Re: [Alsa-user] [External] Re: No sound on Lenovo P1 Gen 6 (00:1f.3 Multimedia audio controller: Intel Corporation Device 51ca (rev 01))

[Ralf Mardorf]

Hi,

if Ubuntu supports your computer, they probably offer a patch or
patches. If so and you build a "vanilla" kernel by just using an Ubuntu
config, then you build without any additional patch that might be (or
might not be) offered by Ubuntu.

IIUC a default install of Ubuntu, Redhat or SuSE might provide a working
audio device, see https://support.lenovo.com/us/en/solutions/ht082374 .
It might not necessarily work, due to different mainboard releases.

How about testing a live media, e.g. Ubuntu from an USB stick?

Did you already check
https://support.lenovo.com/us/en/solutions/ht511743-how-to-download-the-linux-image-from-the-e-support-page
?

I build my own desktop machines. For my 13th Gen Intel Core based
machine Ubuntu offered a kernel supporting everything I need, already
when the machine was new, while for Arch Linux I build the kernel module
for RTL8125 using dkms.

To summarise, if you build your own kernel, you may need one or more
patches in addition to the source code from kernel.org, a kernel
configuration will probably not change anything. You don't necessarily
have to rebuild the whole kernel, but only the corresponding module or
modules, simply with dkms.

Regards,
Ralf


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Re: [Alsa-user] Archeology/ future of alsamixergui

[Ralf Mardorf]

On Sat, 2022-10-29 at 11:59 +0100, Patrick May wrote:
> I wouldn't call it "discontinued" when so many people and existing & new 
> projects are still using it. Imo it's a testament to how much gnome 
> messed things up starting with GTK3.

Hi,

while gtk2 is still available by almost all, if not all Linux distros, a
lot of gtk2 apps already suffer from serious issues, at least on some
machines. De facto gtk2 is EOL.

Fortunately it's possible to workaround the most annoying gtk3 oddities,
at least for users who have the time to do so. However, even gtk3 is
outdated, but still supported. The migration to gtk4 is already done by
several apps and I doubt that it's still possible to workaround the
oddities of gtk4. It's probably even not possible anymore for users who
can spend a lot of time in maintaining their computer environments.

FWIW the GNOME foundation shuts down all mailing lists hosted at
gnome.org end of this month and as a replacement it only offers the most
bizarr instance of a Discourse forum I've ever seen, with a gamification
overdose and a broken mailing list feature.

Starting a new project based upon gtk2 is foolish. Old projects based
upon gtk2 soon or later need to be rewritten or they will go the way of
the dodo.

I'm not an alsamixergui user, but IMO FLTK was a way better choice than
gtk2 is.

Regards,
Ralf


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Re: [Alsa-user] Archeology/ future of alsamixergui

[Ralf Mardorf]

On Sat, 2022-10-29 at 10:25 +0200, Benno wrote:
> Luckily found this https://dev1galaxy.org/viewtopic.php?id=5232, 
> compiled right away and is fully fledged. A seamless integration with 
> XFCE toolbar. Contacted the author, too.

Hi,

it does use the discontinued gtk2 ;).

[rocketmouse@archlinux amixer-gtk-0.1]$ grep gtk debian/control 
Source: amixer-gtk
 libgtkmm-2.4-dev (>= 1:2.24.5-4),
Homepage: https://gitea.devuan.dev/aitor_czr/amixer-gtk
Package: amixer-gtk
  libgtkmm-2.4-1v5 (>= 1:2.24.5-4),
 Amixer-gtk is a desktop mixer application for ALSA's "Simple Mixer Interface"

Regards,
Ralf


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Re: [Alsa-user] Archeology/ future of alsamixergui

[Ralf Mardorf]

On Sat, 2022-10-29 at 09:36 +0200, Benno wrote:
> Stemming from the Alsamixer sources (back in 2001/2? If I read the 
> sources correctly a copy of alsamixer was enhanced with a switch for 
> FLTK in addition to ncurses.)

Hi,

consider to ask for the FLTK version at
https://github.com/alsa-project/alsa-utils/issues .

Maarten de Boer' email account (mdeboer_AT_iua.upf.es) is most likely
not active anymore [1].

   [rocketmouse@archlinux ~]$ grep -emixer -eBuild 
/usr/share/doc/alsa-utils/README.md
   ![Build 
alsa-utils](https://github.com/alsa-project/alsa-utils/workflows/Build%20alsa-utils/badge.svg?branch=master)
   amixer   | a command line mixer
   alsamixer| a ncurses mixer
   
A FLTK GUI isn't mentioned andalsamixer --helpdoesn't show a
switch on my install. Did you read the alsamixer man page? I doubt that
it mentions a switch, but I haven't read the man page.

A little bit of googling:

The website https://command-not-found.com/alsamixergui shows also a dead link: 
http://www.iua.upf.es/~mdeboer/projects/alsamixergui/

https://ccrma.stanford.edu/mirrors/lalists/lad/2001/Nov/0099.html from 2001, 
containing the email address Maarten de Boer (mdeboer_AT_iua.upf.es).

