headphones.

Is this likely to be because it needs some of the driver's quirks
settings for the hardware? Are there particular ones it's likely to be
that I should try first? The tablet DMI-identifies it's made by Insyde,
and the code for the the RT5640 suggests they needed several quirks on
that (similar, bay trail) hardware. 

Many thanks for any pointers/suggestions to get me started!

Luke

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Re: [Alsa-user] ASoC: bytcr_rt5651 - sound on headphones only, no speaker no microphone

[Daimon Wang via Alsa-user]

Any luck? @Luke 

On Friday, December 1, 2017 2:48 PM, Daimon Wang via Alsa-user 
 wrote:
 

 I've ever work with an rt5645 on a Z8350, never get sound output.Since you 
have sound on headphone, maybe u can try with different kernel version? 4.10 ~ 
4.14.
And check if this thread helps, 
https://bugzilla.kernel.org/show_bug.cgi?id=95681Here is a git from plbossart 
(who fix above bug), which is related to his quirks on rt serial sound cards. 
But he said it should be merged into 4.13. 
https://github.com/plbossart/sound.git
Goold luck,Daimon 

On Thursday, November 30, 2017 9:04 PM, Luke Ross  
wrote:
 

 Hi,

I've a Linx 820 tablet (cherrytrail z8350) which seems to have an
RT5651 codec. Using driver bytcr_rt5651 which is loaded automatically,
sound works fine on headphones, but the speaker output appears to play
(in that the the seconds count up and pavucontrol shows "sound" on the
little graphs) but the output is silent (I've checked it's not muted).
Also, PulseAudio denies there is any microphone on the machine and for
headphones the jack detection doesn't work so the output must be
switched manually to headphones.

Is this likely to be because it needs some of the driver's quirks
settings for the hardware? Are there particular ones it's likely to be
that I should try first? The tablet DMI-identifies it's made by Insyde,
and the code for the the RT5640 suggests they needed several quirks on
that (similar, bay trail) hardware. 

Many thanks for any pointers/suggestions to get me started!

Luke

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Re: [Alsa-user] ASoC: bytcr_rt5651 - sound on headphones only, no speaker no microphone

[Daimon Wang via Alsa-user]

Maybe, u can try to contact plbossart (who fix 95681). I saw him give fixes for 
several similar platform, with the quirk related to dmidocede-info.I planned to 
write to him, but just can't find any contact info. 

On Saturday, December 9, 2017 6:01 PM, Luke Ross  wrote:
 

 Hi,

Thanks for your reply, and sorry for my delay.

On Fri, 2017-12-01 at 06:47 +, Daimon Wang wrote:
> I've ever work with an rt5645 on a Z8350, never get sound output.
> Since you have sound on headphone, maybe u can try with different
> kernel version? 4.10 ~ 4.14.

I had a play with a few versions but in that series didn't have any
success. I noted that in the 4.15-rc series the rt5651 driver has been
overhauled, so built and tried that and found it was worse - no sound
at all! It turns out my machine needs one of the flags, MCLK_EN, to not
be set (I've read something which suggests that MCLK_EN is Baytrail-
specific and not something to be done on Cherrytrail). Once that has
been changed I do at least get headphones but still no speakers. I
tried the DMIC, IN1 and IN2 maps the new driver provides but no
microphone on any of them - although I don't know whether this is
because one of the many many mixer controls are set wrongly. I may have
another play with different combinations of recording mixer controls
and see what happens.

Either way, I'll submit a patch to add this machine to the quirk map so
that MCLK_EN is not set by default on this hardware - that way I at
least get to keep the headphone audio.

Many thanks,

Luke

> 

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Re: [Alsa-user] MIDI subscription

[Clemens Ladisch via Alsa-user]

Paul D. DeRocco wrote:
> Under "Subscription", it says "subscription is a connection between two
> sequencer ports." The snd_seq_subscribe_port() function takes a
> snd_seq_port_subscribe_t which specifies a sender and a receiver. But the
> explanation always seems to refer to subscribing "to" a port, rather than
> subscribing "between" two ports.

