Re: [Alsa-user] configuring asoundrc

[Clemens Ladisch via Alsa-user]

James Cloos wrote:
> CL>   pcm.!default {
> CL> type asym
> CL> playback.pcm "plug:hdmi:0,0"
> CL> capture.pcm "plughw:1"
> CL>   }
>
> ALSA lib 
> /var/tmp/portage/media-libs/alsa-lib-1.2.4/work/alsa-lib-1.2.4/src/conf.c:5084:(parse_args)
>  Unknown parameter 1

Oops, the hdmi device does not take a second parameter.
Use:
playback.pcm "plug:hdmi:0"


Regards,
Clemens


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Re: [Alsa-user] configuring asoundrc

[Clemens Ladisch via Alsa-user]

James Cloos wrote:
> pcm.!default {
> type hw
> card Generic
> device 3
> }
>
> ctl.!default {
> type hw
> card Generic
> }

This disables all automatic sample rate/format conversions.

Replace it with:

  pcm.!default {
type asym
playback.pcm "plug:hdmi:0,0"
capture.pcm "plughw:1"
  }


Regards,
Clemens


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Re: [Alsa-user] ALSA MIDI on Raspberry PI: no events from input device (USB keyboard)

[Clemens Ladisch via Alsa-user]

Fernando Carello wrote:
> So, I've seen that a working MIDI keyboard "creates" this device:
>
> /dev/snd/-midiC0D0
>
> while my problematic MIDI keyboard has instead:
>
> /dev/snd/-midiC1D0

"C1" means "card 1"; this is because card 0 is already occupied by another 
device.

> The keyboard seem to send raw midi messages in a stream, but Alsa does
> not translate them in Sequencer events.

Strange. This should happen automatically, especially if the sequencer
device (port 20) is there.

Does the count in /proc/asound/card1/midi0 increase while aseqdump
is running?


Regards,
Clemens


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Re: [Alsa-user] [alsa-user] Error: Unable to install hw params

[Clemens Ladisch via Alsa-user]

Vijay Rakesh Munganda wrote:
> I had tried to record a sample audio using arecord, “arecord hw:0,6 -d 10 
> test.wav” but I got an error as unable to install hw params.

Use plughw:0,x instead of hw:0,x to get automatic sample format conversion.

Or tell arecord to use a sample rate/format that your hardware supports.


Regards,
Clemens


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Re: [Alsa-user] ALSA MIDI on Raspberry PI: no events from input device (USB keyboard)

[Clemens Ladisch via Alsa-user]

Fernando Carello wrote:
> Problem: when I connect my MIDI USB keyboard (GarageKey), it is
> correctly detected, but I don't receive MIDI messages from it.
>
> No events with asqedump:  :-(

Try with "amidi --dump --port hw:1".

Do other USB MIDI devices (or USB devices) work?

> In fact, ALSA "sees" my USB keyboard as an audio card, which sounds odd

This is normal.


Regards,
Clemens


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Re: [Alsa-user] Xonar D2X: Playback only when other devices generate interrupts

[Clemens Ladisch via Alsa-user]

Matthias Leofeine wrote:
> Playback is interrupted (only short pieces of sound are played/repeated)
> or I've got no sound at all; applications log message: "underrun
> occurred".

This sounds like the ASM1083 problem, but that affected only PCI cards.
The PCI/PCIe bridge on the D2X card is not known to have such problems.

Please show the output of "lspci -tvv".


Regards,
Clemens


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Re: [Alsa-user] aplay -l doesn't see my loopback sound card using snd_aloop

[Clemens Ladisch via Alsa-user]

Jeff Tchang wrote:
>    WARNING   ***
> Warning! aplay uses ALSA emulation instead of the native OSS API

This message shows that your system is set up to use not the ALSA
kernel drivers but the OSS drivers.

> I have an embedded system with a custom kernel.

Are you paying someone for support?  Or did you build it yourself?