Regards,
Ralf

[1]
   rocketmouse@archlinux ~]$ ping www.iua.upf.es
   ping: www.iua.upf.es: Temporary failure in name resolution
   [rocketmouse@archlinux ~]$ dig www.iua.upf.es
   
   ; <<>> DiG 9.18.8 <<>> www.iua.upf.es
   ;; global options: +cmd
   ;; Got answer:
   ;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 8091
   ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 1
   
   ;; OPT PSEUDOSECTION:
   ; EDNS: version: 0, flags:; udp: 512
   ;; QUESTION SECTION:
   ;www.iua.upf.es. IN  A
   
   ;; ANSWER SECTION:
   www.iua.upf.es.  37211   IN  CNAME   dtic-ha.ccp.upf.edu.
   
   ;; Query time: 93 msec
   ;; SERVER: 192.168.1.1#53(192.168.1.1) (UDP)
   ;; WHEN: Sat Oct 29 10:29:17 CEST 2022
   ;; MSG SIZE  rcvd: 76
   
   [rocketmouse@archlinux ~]$ dig http://www.iua.upf.es
   
   ; <<>> DiG 9.18.8 <<>> http://www.iua.upf.es
   ;; global options: +cmd
   ;; Got answer:
   ;; ->>HEADER<<- opcode: QUERY, status: NXDOMAIN, id: 33042
   ;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 1, ADDITIONAL: 1
   
   ;; OPT PSEUDOSECTION:
   ; EDNS: version: 0, flags:; udp: 512
   ;; QUESTION SECTION:
   ;http://www.iua.upf.es.  IN  A
   
   ;; AUTHORITY SECTION:
   upf.es.  10573   IN  SOA ns1.upf.es. 
postmaster.upf.edu. 2022102601 86400 7200 2592000 172800
   
   ;; Query time: 10 msec
   ;; SERVER: 192.168.1.1#53(192.168.1.1) (UDP)
   ;; WHEN: Sat Oct 29 10:29:20 CEST 2022
   ;; MSG SIZE  rcvd: 108
   



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Re: [Alsa-user] Archeology/ future of alsamixergui

[Ralf Mardorf]

On Sat, 2022-10-29 at 07:36 +0200, Benno wrote:
> Is there a simple graphical alternative for Alsa in general? (Xfce's mixer 
> applet cannot unmute/ mute and offers only main control. Envy24* and 
> successors only work for ICE-chips.)

Hi,

in my experiences alsamixer without GUI was always fishy when using
ice1712 cards. I used either envy24control from alsa-tools (
https://alsa-project.org/ ) or mudita24 (
http://code.google.com/p/mudita24/ ,
https://github.com/NielsMayer/mudita24 ). I can't comment on other ice
cards or non-ice Envy cards.

On the Arch User Repository website I found
https://aur.archlinux.org/packages/alsamixergui providing just a dead
link, but the PKGBUILD loads the source and a patch from Debian (
https://aur.archlinux.org/cgit/aur.git/tree/PKGBUILD?h=alsamixergui ):

http://ftp.de.debian.org/debian/pool/main/a/${pkgname}/${pkgname}_${pkgver}-1.orig.tar.gz
http://ftp.de.debian.org/debian/pool/main/a/${pkgname}/${pkgname}_${pkgver}-1-9.1.diff.gz

IOW:

http://ftp.de.debian.org/debian/pool/main/a/
http://ftp.de.debian.org/debian/pool/main/a/alsamixergui/

Regards,
Ralf








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Re: [Alsa-user] Characterising a fault with audio playback

[Ralf Mardorf]

On Wed, 2022-10-26 at 20:24 +1300, Paul Dorman wrote:
> one of the USB3 ports

Hi,

did you test different USB ports?

> Whatever is causing it, the problem has persisted through OS upgrades
> and the switch from Pulseaudio to Pipewire.

I'm in favour of using plain ALSA or jackd. For testing purpose don't
use a sound server, test with plain ALSA.

Regards,
Ralf


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Re: [Alsa-user] Setting a PCM as default device - SOLVED

[Ralf Mardorf]

On Wed, 20 Apr 2022 14:57:26 -1000, Joel Roth wrote:
>pcm.!default {
> type asym
> playback.pcm "plughw:CARD=PCH,DEV=7"
>}

Hi,

½ a year ago I started using the following .asoundrc in my home folder
to automatically access HDMI audio after log in on a plain ALSA (and
jackd, but no pulseaudio) install.

[rocketmouse@archlinux ~]$ cat .asoundrc
defaults.pcm.card 3
defaults.pcm.device 7

Unfortunately I can't start jackd to access another device anymore, but
since I'm doing almost all pro-audio related work with Apple, the crappy
display's speaker sound is usually ok for the Linux machine.

To access another device I need to 

[rocketmouse@archlinux ~]$ mv -i .asoundrc .asoundrc_disabled

then log out and in again to use one of devices in alsa-base.conf or an
USB device.

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

I've done the mv and log out and in right now for the very first time.
At least there's no need to reboot :D.

Regards,
Ralf


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Re: [Alsa-user] Cheap USB jack output device

[Ralf Mardorf]

>On 04/03/2022 13:32, Josu Lazkano wrote:
>> Could you recommend any well supported USB output device?

Any _class compliant_ usb audio interface should work.


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Re: [Alsa-user] Line-in volume

[Ralf Mardorf]

On Tue, 21 Dec 2021 19:56:32 -0800, Paul A. Steckler wrote:
>I'd like to play sound from a device connected to my sound card's
>line-in.

So the signal chain is

[DEVICE analog out]->[analog in SOUND CARD analog-to-digital 
converter]->[Linux]->[digital-to-analog converter SOUND CARD analog 
out]->[analog in ACTIVE SPEAKERS without volume control]

?