In most cases, the receiver is your own port.  But in the general case,
all subscriptions are between two ports.

> [...] And then it says "There is another subscription type for opposite
> direction". But if subscription is a connection between a source and
> a destination, what are the two types of subscription?

There are not really two types.  But if you want connections in both
directions, you need to make two subscriptions, with opposite "from"/"to"
ports.

> Later, under "Permissions", it speaks of READ and WRITE permissions being
> needed to send events to or receive events from other ports. Then, it
> speaks of SUBS_READ and SUBS_WRITE permissions, which are needed for
> subscription, implying that subscription isn't the same thing as merely
> passing events from one port to another.

All events have a destination port, which can be a specific port, or
"all subscribers".  A program like aplaymidi sends events to a specific
destination and does not allow subscriptions.

> It then says that these SUBS flags aren't necessary "if the client
> subscribes itself to the specified port". Since a client is just an
> object, not a program or a piece of code, how does a client subscribe
> itself to something, or do anything at all?

All accesses to the ALSA sequencer are done through a client object, so
for practical purposes, a client indeed is a program or a piece of code.

The permission bits allow other clients to do something with the port;
a client is always allowed to do everything with its own ports.

So if you want to subscribe from or to your own port, the other port
needs to allow this; if you subscribe between two other ports, both need
to allow this.

(Please note that events are always sent by the source port, so it is not
possible to read events from another client's port without a subscription,
so the READ bit is pointless without SUBS_READ.)


Regards,
Clemens

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Re: [Alsa-user] Problems opening devices

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> After this I try snd_pcm_open on the IDs, most of which I get -EBUSY.

Did you actually close the device from the previous try?

Check in /proc/asound/cardX/pcm0p/sub0/status if the device is opened.


Regards,
Clemens

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Re: [Alsa-user] Strange i/o problem

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> It works nicely if I either:
> 3. Pipe capture -> playback with a larger buffer size, such as 64.

It's possible that the hardware does not actually handles size 32 correctly.
> The rendering thread is (pseudo code):
>
> while (true) {
> if(capture_active) {
>snd_pcm_wait(capture_handle, timeout);
>read_pcm_data_into_buffer(capture_handle, input_buffer);
> }
> do_callback(input_buffer, output_buffer);
> if (playback_active) {
> snd_pcm_wait(playback_handle, timeout);
> write_pcm_data_from_buffer(playback_handle, output_buffer);
> }
> }

Are you filling the output before the loop?


Regards,
Clemens

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Re: [Alsa-user] Dmix problem

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
>> Is there any other plugin doing the same thing as dmix... but working ?

Yes, dmix with a larger buffer size (i.e., more periods).

> I'd need a mixing plugin that does not do sample rate conversion, i.e. each
> client connecting to it should be forced to use the mix plugs sample rate.

The dmix plugin does not do sample rate conversion, i.e., each client
connecting to it is forced to use the dmix rate.  (If you wanted to, it would
be possible to put a "plug" or "rate" plugin on top of it.)


Regards,
Clemens

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Re: [Alsa-user] Iec958 plugin usage

[Clemens Ladisch via Alsa-user]

maruthi srinivas wrote:
> Iam trying to convert pcm (from wav file) to AES and vice versa.

What exactly do you mean with "AES"?

> I found a alsa plugin iec958. Will that plugin does do what I want
> to achieve ?

The iec958 creates an IEC958 bit stream, which is required for S/PDIF
on some very old C-Media cards.

This plugin is useless if you have any other hardware.


Regards,
Clemens

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Re: [Alsa-user] Dmix problem

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> is there some ALSA plugin that can coalesce buffering ? Meaning that
> the plugin can take f.i. larger period_size than what the dmix device
> is working with ?