Regards,
Clemens


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Re: [Alsa-user] Parallel playback on multiple USB cards

[Clemens Ladisch via Alsa-user]

Alexandre Bouillot wrote:
> ... Raspberry Pi
>
> I'm able to address each of the 6 cards independently correctly.
> However, when I'm trying to have a second channel - not to mention the third 
> - working in parallel with the first, I face problems.

>From what I've heard, the Pi's USB controller has limited bandwidth, and its
driver is not able to distribute all of it among multiple streams.


Regards,
Clemens


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Re: [Alsa-user] New user with an ICUSBAUDIO7D

[Clemens Ladisch via Alsa-user]

Alan Corey wrote:
> arecord -f S16_LE -d 10 -r 16000 --channels=1 --device="hw:2,0" 
> /tmp/test-mic.wav
> Recording WAVE '/tmp/test-mic.wav' : Signed 16 bit Little Endian, Rate 16000 
> Hz, Mono
> arecord: set_params:1345: Channels count non available

I suspect recording always uses channels.
Please show the contents of /proc/asound/card2/stream0.


Regards,
Clemens


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Re: [Alsa-user] documentation for error codes?

[Clemens Ladisch via Alsa-user]

D. R. Evans wrote:
> Many functions say about the return value:
>   0 on success otherwise a negative error code
>
> Where is the documentation for the various possible "negative error code"s?

 has a list
of error codes defined by ALSA itself.  But for all other errors, ALSA just
uses the (negative) POSIX error code.

> I am seeing a -32 under a specific set of circumstances

32 = EPIPE

> but don't know what that returned value means

PCM stream stopped due to overrun or underrun.


Regards,
Clemens


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Re: [Alsa-user] PCM plugin last

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
>> It might be possible to run dmix on a virtual loopback sound card and route 
>> that one to the LADSPA plugin
>
> Can this loopback routing be done entirely within a .asoundrc configuration?

No; some program must be running to record from the loopback card and play to 
the actual device.


Regards,
Clemens


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Re: [Alsa-user] PCM plugin last

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> I’d like to know if there is any way to get the following sound chain through 
> ALSA:
>
> Mediaplayer -> dmix -> LADSPA -> plughw:0,0

The dmix plugin plays shared-memory tricks with the ring buffer and therefore
requires to run on top of a hw device.

It might be possible to run dmix on a virtual loopback sound card and route
that one to the LADSPA plugin, but I guess using PulseAudio would be a better
idea.


Regards,
Clemens


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Re: [Alsa-user] Android Pixel C in MIDI mode not creating a alsa seq port

[Clemens Ladisch via Alsa-user]

Stefan Sauer wrote:
> idProduct=5209 is the Pixel C as a midi device, 520a is when it connects
> as MTP.

Your lsusb output showed 520a. Was it supposed to be in MIDI mode at
that time?

Please show the output of "lsusb -d 18d1:5209 -v".


Regards,
Clemens


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Re: [Alsa-user] Android Pixel C in MIDI mode not creating a alsa seq port

[Clemens Ladisch via Alsa-user]

Stefan Sauer wrote:
> If I use my pixel c tablet, the alsa device is recognized (it shows
> up in /proc/asound/ and also creates a midi0 entry)

Does it show up in the output of "amidi -l"?

Is the ID 5209 or 520a?  Your logs show both.


Regards,
Clemens


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Re: [Alsa-user] Rodecaster Pro USB

[Clemens Ladisch via Alsa-user]

Nils wrote:
> I am trying to get a Rodecaster Pro working, and see it in lsusb, but
> can't get it to work with Alsa.
>
> [ 3490.377201] usb 1-7: parse_audio_format_rates_v2v3(): unable to retrieve 
> number of sample rates (clock 1)

This device appears not to be completely class compliant.

Does it work without a separately installed driver in Windows and OS X?


Regards,
Clemens


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Re: [Alsa-user] RME Exspansion

[Clemens Ladisch via Alsa-user]

Maurizio Mileto Principale wrote:
> Can you tell me if ALSA plans to convert the AE-9 drivers for Linux in the 
> near future?

