:(parse_args)
> Unknown parameter 1
Oops, the hdmi device does not take a second parameter.
Use:
playback.pcm "plug:hdmi:0"
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playback.pcm "plug:hdmi:0,0"
capture.pcm "plughw:1"
}
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ady 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/card
t that your hardware supports.
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ces) work?
> In fact, ALSA "sees" my USB keyboard as an audio card, which sounds odd
This is normal.
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e on the D2X card is not known to have such problems.
Please show the output of "lspci -tvv".
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a custom kernel.
Are you paying someone for support? Or did you build it yourself?
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ve heard, the Pi's USB controller has limited bandwidth, and its
driver is not able to distribute all of it among multiple streams.
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pect recording always uses channels.
Please show the contents of /proc/asound/card2/stream0.
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st
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
d and play to
the actual device.
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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.
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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".
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.
ass compliant.
Does it work without a separately installed driver in Windows and OS X?
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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.
Regar
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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that reads an
environment variable (the "hw" device does this with ALSA_PCM_*),
but the "default" device does not do this.
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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".
Christopher Lansing wrote:
> I can't record from "Capture 1".
How many devices show up in the output of "arecord -l"?
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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
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 - F
configured, it's possible
that nobody knows what that would be.)
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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,
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ht
ossible with PulseAudio.
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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
__
erly
> 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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Alfred Ganz wrote:
> HW Params of device "usb-phone":
What are the capabilities of the actual hardware ("hw:1")?
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Zowie Vital is reported to be expensive but good.
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"cat /dev/snd/pcmC0D0c"), i.e.,
is the resulting error something else than "No such file or directory"?
Regards,
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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.
Reg
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
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
Th
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?
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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er 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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gnificant, call
snd_pcm_hw_params_get_sbits() after you've set the hw params.
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op.org/wiki/Software/PulseAudio/Documentation/User/DefaultDevice/
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multi
...
}
}
Regards,
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hether it actually uses your "default" device.
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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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now if it is related, but please
try to have all snd* modules compiled in, or none.
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(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
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,
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/proc/asound/card?/stream0.
Regards,
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The "PCM Capture Source" control is currently set to "Mic"; try the others.
Regards,
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Luca Toldo wrote:
> Simple mixer control 'PCM Capture Source',0
> Items: 'Mic' 'Line' 'IEC958 In' 'Mixer'
Please try all settings.
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l 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
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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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,
C
uments device (OPA2134 OP Amp) does
> appear under USB devices (lsusb).
What exactly does lsusb show?
Regards,
Clemens
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nected 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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Alsa-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,
C
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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;hexdump -C -n48 ./t" and
"espeak --stdout 'words to speak' | hexdump -C -n48"?
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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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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.
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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lightly adjust the sample rate, relative to the
bus clock.
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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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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"
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
Andreas Böhler wrote:
> all recordings are completely silent
Please show the output of "amixer -c2".
Regards,
Clemens
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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 l
e 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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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
uration in the driver?
Regards,
Clemens
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then re-load the driver (modprobe).
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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 k
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
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.
>
USB audio driver does not have these settings.
Regards,
Clemens
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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,
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by software.
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like this:
https://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg12236.html
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ng the device ?
Every application has its own instance of the Jack plugin. Doesn't Jack
mix the streams together?
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sactl 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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nalogue Surround 4.0" classification is done by
PulseAudio, so it is not clear what problem you actually have.
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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,
C
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 l
convert either way and write to a file.
What for? There is no standard for AES data in a file.
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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
S/PDIF
on some very old C-Media cards.
This plugin is useless if you have any other hardware.
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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" plu
write_pcm_data_from_buffer(playback_handle, output_buffer);
> }
> }
Are you filling the output before the loop?
Regards,
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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,
Clem
t, 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 SUB
put work?
>
> how to test that?
Connect anything to the line input, and set it as source.
Regards,
Clemens
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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
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 show
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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> 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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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
s occuring during the playback of
> a single file will be reported separately?
Yes.
Regards,
Clemens
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in 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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ugh 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,
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u know how to do that.
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ws
> "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 t
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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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 t
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