What problem would that solve?


Regards,
Clemens

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Re: [Alsa-user] Iec958 plugin usage

[Clemens Ladisch via Alsa-user]

(please don't top-post)

maruthi srinivas wrote:
> By AES, i mean AES standard as in:
> https://en.m.wikipedia.org/wiki/AES3

This is a hardware standard; it does not define the format of samples in
memory.

> This will add 4bit before and 4 bit after pcm sample.it seems iec958 plugin
> adds those 8bits to encode pcm to iec958/AES standard and does reverse to
> decode to pcm. Is my understanding correct ?

The on-the-wire format has the premable, which does not correspond to any
correctly-encoded bit value, so you cannot talk about bits here.

> I want to use plugin to convert either way and write to a file.

What for?  There is no standard for AES data in a file.


Regards,
Clemens

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Re: [Alsa-user] Iec958 plugin usage

[Clemens Ladisch via Alsa-user]

maruthi srinivas wrote:
> On Monday, January 22, 2018, Clemens Ladisch 
> wrote:
>> maruthi srinivas wrote:
>>> I want to use plugin to convert either way and write to a file.
>>
>> What for?  There is no standard for AES data in a file.
>>
> Aplay help lists IEC958_SUBFRAME_LE format.

That is a format used by certain hardware, but it cannot be stored in
.wav files.  (This is why the plugin is needed.)


Regards,
Clemens

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Re: [Alsa-user] Best course of action with Audient iD14

[Clemens Ladisch via Alsa-user]

Charles Mulder wrote:
> I would like to ask your help to establish what the best course of action
> would be.

Ask the PulseAudio developers how to configure it to detect the device 
correctly:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Community/


Regards,
Clemens


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Re: [Alsa-user] Best course of action with Audient iD14

[Clemens Ladisch via Alsa-user]

Charles Mulder wrote:
> I am trying to set up a dedicated audio workstation for composing and
> would like to keep it as minimal as possible. I was thinking of using
> ALSA and Jack without Pulseaudio.
>
> In your opinion, is that a mistake?

Probably not.  But the "Analogue Surround 4.0" classification is done by
PulseAudio, so it is not clear what problem you actually have.


Regards,
Clemens

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Re: [Alsa-user] alsactl sometimes fails to restore mixer settings at boot

[Clemens Ladisch via Alsa-user]

Nikos Chantziaras wrote:
> alsactl sometimes fails to restore the mixer settings at system boot. Failure 
> rate is about 50/50.
>
> systemd-udevd[151]: Process '/usr/sbin/alsactl restore 1' failed with exit 
> code 99.
> systemd-udevd[149]: Process '/usr/sbin/alsactl restore 0' failed with exit 
> code 99.

This sounds as if the card numbers might be mixed up.  How do you control
which card gets which index?


Regards,
Clemens

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Re: [Alsa-user] dmix with non-hw slave ?

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> Is it possible on the ALSA side to setup dmix to use the "jack" type
> plugin as backend ?

No; dmix requires a feature that is supported only by the kernel driver.

> Or will the jack plugin do the same job as dmix, i.e. mix together
> applications using the device ?

Every application has its own instance of the Jack plugin.  Doesn't Jack
mix the streams together?


Regards,
Clemens

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Re: [Alsa-user] splitting a 4 channel usb sound card into master and headphone

[Clemens Ladisch via Alsa-user]

Samuel Nicholas wrote:
> I have a Pioneer DDJ-WeGO, 4 channel USB device that outputs stereo
> master and stereo headphone.
> I've been scouring the internet trying to find out how I can possibly
> split up the ports as it presents as a surround 4.0 device.

Try something like this:
https://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg12236.html


Regards,
Clemens

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Re: [Alsa-user] splitting a 4 channel usb sound card into master and headphone

[Clemens Ladisch via Alsa-user]

Samuel Nicholas wrote:
> src/conf/cards/USB-Audio.conf
>
> on lines 22 starts the description of configuring the usb device to be
> two stereo devices not a surround40

No, that is for devices that implement two stereo devices in hardware
and should be seen as a single 4.0 device by software.