There does not exist any public information about Creative hardware.
Linux drivers are written by Creative.  Or not, when they don't feel like it.


Regards,
Clemens


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Re: [Alsa-user] Can I use amidi to send (midi) data to the midi input port on a soft-synth?

[Clemens Ladisch via Alsa-user]

Jaime wrote:
> Can I use amidi to send this sysex message to linuxsampler?

No; amidi works only with raw MIDI ports.

Load the snd-virmidi module to get a virtual device that you can
connect to linuxsampler.


Regards,
Clemens


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Re: [Alsa-user] alsa-only applications (such as jitsi) via jackd

[Clemens Ladisch via Alsa-user]

Peter P. wrote:
> Can I set pcm device names per application as defaults?

If the application does not allow you to set the ALSA device name,
you have to replace the device definition that the application used
(probably "default").

It would be possible to create a device definition that reads an
environment variable (the "hw" device does this with ALSA_PCM_*),
but the "default" device does not do this.


Regards,
Clemens


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Re: [Alsa-user] Capturing audio on second capture input

[Clemens Ladisch via Alsa-user]

Christopher Lansing wrote:
> card 0: SB [HDA ATI SB], device 2: ALC892 Alt Analog [ALC892 Alt Analog]

You have to tell whatever program you're using to use this device.
For example, "arecord -D plughw:0,2 -f dat something.wav".


Regards,
Clemens


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Re: [Alsa-user] Capturing audio on second capture input

[Clemens Ladisch via Alsa-user]

Christopher Lansing wrote:
> I can't record from "Capture 1".

How many devices show up in the output of "arecord -l"?


Regards,
Clemens


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Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

[Clemens Ladisch via Alsa-user]

D. R. Evans wrote:
> Clemens Ladisch via Alsa-user wrote on 2/20/20 1:56 PM:
>> Please show the output of "amixer -c0 scontents".
>
> Output attached (it seemed a bit long to put in the body of the e-mail).

Looks OK.

Try changing the Front Panel Audio Enable jumper.
But it does not look as if it were possible to do anything from the
software side.


Regards,
Clemens


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Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

[Clemens Ladisch via Alsa-user]

D. R. Evans wrote:
> Clemens Ladisch via Alsa-user wrote on 2/20/20 12:48 AM:
>> Please show the output of "amixer scontents".
>
> Simple mixer control 'Master',0
>   Capabilities: pvolume pswitch pswitch-joined
>   Playback channels: Front Left - Front Right
>   Limits: Playback 0 - 65536
>   Mono:
>   Front Left: Playback 65536 [100%] [on]
>   Front Right: Playback 65536 [100%] [on]
> Simple mixer control 'Capture',0
>   Capabilities: cvolume cswitch cswitch-joined
>   Capture channels: Front Left - Front Right
>   Limits: Capture 0 - 65536
>   Front Left: Capture 65536 [100%] [on]
>   Front Right: Capture 65536 [100%] [on]

Oops, that's PulseAudio.

Please show the output of "amixer -c0 scontents".

> The motherboard documentation doesn't suggest that anything special is
> necessary

It would be something done automatically by the driver.


Regards,
Clemens


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Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

[Clemens Ladisch via Alsa-user]

D. R. Evans wrote:
> the audio seems to have very low output.

Please show the output of "amixer scontents".

> this issue was raised 11 years ago

"ALC888" does not mean anything; what matters is how the chip is wired up.
(If you hardware requires something special to be configured, it's possible
that nobody knows what that would be.)


Regards,
Clemens


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Re: [Alsa-user] Creative Sound Blaster AE-9 drivers

[Clemens Ladisch via Alsa-user]

Ethan Grizzle wrote:
> Is it being worked on at all, or is it not a priority at the moment?

Neither Creative nor anybody else is writing a driver.