On Wed, 22 Dec 2021 10:12:21 -0800, Paul A. Steckler wrote:
>On Tue, Dec 21, 2021 at 11:23 PM Ralf Mardorf wrote:
>> Please describe in what way the signal is "too loud". Is the signal
>> distorted? Is the speaker's/headphone's sound too loud?  
>
>The sound from the speakers is too loud, although not distorted.
>
>There are attenuator devices you can purchase, which might be a
>solution. I'm hoping for a software solution, though.

If the above sound chain is correct, then you don't need to adjust
the "line in" level. You need to adjust the output level. If you can't
do it by the audio device, then yes, you can use software. Btw. an
attenuator device could be a simple potentiometer or a cheap mixer.

Why is the "device" connected to the Linux computer? Aren't you already
using software that allows to adjust the output level?

It's not unusual that you can't adjust levels using alsamixer, e.g.
when using a class compliant USB audio interface. I can't comment on
pulseaudio, since I never ever would install it. I'm using either jackd
or plain ALSA, but to adjust speaker volume I'm using an analog
amplifier.






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Re: [Alsa-user] Line-in volume

[Ralf Mardorf]

On Tue, 21 Dec 2021 19:56:32 -0800, Paul A. Steckler wrote:
>I'm running Linux Mint 20.2 (x64 / Cinnamon).
>
>I'd like to play sound from a device connected to my sound card's
>line-in. I'm able to do so by enabling the loopback device in
>alsamixer. But even with the line-in level and line-in booster level
>set to 0, the volume is too loud.
>
>Adjusting the line-in volume level in pavucontrol seems to have no
>effect.
>
>Is there a way to reduce the volume level to an acceptable level?

Hi,

roughly speaking: As long as the analog input circuit is fed with an
appropriate analog audio signal and the digital input domain doesn't
exceed 0 dBFS, the signal is "acceptable". Attenuation is only required
when mixing (especially non-floating point) digital audio signals, but
when doing this, you are anyway using e.g. the digital mixer of a DAW,
that allows to adjust the signals.

You probably want to adjust the output signal of your audio interface
or better, just the level of the speaker's/headphone's analog amplifier.

Or else, if the analog output of the device connected to the analog
input of your digital audio interface doesn't fit, you might need to
convert impedance.

IOW I can hardly imagine something that requires to adjust line in
levels, apart from some exceptions, for this you will use gear that
provides to adjust the analog input levels of your digital audio
interface. Let alone that for such a setup, you probably want to remove
pulseaudio.

Please describe in what way the signal is "too loud". Is the signal
distorted? Is the speaker's/headphone's sound too loud? That each and
any app has got volume controls nowadays is a silly custom. Only very
experienced audio engineers know how to use layers of level controls.
When connecting line out with line in typically no "line" level control
is needed at all.

Regards,
Ralf
-- 
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― Charles Ives 


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Re: [Alsa-user] Volume Control for Focusrite Scarlett 2i2

[Ralf Mardorf]

On Thu, 9 Dec 2021 15:31:19 -0600, Terry O'Connor wrote:
>I simply get the message "This sound device does not have any
>controls."

Hi,

this is a class compliant device, hence there are usually no volume
controls at all.

For a 1st generation 2i2 a mixer is available, see
https://github.com/x42/scarlett-mixer . It might work with a 2nd and
3rd generation 2i2, too.

It's a long time ago that I tested it with a 2nd generation 18i20, but
it didn't work for this device.

However, there's absolutely no need at all for Scarlett mixer volume
controls, unless you want to rout for monitoring purpose. Loudness and
distortion are caused by something, that is wrong with your setup, so
you need to fix that.

Regards,
Ralf

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Re: [Alsa-user] RME Fireface 802

[Ralf Mardorf]

On Sun, 11 Jul 2021 22:48:21 +0200 (CEST), Mika Kilpi wrote:
>Are there any RME 802 drivers available? And is there a replacement
>option for Totalmix FX?

Hi,

regarding
https://www.rme-audio.de/search-results.html?keywords=class+compliant
the device is USB class compliant. In this case you might get no mixer
for Linux at all, but it might work without a mixer. I can't comment on
Firewire.

AFAIK for Linux there's no modern version of TotalMix available. An
ancient version of TotalMix, without FX at all, is provided by
hdspmixer, but it's unlikely usable in class compliant mode.

I never got all ADAT channels working on Linux for my HDSPe AIO, a PCIe
card. However, for the channels that are working with Linux, hdspmixer
works.

Nice, https://rme-audio.com/fireface-802.html mentions TotalMix FX for
iPad. I migrated for most audio related purposes from Linux to
iOS/iPadOS and I already planned to migrate from a Focusrite Scarlett
18i20 2nd Gen to a RME class compliant audio interface.

However, I would use an USB class compliant RME audio device for Linux
and iOS/iPadOS without a mixer available. Actually I'm using the
Focusrite without a mixer, too. For my needs it has got some minor
inconveniences related to monitoring, that are solvable by workarounds.

Regards,
Ralf


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Re: [Alsa-user] USB sound card with Mic Input

[Ralf Mardorf]

On Sat, 10 Jul 2021 03:35:47 +0100, Dmitri Seletski wrote:
>I have paid attention to Magni Heresy, as per suggestion. Seems like a 
>good AMP.
>
>Focusrite Scarlett 2i2 - XLR mic, i have 3.5mm, so not much use for me 
>at this time... And it seems oriented at music semi-pro/semi-pro
>youtubers.