Regards,
Clemens

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Re: [Alsa-user] alsactl sometimes fails to restore mixer settings at boot

[Clemens Ladisch via Alsa-user]

Nikos Chantziaras wrote:
> How do I assign numbers?

With a line

  options snd slots=snd-virtuoso,snd-usb-audio

in some .conf file in /etc/modprobe.d/.


Regards,
Clemens

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Re: [Alsa-user] USB Audio and selecting optical out

[Clemens Ladisch via Alsa-user]

Marc Haber wrote:
> Is it possible that the USB device is only able to play back at 48 kHz?

Yes, it's possible.  Check /proc/asound/cardX/stream0.

> On Sun, Apr 08, 2018 at 05:20:13PM +0100, James wrote:
>> You might also need to mess with AES0 settings. Google for that.

The USB audio driver does not have these settings.


Regards,
Clemens

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Re: [Alsa-user] USB Audio and selecting optical out

[Clemens Ladisch via Alsa-user]

Marc Haber wrote:
> On Sun, Apr 22, 2018 at 10:45:11PM +0200, Clemens Ladisch via Alsa-user wrote:
>> Marc Haber wrote:
>>> Is it possible that the USB device is only able to play back at 48 kHz?
>>
>> Yes, it's possible.  Check /proc/asound/cardX/stream0.
>
> Playback:
> Rates: 44100, 48000
> Capture:
> Rates: 44100, 48000
>
> Looks to me like it at least claims to be able to play back at 44.1 kHz.
> Any other reason why the DAT deck won't sync if speakertest runs with
> rate 44100?

Check if it's actually using 44.1 kHz when playing.

> Is this likely a sloppy implementation of the Device or a bug in the driver?

S/PDIF has lots of metadata bits, but the USB audio 1.x specification does
not have any mechanism to change them.  These bits must be set automatically
by the device.  (Usually, receivers don't really care about them.)

The CM106 datasheet mentions a register to control the S/PDIF output, but
its contents are undocumented.  It's possible that this works only with the
Windows driver.


Regards,
Clemens

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Re: [Alsa-user] USB Audio and selecting optical out

[Clemens Ladisch via Alsa-user]

Marc Haber wrote:
> On Tue, Apr 24, 2018 at 02:43:01PM +0200, Clemens Ladisch via Alsa-user wrote:
>> Marc Haber wrote:
>>> Looks to me like it at least claims to be able to play back at 44.1 kHz.
>>> Any other reason why the DAT deck won't sync if speakertest runs with
>>> rate 44100?
>>
>> Check if it's actually using 44.1 kHz when playing.
>
> How would I do that?

Check /proc/asound/cardX/stream0.

> I am open to suggestions for devices that will do what I want and work
> on Linux.

I do not know what your DAT actually requires.

But as far as I am aware, there is no USB solution that allows as much
control as your C-Media PCI card.


Regards,
Clemens

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Re: [Alsa-user] USB Audio and selecting optical out

[Clemens Ladisch via Alsa-user]

Marc Haber wrote:
> On Tue, Apr 24, 2018 at 02:54:44PM +0200, Clemens Ladisch via Alsa-user wrote:
>> Marc Haber wrote:
>>> On Tue, Apr 24, 2018 at 02:43:01PM +0200, Clemens Ladisch via Alsa-user 
>>> wrote:
>>>> Check if it's actually using 44.1 kHz when playing.
>>>
>>> How would I do that?
>>
>> Check /proc/asound/cardX/stream0.
>
> Now we're turning around in circles. Are you trying to say that the
> contents of /proc/asound/cardX/stream0 will not only show the card's
> capabilities but also what is currently going on while there is
> something playing?