Regards,
Clemens


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Re: [Alsa-user] Connection of multiple headphones to kodi simultaneously (on rpi3)

[Clemens Ladisch via Alsa-user]

Harald Heigl wrote:
> if one of my devices (bluetooth headphones turned off) is
> failing, the whole configuration (another bluetooth headphone+hdmi) is
> failing. Is there an easy way to handle alsa failures of one device
> gracefully?

Not with ALSA plugins.

This should be possible with PulseAudio.


Regards,
Clemens


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Re: [Alsa-user] Routing hw PCM

[Clemens Ladisch via Alsa-user]

Gabriele Zampieri wrote:
> My goal is to bridge those streams (|bt-capture -> gsm-playback| and 
> |gsm-capture -> bt-playback|) in a fullduplex configuration to handle calls. 
> In fact, the bluetooth will be connected to an handset.
>
> I was testing different asound.conf configuration, but I cannot figure how to 
> automatically route those streams.

The asound.conf definitions are used when a program accesses a PCM stream.
Without a program, nothing happens.

You have to use a tool like alsaloop.


Regards,
Clemens


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Re: [Alsa-user] a mono phone interface

[Clemens Ladisch via Alsa-user]

Alfred Ganz wrote:
> I have an old Skype phone that I have used under Centos-6 that I have not
> been able to get to work under Centos-7 and Centos-8. I have tested it
> under Centos-6 (2.6.32-754.6.3.el6.i686 #1 SMP) with:
> aplay -D usb-phone -t wav /usr/share/sounds/alsa/Front_Center.wav
> and it works with voice output saying "Front Center".
>
> When I try the same under Centos-7 (3.10.0-1062.9.1.el7.x86_64 #1 SMP)
> and Centos-8 (4.18.0-80.11.2.el8_0.x86_64 #1 SMP), I get two short
> rumbling segments of noise (That I suspect being the not properly
> processed words from the sound file).

What is the output when you add -v to the aplay command line?
If you still have the old Centos, is there any difference between
the working and broken -v outputs?


Regards,
Clemens


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Re: [Alsa-user] a mono phone interface

[Clemens Ladisch via Alsa-user]

Alfred Ganz wrote:
> HW Params of device "usb-phone":

What are the capabilities of the actual hardware ("hw:1")?


Regards,
Clemens


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Re: [Alsa-user] need hardware suggestion

[Clemens Ladisch via Alsa-user]

Dmitri Seletski wrote:
> My Xonar DX is dying long painful death(power port is loose).

It should be possible to resolder it (for somebody who has experience).

> HD 598, cushions break apart and don't seem replacable, somewhat loose cable.

Parts that wear out are designed to be replacable:
https://en-de.sennheiser.com/accessories--hd-598--ear-cushion
https://en-de.sennheiser.com/accessories--hd-518--hd-558--hd-598--connection-cable
(also available on Amazon etc.)

> ideally with low latency(for gaming). Ideally USB2/3

PCI(e) can have lower latency than USB.

Anyway, the Zowie Vital is reported to be expensive but good.


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
>>> openat(AT_FDCWD, "/dev/snd/pcmC0D0c", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 
>>> ENOENT (No such file or directory)

Are there ACL restrictions on this file? ("getfacl /dev/snd/pcmC0D0c")

Does accessing the device node directly work ("cat /dev/snd/pcmC0D0c"), i.e.,
is the resulting error something else than "No such file or directory"?


Regards,
Clemens


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Re: [Alsa-user] connecting ALSA from a VM to the host?

[Clemens Ladisch via Alsa-user]

Rich Morin wrote:
> I'd like to set up ALSA to use the host's audio output system.

You have to configure the VM to provide a (virtual) sound hardware device
(https://www.virtualbox.org/manual/ch03.html#settings-audio), or to pass
through a USB audio device from the host.


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
> On Thu, Dec 12, 2019 at 8:01 AM Clemens Ladisch  
> wrote:
>> Then I cannot explain the "No such file or directory" message.
>> What is the output of "strace -e open arecord -D plughw:1 test.wav"?
>
> openat(AT_FDCWD, "/dev/snd/pcmC0D0c", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 
> ENOENT (No such file or directory)

pcmC0* is card 0, a different device.