Hi,

you mentioned "I am beginner audiophile", what ever "audiophile" is
for, you probably want a decent sound. I've got a similar (more or less
equal) setup as Mark, a RME HDSPe AIO and for USB a Focusrite Scarlett
18i20 2nd Gen.

If you really want decent audio quality get a RME device, for USB it
would be a RME Babyface. For just one headphone, you don't need a
headphone amp, since the RME headphone outputs are known to be
excellent. RME audio devices are professional grade, as reference, they
sound better than usual consumer hi-fi gear from the 80s/90s.

The pro-sumer Focusrite Scarlett series is quite nice for modest
home recording, but the audio quality is not as good as of
usual consumer hi-fi gear from the 80s/90s, but still way better than
all those computer Realtek and Co. thingies.

You might not care too much about the jacks. The Scarlett jacks aren't
XLR, they are a combination of XLR and 6.3 mm jacks. You could use
an adapter cable, XLR to 3.5 mm or 6.3 mm to 3.5 mm. Some devices,
including the Scarlett series, even do not require special attention
related to balanced vs unbalanced IOs. However, when using some other
devices, converting between balanced and unbalanced requires some
attention, but it isn't rocket science either, so it doesn't harm to
care about it always, even if it's not necessary.

FWIW the Scarlett 1st and 2nd Gen might be completely different to the
current available 3rd Gen. I can only comment on the 2nd generation and
heard a lot about the 1st generation.

Regards,
Ralf
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Re: [Alsa-user] How do I set up a filter/plugin?

[Ralf Mardorf]

>>> I am looking at jack too, i'll eventually install pipewire-jack as I
>>> am already running on pipewire, I'll look up for jack-compatible
>>> plugins.

Hi,

you need a jack compatible host application to load audio plugins such
as LADSPA or LV2 plugins, jack by itself can't integrate plugins.
However, Ardour for example can be used with jack or plain ALSA and
it's also a host for LADSPA and LV2. Pipewire seems not to be without
issues and probably not ready for a reliable recording environment yet.

Regards,
Ralf


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Re: [Alsa-user] How to combine two alsa cards as one

[Ralf Mardorf]

Hi,

it either requires resampling by software, to sync the drifting clocks
of the two cards or sync by hardware. Optimal is word clock, but sync
by S/PDIF might work, too.

IOW it depends on the used sound cards.

Regards,
Ralf


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[Alsa-user] PS: s/pdif output not working on Asus Xonar SE card

[Ralf Mardorf]

On Thu, 29 Apr 2021 19:32:26 +0200, Ralf Mardorf wrote:
>Does your card provide coaxial and optical SPDIF?

IIUC after googling for your card, it has got an optical SPDIF
output only and no ADAT that could also use this port. IOW a possible Ω
coaxial issue can be ruled out, as well as a wrong routing. You also
don't have got a word clock port, so a termination resistor issue can be
ruled out, too. That being said, a cable's Ω or a termination resistor
issue anyway unlikely could be culprits related to your problem, I just
was curious.

alsamixer shows "coaxial", "optical" and "internal" for my RME's SPDIF.

I'm to lazy to test or google what "internal" is for. A wild guess, a
connection for a CD/DVD drive. So maybe another pitfall?!




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Re: [Alsa-user] s/pdif output not working on Asus Xonar SE card

[Ralf Mardorf]

On Thu, 29 Apr 2021 12:25:54 -0400, Mary Strimel wrote:
>I have a new Asus Xonar SE card that I have just installed. The analog
>speaker jack works, but S/PDIF out does not work. I have tried fiddling
>with every setting in alsamixer

Hi,

in my experiences with PCIe (RME), PCI (TerraTec) and USB (Focusrite)
SPDI without pulseaudio isn't an issue.

$ aplay -l | grep card
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712
multi]
card 3: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0]
card 3: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1]
card 3: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2]
card 3: HDMI [HDA Intel HDMI], device 9: HDMI 3 [HDMI 3]
card 3: HDMI [HDA Intel HDMI], device 10: HDMI 4 [HDMI 4]
card 4: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio]
$ arecord -l | grep card
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712
multi] card 4: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio]

You did not accidentally select a wrong card using alsamixer? The F6
key allows to chose a particular card.

FWIW my pro-sumer devices, the TerraTec and Scarlett work without
issues, but just 2 of the 8 ADAT channels of my professional RME card
do work. I never found a way to get all of them working with Linux.
I tested them installing FreeBSD and Windows and there they work/ed.

SPDIF works for all of them, IIRC whatever clock I select. Did you try
different clocks? What are "every settings" you tried?

FWIW I don't have got pulseaudio installed, I either use plain ALSA or
jack2. Did you test without pulseaudio?

Btw. while I didn't spend much time on it, HDMI audio works with Ubuntu
live media, but not with my Arch Linux install.

Does your card provide coaxial and optical SPDIF?

If I were you, I would at least for testing purpose get rid of
pulseaudio.

Sometimes Linux audio is fishy, sometimes a user's setup is fishy, but
in my experiences pulseaudio does ask for trouble and should be the
first thing get removed, to rule out pulseaudio related pitfalls, before
continuing trouble shooting, at least if nobody has got a better idea.

Regards,
Ralf

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Re: [Alsa-user] USB ALSA card number

[Ralf Mardorf]

Hi,

I'm using the below /etc/modprobe.d/alsa-base.conf file.