Yes, the "Status: Stop" part shows more information when not stopped.


Regards,
Clemens

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Re: [Alsa-user] how to purge old ALSA controls without reboot

[Clemens Ladisch via Alsa-user]

frede...@ofb.net wrote:
> I'm trying to figure out how to delete unused volume controls which
> had appeared in previous versions of my .asoundrc.
>
> Can this be done without rebooting my computer?

Unload the sound driver (rmmod), then remove the entries from asound.state,
then re-load the driver (modprobe).


Regards,
Clemens

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[Alsa-user] Terratec EWX 24/96 problems, cracking and distorbing sound

[Christian First via Alsa-user]

Hello, i am running Ubuntu 16.04.4 64bit with Lubuntu core. I am using a 
Terratec EWX 24/96 pci soundcard with ice1712 chip. The sound is always 
cracking and distorbing. Once i managed it with the order "alsa reload". But 
after i restarted my computer the sound is still cracking and distorbing. The 
alsa reload order doesn't work anymore. The guys from ubuntuusers.de forum 
couldn't help me. --
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[Alsa-user] Get triggered when the device opens

[Frank Thomas via Alsa-user]

Hello,

I'd like to apologise beforehand if I have not read enough
documentation and/or source code to be posting here. Please do not
hesitate to tell me to RTFM ☺

My application needs to get triggered when a mechanism any application
is opening or closing a certain list of ALSA devices. It seems that
ALSA hooks would be a way to do it. AFAIK I can only use a hook to
trigger a function a in a pcm control plugins with certain arguments. I
would need implement a control plugin that notifies my application for
me and use a hook to trigger the trigger ☺

Is there another way, e.g. with the ALSA API, to get triggered on such
events?

Thanks in advance.


Regards,
Thomas Frank
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Re: [Alsa-user] Get triggered when the device opens

[Frank Thomas via Alsa-user]

I would like to further clarify some things.

I actually need get triggered when the ALSA device started to run,
meaning the device is producing or outputting samples.

FYI: The actual sound controller is connected via an I²S interface and
synchronizes on its the word select signal. After the synchronisation,
additional configuration must to occur before you can get valid samples
from the ALSA interface. The configuration is lost upon synchronisation
loss (ALSA device is not running anymore).
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Re: [Alsa-user] Get triggered when the device opens

[Clemens Ladisch via Alsa-user]

Frank Thomas via Alsa-user wrote:
> The actual sound controller is connected via an I²S interface and
> synchronizes on its the word select signal. After the synchronisation,
> additional configuration must to occur before you can get valid samples

Then why don't you do the configuration in the driver?


Regards,
Clemens

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Re: [Alsa-user] Get triggered when the device opens

[Frank Thomas via Alsa-user]

On Mon, 2018-06-04 at 15:23 +0200, Clemens Ladisch via Alsa-user wrote:
> Frank Thomas via Alsa-user wrote:
> > The actual sound controller is connected via an I²S interface and
> > synchronizes on its the word select signal. After the
> > synchronisation,
> > additional configuration must to occur before you can get valid
> > samples
> 
> Then why don't you do the configuration in the driver?

The configuration is done with an userspace stack/framework which is
provided by the manufacturer of the controller. This stack is important
for diagnostics purposes during configuration and runtime.

Regards,
Thomas
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Re: [Alsa-user] Get triggered when the device opens

[Clemens Ladisch via Alsa-user]

Frank Thomas via Alsa-user wrote:
> On Mon, 2018-06-04 at 15:23 +0200, Clemens Ladisch via Alsa-user wrote:
>> Frank Thomas via Alsa-user wrote:
>>> The actual sound controller is connected via an I²S interface and
>>> synchronizes on its the word select signal. After the synchronisation,
>>> additional configuration must to occur before you can get valid samples
>>
>> Then why don't you do the configuration in the driver?
>
> The configuration is done with an userspace stack/framework which is
> provided by the manufacturer of the controller.