Did you really use "plughw:1"?  Did you make any modifications to the "hw"
or "plughw" definitions in one of the ALSA configuration files?


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
> On Thu, Dec 12, 2019 at 1:35 AM Clemens Ladisch  
> wrote:
>> Does your user have access rights to all device nodes in there?
>
> yes, I am part of the audio group:
>
> $ ls -l /dev/snd/
> crw-rw+ 1 root audio 116,  2 Dec 12 05:34 pcmC1D0c

Then I cannot explain the "No such file or directory" message.
What is the output of "strace -e open arecord -D plughw:1 test.wav"?


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
> $ arecord -D plughw:1 test.wav
> arecord: main:828: audio open error: No such file or directory

What is the output of "ls -l /dev/snd/"?
Does your user have access rights to all device nodes in there?


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
> $ arecord -D plughw:1 test.wav
> arecord: main:828: audio open error: No such file or directory

What is the output of "arecord -l"?


Regards,
Clemens


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Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

[Clemens Ladisch via Alsa-user]

Jesse Hathaway wrote:
> I just purchased a Kingston HyperX Amp USB soundcard[1]. The microphone input
> does not work correctly in Alsa

What exactly does not work?

What happens when you run "arecord -D plughw:1 test.wav"?

> When I try to choose it with alsamixer

alsamixer just shows mixer controls; choosing a device in alsamixer
has no effect on the rest of the system.

> $ alsamixer -c 1
> cannot load mixer controls: Invalid argument

Any error messages in the system log when this happens?


Regards,
Clemens


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Re: [Alsa-user] 'Supported bit depths' according to alsa-lib seem to differ from real values

[Clemens Ladisch via Alsa-user]

f...@interkatz.de wrote:
> according to hda-verb, the supported capturing parameters are 16/20/24
> bits per sample

These are the sample formats that are transmitted on the HDA bus.

> Alsa-lib, on the other hand, claims that the supported formats (for
> `hw:0,0` in this case) are S16LE and S32LE

These are the sample formats that the HDA controller writes into memory,
and that your application sees in the buffer.

> I expected that ALSA returns something like: S16LE, S20LE, S24LE.

S20 and S24 are smaller samples that are aligned at the least
significant bit in the 32-bit memory word.  This would make them harder
to handle, and almost no DMA controller is stupid enough to use them.

The 20- or 24-bit samples you get from HDA are aligned at the most
significant bit, i.e., you can handle them like 32-bit values.  If you
really want to know how many bits are actually significant, call
snd_pcm_hw_params_get_sbits() after you've set the hw params.


Regards,
Clemens


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Re: [Alsa-user] USB Sound Card & default device?

[Clemens Ladisch via Alsa-user]

Hans Malissa wrote:
> Can anyone tell me how to configure CMUS to use the Fulla 3 or the internal 
> sound card, no matter how ALSA sorts the cards?

ALSA itself has no built-in mechanism to dynamically select a fallback device.

I'd recommend to use PulseAudio:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/DefaultDevice/


Regards,
Clemens


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Re: [Alsa-user] Using the multi plugin with a device with different number of inputs and outputs

[Clemens Ladisch via Alsa-user]

Francesco Cameli wrote:
> Is there a way to create a multi device using a different input and output 
> channels number?

Only by creating two devices, one for each direction:

pcm.multiES8 {
type asym
playback.pcm {
type multi
...
}
capture.pcm {
type multi
...
}
}


Regards,
Clemens


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Re: [Alsa-user] arecord format-conversion

[Clemens Ladisch via Alsa-user]

Alex Vazquez wrote:
> pcm.!default {
> ...
>
> arecord -r 16000 -f S16_LE -c 1 -d 3 rec.wav
> Recording WAVE 'rec.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono
> arecord: set_params:1361: Unable to install hw params:

Please check with the -v option whether it actually uses your "default" device.