It contains 2 instances of snd_ice1712, but just one of those cards is
actually built-in, so no card can become hw:2.

You can also add snd_usb_audio as a placeholder.

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

[rocketmouse@archlinux ~]$ aplay -l # USB audio interface isn't attached
 List of PLAYBACK Hardware Devices 
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712 multi]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 10: HDMI 4 [HDMI 4]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
[rocketmouse@archlinux ~]$ aplay -l # USB audio interface is attached
 List of PLAYBACK Hardware Devices 
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712 multi]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 3: HDMI [HDA Intel HDMI], device 10: HDMI 4 [HDMI 4]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 4: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
[rocketmouse@archlinux ~]$

Regards,
Ralf


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Re: [Alsa-user] arecord command line options

[Ralf Mardorf]

Oops

Begin forwarded message:

Date: Tue, 29 Sep 2020 05:19:54 +0200
From: Ralf Mardorf 
To: alsa-user@lists.sourceforge.net
Subject: PS: [Alsa-user] arecord command line options

>It's for 24 bit, 192000 Hz sample rate. FWIW professional
>studio audio recordings are usually done at 48000 Hz.
>A sample rate > 48000 Hz doesn't gain better audio quality. To increase
>audio quality you need to replace the prosumer by a professional audio
>interface. Even if the E-MU 0202 USB should use good converters, it
>much likely suffers from a not that good analog component. Loosely
>speaking, recording frequencies that are inaudible for humans and/or
>not playable by analog audio equipment, doesn't increase the audio
>quality. It's common to cut frequencies that are inaudible for good
>reasons. When taking photos or when filming you don't increase the
>image quality by including ultraviolet light, actually it's better to
>get rid of ultraviolet light.
>Again loosely speaking, the higher the bit rate, the better, but very  
 this should read bit
depth ;)

>high bit rates are just an advantage for further processing, not for  
  ^

>just playing and listening.
>In short, with your device you much likely get the best result with 24
>bit, 48000 Hz. When using a sample rate > 48000 Hz you unlikely will
>benefit from a possible advantage, you more likely will suffer from
>a possible negative side effect


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Re: [Alsa-user] arecord command line options

[Ralf Mardorf]

On Mon, 28 Sep 2020 22:02:10 -0400, Alan Corey wrote:
>On 9/28/20, Zsolt Ero  wrote:
>> I have a few questions related to arecord, which I couldn't find in
>> the man pages nor anywhere on the internet.
>>
>> My use case is very simple, I'd like to record stereo 24/192 audio
>> into a WAV file from an E-MU 0202 USB interface using an old laptop
>> (Core 2 Duo).
>>
>> For this I thought of using the device which starts with "hw:", as it
>> seems to me that it provides the least processings over it. The
>> recording works without problems, however I'm confused about the
>> options.
>>
>> What is not clear to me:
>> --mmap - should I use it or not? I'd be writing from USB to hard
>> drive a wav file.
>> --period-time, buffer-time, period-size, buffer-size - I'm totally
>> confused about these. I'd like the highest possible recording
>> quality, latency doesn't matter to me.
>> --avail-min - what is a wakeup?
>> --disable-resample/channels/format/softvol - do I need this if I
>> selected the hw: device? I'd like to record without any kind of
>> processing.
>> --test-position/coef/nowait - do I need this for my use case?
>>
>I would say you're in danger of overthinking it, try the defaults
>first.
>
>Except when you say 24/192 I'm thinking 192 kbits/sec?  My usb sound
>cards can only do 48 max.  Yours may do better.  I was just looking
>into it for SDR purposes, the tuning range depends on the max sample
>rate.  SPDIF has always seemed like the way to go, or maybe firewire.
>But you need to study the E-MU 0202 specs to be sure it can do 192.

It's for 24 bit, 192000 Hz sample rate. FWIW professional
studio audio recordings are usually done at 48000 Hz.
A sample rate > 48000 Hz doesn't gain better audio quality. To increase
audio quality you need to replace the prosumer by a professional audio
interface. Even if the E-MU 0202 USB should use good converters, it
much likely suffers from a not that good analog component. Loosely
speaking, recording frequencies that are inaudible for humans and/or not
playable by analog audio equipment, doesn't increase the audio quality.
It's common to cut frequencies that are inaudible for good reasons.
When taking photos or when filming you don't increase the image quality
by including ultraviolet light, actually it's better to get rid of
ultraviolet light.
Again loosely speaking, the higher the bit rate, the better, but very
high bit rates are just an advantage for further processing, not for
just playing and listening.
In short, with your device you much likely get the best result with 24
bit, 48000 Hz. When using a sample rate > 48000 Hz you unlikely will
benefit from a possible advantage, you more likely will suffer from
a possible negative side effect

-- 
“Awards are merely the badges of mediocrity.”

― Charles Ives


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Re: [Alsa-user] This appears to be the problem I am getting with this list - and now other lists - DMARC / DKIM / SPF

[Ralf Mardorf]

On Tue, 04 Aug 2020 15:52:05 +1000, Philip Rhoades wrote:
>   Yahoo breaks every mailing list in the world

Yes, but the breakage isn't fatal, at least it isn't fatal as long as
it's a well maintained 'mailman 2' mailing list.