And how does that manufacturer tell you to use it?

Anyway, it might be possible to write an external filter plugin:
https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html

Does the kernel driver start the I²S clocks in hw_params() or in trigger()?


Regards,
Clemens

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Re: [Alsa-user] Xonar D2X optical out problem

[Clemens Ladisch via Alsa-user]

David Woodfall wrote:
> I've have had a Xonar D2X for some time now and normally use coax
> spdif output to an SMSL Sanskrit DAC.
> [...]
> I couldn't get any signal via the optical out at all and there isn't
> any light that you normally see shining through the output socket.

The coax and optical output should work in the same way (and the
software has no way of accessing them separately).

If coax works and optical does not, then this sounds as if some part
of your hardware is broken.


Regards,
Clemens

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Re: [Alsa-user] Get triggered when the device opens

[Frank Thomas via Alsa-user]

> And how does that manufacturer tell you to use it?

Get the clock started and configure. The loss of the WS signal is not intended 
and is a failure. A xrun results in configuration loss.


> Anyway, it might be possible to write an external filter plugin:
> https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html

I am doing that right now. Right now. It compiles but does not start:

$ cat ~/.asoundrc 
pcm_hw_internal {
   type hw
   card 0
   device 0
}
pcm.myplugin {
   type myplugin
   slave {
  pcm pcm_hw_internal
   }
}

$ arecord -D myplugin /dev/zero
ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol 
_snd_pcm_myplugin_open
ALSA lib dlmisc.c:263:(snd1_dlobj_cache_get) symbol _snd_pcm_myplugin_open is 
not defined inside 
/usr/lib/x86_64-linux-gnu/alsa-lib/libasound_module_pcm_myplugin.so
arecord: main:722: audio open error: No such device or address

This looks like a compile problem, but I don't what exactly. It is compiled 
like this:
$ gcc -I/usr/include/alsa -Wall -g -lasound -fPIC -DPIC -shared -o 
build/libasound_module_pcm_myplugin.so alsa-plugin.c

FYI: I am just testing this on my local machine (ubuntu 16.04) with pulseaudio 
killed (and autospawn=no).

> Does the kernel driver start the I²S clocks in hw_params() or in trigger()?

I cannot tell because I am not versed enough in device/ALSA/SOC drivers but I 
made a step by step device initialization via alsa-lib and the clock really 
starts only the device is written or read. The SOC manufacturer supplies the 
ALSA device drivers that utilizes the I2S/PCM interface to talk to the audio 
controller which I need to configure.

Regards,
Thomas Frank

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Re: [Alsa-user] No sound cards seen with kernel 4.17.2 and 4.17.3

[Clemens Ladisch via Alsa-user]

Pete wrote:
> Linux phoenix 4.16.16:
>  1 [Generic]: HDA-Intel - HD-Audio Generic

"Generic" shows that the kernel does not know the HDA controller
in your chipset.

> Linux phoenix 4.17.3:
> --- no soundcards ---

There should be an error message in the system log.


Regards,
Clemens

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Re: [Alsa-user] Phonic Helix Board 12 Universal: Recording is silent

[Clemens Ladisch via Alsa-user]

Andreas Böhler wrote:
> all recordings are completely silent

Please show the output of "amixer -c2".


Regards,
Clemens

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Re: [Alsa-user] No sound cards seen with kernel 4.17.2 and 4.17.3

[Clemens Ladisch via Alsa-user]