Regards,
Clemens


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Re: [Alsa-user] alsa not seeing sound cards

[Clemens Ladisch via Alsa-user]

Brian J. Murrell wrote:
> # aplay -l
> aplay: device_list:272: no soundcards found...

Are the sound drivers loaded? (see the output of "lsmod")
Are the devices accessible? (see "ls -l /dev/snd/")


Regards,
Clemens


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Re: [Alsa-user] Unable to see MIDI keyboard with the aconnect -i command

[Clemens Ladisch via Alsa-user]

Guy Tucotte wrote:
> The command 'amidi -l' shows the device
> BUT, the command 'aconnect -i' doesn't show the device

Does it help to re-plug the device?

> The snd_rawmidi module as been compiled inside the kernel image.

Why only some modules?  I don't know if it is related, but please
try to have all snd* modules compiled in, or none.


Regards,
Clemens


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Re: [Alsa-user] USB midi interface named 'USB Device 0x86a'

[Clemens Ladisch via Alsa-user]

Stefan Sauer wrote:
> 'lsusb' shows the device as:
> Bus 006 Device 003: ID 086a:0001 Emagic Soft- und Hardware GmbH Unitor8
>
> 'amidi -l' shows a human readable name for all midi devices, but the 'Emagic
> Unitor amt 8' (hw:4,0) show up with a generic name:
> Dir DeviceName
> IO  hw:4,0,0  USB Device 0x86a:0x01 MIDI 1

lsusb has a database of names.

The snd-usb-audio driver asks the device for what its name is.  You can
see these values in the output of "sudo lsusb -v" for "iManufacturer"/
"iProduct".  (Please check this.)  When these values cannot be read, you
get that "USB Device" name.

It is possible to override this name by adding a .product_name entry
in the sound/usb/quirks-table.h file and recompiling the kernel.
(At least the MT4 does not need this override.  If the Unitor8 or the
AMT8 require it, this change can be made permanent.)


Regards,
Clemens


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Re: [Alsa-user] human readable names for audio and midi channels

[Clemens Ladisch via Alsa-user]

Stefan Sauer wrote:
> when using multi channel audio interfaces, it would be awesome to be
> able to label the ports to make it easier to select the right (pair of)
> channels in the software.

ALSA has no mechanism to associate names to multiple channels belonging to
the same PCM device.

> Likewise for multi midi interfaces (like the classic emagic amt8).

It is possible for each port to have its own name:

$ aplaymidi -l
 PortClient namePort name
 ...
 40:0SC-8820SC-8820 Part A
 40:1SC-8820SC-8820 Part B
 40:2SC-8820SC-8820 MIDI

However, only the port's owner can change the name, so the USB audio
driver would have to know about it.

> Are there any thoughts on this to support this on the Alsa level?

These labels should survive a reboot, but are not a property of the hardware,
so they must be stored somewhere else.  It is not clear who should be
responsible for that.

And if there were a new API, most programs would not know of it.


Regards,
Clemens


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Re: [Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability

[Clemens Ladisch via Alsa-user]

Nicolas wrote:
> https://www.esi-audio.com/products/gigaporthd+/
>
> As I understand it, there is no specific Linux driver for this DAC as there
> is one in Windows.

The web site claims native CoreAudio supports, so there should not be any
secret sauce.

Please show the contents of /proc/asound/card?/stream0.


Regards,
Clemens


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Re: [Alsa-user] CM6206-LX linein mute ?

[Clemens Ladisch via Alsa-user]

The "PCM Capture Source" control is currently set to "Mic"; try the others.


Regards,
Clemens


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Re: [Alsa-user] CM6206-LX linein mute ?

[Clemens Ladisch via Alsa-user]

Luca Toldo wrote:
> Simple mixer control 'PCM Capture Source',0
>   Items: 'Mic' 'Line' 'IEC958 In' 'Mixer'

Please try all settings.


Regards,
Clemens


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Re: [Alsa-user] CM6206-LX linein mute ?