To post to the 'FreeBSD questions' list I used a rocketmail = yahoo
address.
I had no 'mailman 2' bounce score, nothing odd happened, excepted that
my posts had the header 'From: Ralf Mardorf via freebsd-questions'
showing the mailing list address. I migrated to a 'riseup.net' address
for this list and now the header is 'From: Ralf Mardorf' showing my
address.

On the 'claws-mail list', also a 'mailman 2' list, I'm using an original
yahoo address, on this list it's the same, no bounces, just my posts
have a 'From: ... via ..." header, with the mailing list address instead
of mine. Some time ago mails sent to this list didn't came through to
myself (while they came through the list), but even this issue doesn't
exist anymore.


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Re: [Alsa-user] Class-2 compliant USB device (RODECaster Pro) unable to retrieve number of sample rates

[Ralf Mardorf]

On Fri, 26 Jun 2020 14:02:37 +0100, James Conroy-Finn wrote:
>pulseaudio

Hi,

I can't comment on the RØDECaste Pro. However, at least for
troubleshooting consider to disable or remove pulseaudio.
Test with plain ALSA and/or the jack soundserver.

All my Linux machines are using plain ALSA or jack on demand.
I install empty dummy pulseaudio packages, to fulfil potential
dependencies against the pulseaudio package. For my needs pulseaudio
makes no sense at all.

I don't know how good or bad pulseaudio nowadays is, but if an issue
happens and it is involved, a troubleshooting step should be to get
rid of pulseaudio, to see if the issue remains.

Just that pulseaudio output mentions an ALSA issue, not necessarily
means that pulseaudio isn't the actual culprit.

Regards,
Ralf


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Re: [Alsa-user] RME HDSPe AIO and AO4S -192 AIO

[Ralf Mardorf]

On Tue, 2020-05-19 at 11:03 +0200, Maurizio M wrote: 
> Yes I confirm that the RME cards work without problems on Linux, only I 
> tried to understand how to enable the other 2 outputs but if you are not 
> aware of the situation ok I understand, don't worry.

Hi Maurizio,

do not count your chickens before they are hatched ;). Unless not all
IOs are working, it obviously doesn't work on your Linux install without
issues, too.

The question is, if it is possible to use all IOs on Linux. I get the
two analog IOs + 2 ADAT IOs working. 6 ADAT IOs don't work at all. It's
been a while that I tested the IOs. IIRC I didn't test AES/EBU. IIRC
SPDIF failed to work, too. Phones do work well.

Please note, I'm using the card on Linux only. It was tested around 9
years ago on FreeBSD and Windows, to ensure that the card isn't broken
and to ensure that using hdspmixer doesn't fail due to user errors.

It worked with FreeBSD, IIRC without a mixer and it worked with Windows
using TotalMix. With my Linux installs on different machines it only
worked and still works partially.

Good luck!
Ralf




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Re: [Alsa-user] Out AO4S 192 AIO

[Ralf Mardorf]

On Mon, 2020-05-18 at 19:54 +0200, Maurizio M wrote:
> Hi Ralph, yes I was missing the third part in fact you are absolutely 
> right. I did step 3 and already saved 2 presets thank you very much. I 
> wanted to ask you for more information then if possible, I have 2 more 
> outputs from the expansion card that I have just installed, the 
> AO4S-192-AIO, which however I can't activate. I can't get anything out. 
> The outputs of the card are 4 monophinic that are used to make 2 stereo 
> outputs and while the output 3 and the output 4 of the card, I am 
> currently using them successfully and I managed to direct them to 
> channel 19 and channel 20 of the mixer, the outputs 1 and 2 of the card 
> which should go to channel 17 and 18 of the mixer, I can't address them. 
> I put 1-2 AN then analog outputs in the drop-down menu, I also put 3-4, 
> 5-6 I tried them all but I couldn't. Can you tell me where I'm wrong? Thanks

Sorry, I can't help with this. To ensure that you understand how
hdspmixer works, I recommend to temporarily install Windows and
TotalMix. Maybe a dated version of TotalMix. When I installed Windows
and TotalMix around 9 years ago, TotalMix was already more advanced than
hdspmixer, but there was no real difference related to the GUI's routing
principle. The latest version of TotalMix might be completely different.
I don't know.

However, I know how to use an aged version of TotalMix and
seemingly/likely how to use hdspmixer, too, but never got all available
IOs working on Linux. For some reason the card also doesn't work, when
trying to use Ardour with plain ALSA instead of jackd. I guess the
driver is fishy, or some ALSA config was and still is missing on my
machines. Some people claimed, the driver does work for the HDSPe AIO
without issues, too.



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Re: [Alsa-user] RME HDSPe AIO + RME AO4S 192 AIO

[Ralf Mardorf]

On Mon, 18 May 2020 18:30:08 +0200, Ralf Mardorf wrote:
>Take a look at the attached pic. You only made step 1, but you need to
>do step 2 and step 3, too.

It seems you made step 1 and step 2, but you are missing step 3, "Make
current file default". If you already made step 2, you don't need to do
the settings again, just open the file and then choose "Make
current file default".


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Re: [Alsa-user] RME HDSPe AIO + RME AO4S 192 AIO

[Ralf Mardorf]

Hi Maurizio,

thank you for reporting back.

Take a look at the attached pic. You only made step 1, but you need to
do step 2 and step 3, too.

CCing, since the attached png might not make it through the mailing
list.

Regards,
Ralf



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Re: [Alsa-user] RME Exspansion

[Ralf Mardorf]

Hi Maurizio,

I'm not an ALSA developer. Likely most subscribers of this users
mailing list aren't ALSA developers. Perhaps most of us are users or
developers of other software.