Peter Chant wrote:
> Clemens wrote:
>> Pete wrote:
>>> Linux phoenix 4.17.3:
>>> --- no soundcards ---
>
>> There should be an error message in the system log.
>
> Jul  2 19:44:10 phoenix kernel: [   12.072075] snd_hda_intel :09:00.1: 
> Handle vga_switcheroo audio client
> Jul  2 19:44:10 phoenix kernel: [   12.072299] snd_hda_intel :09:00.1: 
> SME is active, device will require DMA bounce buffers
> Jul  2 19:44:10 phoenix kernel: [   12.072525] snd_hda_intel :09:00.1: 
> SME is active, device will require DMA bounce buffers
> Jul  2 19:44:10 phoenix kernel: [   12.072827] snd_hda_intel :0b:00.3: 
> SME is active, device will require DMA bounce buffers
> Jul  2 19:44:10 phoenix kernel: [   12.073056] snd_hda_intel :0b:00.3: 
> SME is active, device will require DMA bounce buffers

No errors here.

You might try disabling SME (secure memory encryption) in the BIOS,
but in theory, this should not make any difference.


Regards,
Clemens

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Re: [Alsa-user] Phonic Helix Board 12 Universal: Recording is silent

[Clemens Ladisch via Alsa-user]

Andreas Böhler wrote:
> On 02/07/18 23:02, Clemens Ladisch via Alsa-user wrote:
>> Andreas Böhler wrote:
>>> all recordings are completely silent
>>
>> Please show the output of "amixer -c2".
>
> There is no output at all. "alsamixer -c2" reports "This sound device does 
> not have any controls."

Then there isn't much that could be done in software.

It's possible that this devices requires a vendor-specific Windows
driver that knows some secret command to make it behave.


Regards,
Clemens

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[Alsa-user] Help with Audient iD14

[Charles Mulder via Alsa-user]

Hi everyone. 

I bought an Audient iD14 audio interface, which is supposed to be USB Audio 
Class 2 compliant, but thus far I have been unable to get it to work as 
expected.

I am unable to select the device using alsamixer. I did manage to hear stereo 
playback, but front right is extremely soft compared to front left. 

I ran also-info which uploaded debug info to:

http://www.alsa-project.org/db/?f=47fc359b76937a58100abbca82c3b77750599c54

Any help or guidance will be appreciated.

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Re: [Alsa-user] Help with Audient iD14

[Clemens Ladisch via Alsa-user]

Charles Mulder via Alsa-user wrote:
> I am unable to select the device using alsamixer.

What exactly happens when you try it?


Regards,
Clemens

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Re: [Alsa-user] Get triggered when the device opens

[Frank Thomas via Alsa-user]

Hello,

just to finish this thread, a colleague of mine put me on the right track: 
timers. If you look at the alsa-lib timer example 
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2timer_8c-example.html , 
you see the timer interface being used to register on master timer ticks. The 
read loop simply waits on poll.

Actually, timer work on a variety of events, see 
http://www.alsa-project.org/alsa-doc/alsa-lib/group___timer.html ,  enum  
snd_timer_event_t . To register on specific events, you create a filter mask of 
your events ( 1 << event1 || 1 << event2 || ... ) and set it to your timer 
parameters with snd_timer_params_set_filter . In read loop of the timer 
example, you just need to use snd_timer_tread_t instead of snd_timer_read_t .

In my case, I registered on the timer events SND_TIMER_EVENT_MSTART and 
SND_TIMER_EVENT_MSTOP and it works as expected.

Regards,
Thomas

From: Frank Thomas
Sent: 04 June 2018 08:53:57
To: alsa-user@lists.sourceforge.net
Subject: Get triggered when the device opens

Hello,

I'd like to apologise beforehand if I have not read enough
documentation and/or source code to be posting here. Please do not
hesitate to tell me to RTFM ☺

My application needs to get triggered when a mechanism any application
is opening or closing a certain list of ALSA devices. It seems that
ALSA hooks would be a way to do it. AFAIK I can only use a hook to
trigger a function a in a pcm control plugins with certain arguments. I
would need implement a control plugin that notifies my application for
me and use a hook to trigger the trigger ☺

Is there another way, e.g. with the ALSA API, to get triggered on such
events?

Thanks in advance.


Regards,
Thomas Frank
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