[Clemens Ladisch via Alsa-user]

Luca Toldo wrote:
> It works fine when I connect the USB Sound microusb to my MacBook, however 
> when I try to connect it to the Raspberry, it is mute.

Please show the output of "amixer".

> https://www.raspberrypi.org/forums/viewtopic.php?f=38=233817=1430727

All three inputs _except_ the line input are enabled.

> I feel the problem is with the configuration in the sound/usb/quirks.c that 
> are valid for the CM6206 but do not differentiate for the LX

Those settings should not affect the line input.


Regards,
Clemens


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Re: [Alsa-user] Thunderbolt audio driver?

[Clemens Ladisch via Alsa-user]

Jacob Juul Klejs Kolding wrote:
> All I know is that the main chip is an FPGA.
>
> Are there any tools for Windows that can snif the communication?

For PCI(e), this cannot be done in software.


Regards,
Clemens


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Re: [Alsa-user] Thunderbolt audio driver?

[Clemens Ladisch via Alsa-user]

Jacob Juul Klejs Kolding wrote:
> Are there currently any Thunderbolt audio devices supported in Linux?

>From a software point of view, Thunderbolt is just a PCIe bus.
So this depends on what PCI(e) audio chip the device is using.


Regards,
Clemens


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Re: [Alsa-user] A query on the support status for the headset jack on the Clevo P751DM2-G (and other systems) with the Realtek ALC892+TI Burr Brown OPA2134 operational Amplifier

[Clemens Ladisch via Alsa-user]

Dennis Mungai wrote:
> One thing I've never been able to get to work, even with hdajackretask (
> https://fossies.org/linux/alsa-tools/hdajackretask/README), is the
> headphone jack.

We'd have to know how the hardware is wired up.

> The oddity here is that this Texas Instruments device (OPA2134 OP Amp) does
> appear under USB devices (lsusb).

What exactly does lsusb show?


Regards,
Clemens


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Re: [Alsa-user] Stereo Output on Asus Xonar DGX

[Clemens Ladisch via Alsa-user]

Chad Skeeters wrote:
> It plays "Front Left", "Front Right" out of *both* speakers of headphones
> connected to the green headphone port on the back of the card.  Then it
> plays "Rear Left", "Rear Right" out of both speakers of headphones
> connected to the black surround port on the back of the card.

The channels are entirely separated in the hardware.

This is likely to be a problem with the connector (not plugged in all the
way?) or with the headphones.


Regards,
Clemens


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Re: [Alsa-user] ALSA on non-Linux systems

[Clemens Ladisch via Alsa-user]

Kevin Smallman wrote:
> is that down to the hardware support provided by the kernel, or something 
> like ALSA or OSS?

ALSA consists of three parts:
- the userspace library, alsa-lib;
- the ALSA framework in the kernel; and
- various ALSA drivers, also in the kernel.

The library can use other backends (such as OSS or PulseAudio), so it might
make sense to compile it on other OSes.  However, this does not help you
if you want to use the ALSA FireWire driver.

FFADO does not use ALSA but a generic FireWire driver, but that is Linux
specific, too.


Regards,
Clemens


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Re: [Alsa-user] List devices

[Clemens Ladisch via Alsa-user]

Robert Bielik wrote:
> I cannot get it listed with ALSA API function snd_device_name_hint.

This is undocumented; see the "hint" entries in the standard .conf files.


Regards,
Clemens

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Re: [Alsa-user] aplay weirdness

[Clemens Ladisch via Alsa-user]

Philip Rhoades wrote:
> This produces a crashing static sound:
>
>   espeak --stdout 'words to speak' | aplay
>
> but this works as expected:
>
>   espeak --stdout 'words to speak' > ./t
>   aplay ./t

Is there a difference in the output of "hexdump -C -n48 ./t" and
"espeak --stdout 'words to speak' | hexdump -C -n48"?