Regards,
Ralf


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Re: [Alsa-user] HDSPConf in the RME HDSPe AIO

[Ralf Mardorf]

On Mon, 11 May 2020 20:24:40 +0200, Maurizio M wrote:
>Hi Ralf, thanks for your reply. If you tell me it can't be used then
>ok, it means that everything is normal and not a bug in the Power
>version. I use AlsaMixer and HDSPMixer even though I noticed that if I
>save the AlsaMixer settings and then open HDSPMixer, the AlsaMixer
>settings are lost and it returns as it was previously and I have to
>adjust everything again. For the rest the card works great, its
>quality is really high, full, detailed, dynamic and clean sound,
>really exceptional. Now I also bought the output expansion called AO4S
>/ 192, to expand the outputs of the AIO. It allows me to have 4
>additional stereo outputs and has its analog section and its DAC,
>independent and dedicated. Thanks for your informations.
>Maurizio
>Italy

Hi Maurizio,

next time please reply to the mailing list, instead of sending a reply
to my email address.

Yes, the audio quality of this and probably all RME cards is way better
than of all those prosumer audio devices. You can save hdspmixer
settings by the hdspmixer menus and you can make one of the setting
files the default, this way you only need to start hdspmixer. Running
alsamixer is usually unneeded, it provides a few additional settings.

Have you tested all 8 ADAT IOs? If so, do they work?
As already mentioned by my previous reply, here only 2 of the 8 ADAT
channels do their job.

I never tested an expansion. It works?

Regards,
Ralf


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Re: [Alsa-user] HDSPConf in the RME HDSPe AIO

[Ralf Mardorf]

PS: Maybe you don't see the menu bar of hdspmixer. When using a top
panel, it could happen, that hdspmixer opens with the menu bar hidden
under the top panel. I'm using a script to automagically move the
window. Almost all, if not all window managers allow to move the window
by holding the Alt-key and left mouse button with the mouse cursor
anywhere inside the window.


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Re: [Alsa-user] HDSPConf in the RME HDSPe AIO

[Ralf Mardorf]

On Sun, 2020-05-10 at 20:01 +0200, Maurizio M wrote:
> Hello excuse me, I am writing to you because I have a problem with your 
> application dedicated to RME HDSPe AIO sound cards. When I install Alsa 
> Tools Gui, both the HDSPE Mixer and the hammer icon called HDSPconf are 
> installed on the computer. Just this executable with the hammer called 
> HDSPConf, it doesn't start, there must be some bugs because it doesn't 
> work. Attention I use IBM Power architecture so I don't know if the 
> problem is specific to my architecture while on X86 it is not there. Do 
> you have news about it? Thanks

Hi,

I bought my HDSPe AIO in May 2011 for exclusive usage with Linux.

[rocketmouse@archlinux ~]$ aplay -l | grep card | grep -v HDMI
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712 multi]
card 4: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio]

On Linux I never got it working with anything, but the 2 analog channels
and just 2 of 8 ADAT channels. For testing purpose I installed Windows
and FreeBSD. The card is not broken, I know how to use TotalMix, I got
it running with all features and channels available, just not on Linux,
on more than just one machine. It suffers a lot from latency and xrun
issues, but is still usable for pro-audio usage, if not more than 4
channels are required. When using Ardour it does not work with plain
ALSA, you must use jackd. YMMV!

HDSPConf can't be used. You need to use alsamixer and hdspmixer.

Regards,
Ralf



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Re: [Alsa-user] Fulla Schiit asound file

[Ralf Mardorf]

On Sat, 2020-01-25 at 13:43 +, Dmitri Seletski wrote:
> > Oops:
> > 
> > > However, accessing the software mixer requires a driver.
> >^^
> >actually it's the device's hardware mixer
> >that is doing internal routing, providing
> >digital effects or any other special feature.
> >You could always use a software mixer, such as
> >the one available by Ardour. However, such a
> >software mixer doesn't allow to benefit from
> >the device's hardware (mixer) features.
> 
> Having a software mixer beats not having it at all.

Depending on what you want to do, there are different kinds of Linux
internal mixers available.

I can use Ardour's mixer for my needs, regarding output volumes,
regarding input volumes I either need to use the audio devices volume
control potentiometers and PAD switches or a "real" mixing console. Not
all audio devices provide input potentiometers or PAD switches, let
alone a way to select between +4dBu or -10dBV. 

For other usages than professional audio recording, even a sound server
such as pulseaudio probably provides volume controls. However, output
usually isn't an issue at all, but by using a plain Linux software
mixer, that can't access the input hardware controls, you can't
compensate what going wrong in the first place. Assuming the hardware
input is distorted, you can't fix it by a software mixer's input volume.

For people using multi-channel devices for audio recording, monitor
routing could become an issue. This probably isn't an issue for your
seemingly stereo device.

IOW output volume shouldn't be an issue for you, there should be one or
the other way that fits to your needs.

Input volume shouldn't be an issue either, but it could be an issue.
Assuming the input should be in a "sane" range in the first place. then
you could control the input volume by an Linux internal software mixer,
but if the input volume should be that loud, that it does cause
unpleasant distortion or assuming it should be way lower than the
circuits noise level, then there's no way to fix it by a Linux internal
software mixer.