Regards,
Clemens

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Re: [Alsa-user] MIDI events of custom USB device not appearing in aseqdump

[Clemens Ladisch via Alsa-user]

Kevin Cuzner wrote:
> My device presents a USB Midistreaming interface with the following
> characteristics:
> - A single embedded MIDI IN jack with ID 1
> - A single embedded MIDI OUT jack with ID 2
> - A single element with ID 3
>   - The embedded MIDI IN jack is attached to the single input pin.
>   - The embeeded MIDI OUT jack is attached to the single output pin.

Please show the output of "lsusb -v" for your device.

> My device is the "Midi-Fader". When I run "aseqdump -p 28:0" I see
> nothing appear.

Try amidi, which will show even invalid bytes.

> However, Wireshark shows that my device is sending USB-MIDI Event
> Packets (just Bulk IN packets with a length of 4) with the following
> binary content:
>
> 2b 07 31 00

07 is not a valid MIDI status byte.  The first byte of the USB-MIDI
Event Packet is not MIDI; the remaining bytes must be valid MIDI.  See
the second example in table 4-2.

> - Cable 2

Section 4 of the USB MIDI specification says:
| The Cable Number (CN) is a value ranging from 0x0 to 0xF indicating
| the number assignment of the Embedded MIDI Jack associated with the
| endpoint that is transferring the data.

That "number assignment" refers to the following algorithm (which
everybody pretends is obvious): the cable numbers of an endpoint are
from zero to one less than bNumEmbMIDIJack.

> I have a Yamaha MOXF8, but it appears to enumerate as a custom device
> and does not use the standard USB-MIDI interface.

Yamaha devices use the standard protocol, and even have a list of
standard-looking jack descriptors; they just claim to be vendor-specific
devices in order to prevent being used by the built-in Windows driver.


Regards,
Clemens

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Re: [Alsa-user] MOTU Microbook iic support

[Clemens Ladisch via Alsa-user]

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] Bower's & Wilkins PX headphone - snd-usb-audio error

[Clemens Ladisch via Alsa-user]

a...@philippludwig.net wrote:
> usb 3-4: New USB device found, idVendor=19b5, idProduct=0021, bcdDevice=25.20
> usb 3-4: Product: PX USB
> usb 3-4: Manufacturer: Bowers & Wilkins
> snd-usb-audio: probe of 3-4:1.0 failed with error -22
> usb 3-4: 2:1: cannot get min/max values for control 2 (id 2)
> snd-usb-audio: probe of 3-4:1.1 failed with error -22
>
> Any advice on how I could debug this further?

Please show the output of "lsusb -v" for this device.


Regards,
Clemens

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Re: [Alsa-user] USB Synchronization w/ALSA via g_audio

[Clemens Ladisch via Alsa-user]

Rob wrote:
> I am capturing data from my microphone, and transmitting it via
> pcm_write() to the USB gadget card, and arecording on the USB host.
> If I query the buffer, I begin to see a creeping fill on the device
> side buffers, such that after a few hours of constant streaming, the
> buffer is filled, and my pcm_write is returning "we only wrote SOME
> of your data".

The ALSA API was designed for 'normal' sound cards where only the
hardware itself can control the sample clock.  This means that the USB
audio gadget does not give you control over its clock, and that samples
are moved synchronously with the USB bus clock.

You'd have to measure the clock difference, and dynamically resample
the stream.  (Zita-ajbridge does something like this.)

Alternatively, it might be possible to extend the audio gadget driver
to add a mechanism to slightly adjust the sample rate, relative to the
bus clock.


Regards,
Clemens

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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] 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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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:
> 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]

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] 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

[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] 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] 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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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] 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 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:
> 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] 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] 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] 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] 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] 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] 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] 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] Iec958 plugin usage

[Clemens Ladisch via Alsa-user]

maruthi srinivas wrote:
> On Monday, January 22, 2018, Clemens Ladisch <cladi...@googlemail.com>
> 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] 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] 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]

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 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] 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] 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] 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] 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] 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]

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:
> 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] [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] 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] 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] 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] 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] 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] 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:
>> 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:
>> 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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