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Re: [Alsa-user] Fulla Schiit asound file

[Ralf Mardorf]

On Sat, 2020-01-25 at 10:49 +0100, Ralf Mardorf wrote:
> On Fri, 2020-01-24 at 23:30 +, Dmitri Seletski wrote:
> > https://www.schiit.com/products/fulla-1
> 
> Since it's an USB device, it's most likely class compliant...
> 
> > Just connected  it, alsa works by default, but no software mixer.
> 
> ...this means that it works for Linux and Apple without the need of a
> proprietary driver. Nobody needs to do reverse engineering or to beg to
> get detailed information from the vendor, if the vendor doesn't provide
> a driver.

Oops:

> However, accessing the software mixer requires a driver.
 ^^
 actually it's the device's hardware mixer
 that is doing internal routing, providing
 digital effects or any other special feature.
 You could always use a software mixer, such as
 the one available by Ardour. However, such a   
 software mixer doesn't allow to benefit from   
 the device's hardware (mixer) features.
> 
> Btw. among other prosumer and pro audio devices I own a prosumer
> Focusrite Scarlett series class compliant USB audio device (a 18i20) of
> the 2nd generation. For at least some, if not all of the 1st generation
> Scarlett audio devices, Linux provides access to the hardware mixer, but
> not for the 2nd generation. By now already a 3rd generation exists.
> 
> IOW just because a driver for a particular revision of hardware does
> exist, it doesn't mean that all revisions are supported.



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Re: [Alsa-user] Fulla Schiit asound file

[Ralf Mardorf]

On Fri, 2020-01-24 at 23:30 +, Dmitri Seletski wrote:
> https://www.schiit.com/products/fulla-1

Since it's an USB device, it's most likely class compliant...

> Just connected  it, alsa works by default, but no software mixer.

...this means that it works for Linux and Apple without the need of a
proprietary driver. Nobody needs to do reverse engineering or to beg to
get detailed information from the vendor, if the vendor doesn't provide
a driver.

However, accessing the software mixer requires a driver.

Btw. among other prosumer and pro audio devices I own a prosumer
Focusrite Scarlett series class compliant USB audio device (a 18i20) of
the 2nd generation. For at least some, if not all of the 1st generation
Scarlett audio devices, Linux provides access to the hardware mixer, but
not for the 2nd generation. By now already a 3rd generation exists.

IOW just because a driver for a particular revision of hardware does
exist, it doesn't mean that all revisions are supported.



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Re: [Alsa-user] How to redirect audio to a stream (e.g. as input for mpd)

[Ralf Mardorf]

On Sat, 18 Jan 2020 21:53:23 +0100, Jürgen Gluch wrote:
>I want to redirect audio within alsa to stream, so that another
>software can use it other output.
>
>Let me explain my setup. My multiroom audio for 4 rooms runs on my
>sever and the audio is hard wired to 4 stereo amplifiers. The server
>(on Xubuntu 19.04)  has two soundcards (HDMI as hw0,0 and a 7.1
>soundcard as hw1,0). So far I only used the 7.1 for the multiroom
>audio. The 7.1 soundcard is remapped to dmixer and four 2.0 outputs,
>that are used by four independent mpd's (music player deamon). So
>every family member can run its own music player client on the mobile
>phone or use the wall mounted tablet to choose music (mp3 file
>database or web radio streams) and the rooms were the output should be
>active. This runs stable and nicely for years now. Since some time we
>enjoy spotify and I want to stream the spotify audio also to the
>multiroom speakers. Therefore a spotify client (the original) is
>running on the server and its output goes to "pcm.!default", which I
>set to hw0,0. The different spotify apps play also nice together, but
>I cant find a solution for the connection from spotify output to mpd
>input.
>
>Is there a possibility to stream the "pcm.!default" to file and make it
>available to mpd as a web stream (e.g. localhost, any port, as raw,
>wave or ogg)? I tried to use "type file" but was not successful.

An off-topic side note:

IMO this is an odd approach. Hard wiring 4 amps in 4 different rooms
with obviously consumer grade audio devices of one machine, gains
absolutely nothing over 4 completely independent audio setups, each for
one room. Apart from the issue you are experiencing now, that probably
is solvable, the main issue is, that if the one machine gets damaged,
audio for 4 rooms doesn't work anymore. IMO the better approach would
be to allow access to a server data base of audio files and links to
web radio stations, but to have separated audio systems for each room.

For consumer grade audio quality smart phones and the wall mounted
tablet PC, accessing the server to get audio files and web radio links,
but connecting directly via e.g. bluetooth to an audio system in a
particular room, is way better, since less failure-prone.

It would be different, if one machine has got an expensive professional
garde audio device and 4 rooms should benefit from the highly graded
audio quality.

Your way IMO is making something very simple way to complicated. I
dislike the smart home approach, that usually isn't anything but
smart, but just asking for trouble.


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Re: [Alsa-user] Help with getting Traktor Kontroller Z1 to work

[Ralf Mardorf]

Hi,

actually the pulseaudio sound server does use the ALSA driver to get
access to the audio interface. If pulseaudio has already grabbed the
audio interface via the ALSA driver, you first need to disable
pulseaudio, before other software can access the hardware
by the ALSA driver.

I can't help you with disabling pulseaudio on demand or how to use
other workarounds, since I either use ALSA directly or the jack sound
server. On my machine pulseaudio isn't installed at all.

The software you want to use seemingly supports jack:

[rocketmouse@archlinux ~]$ pacman -Si mixxx | grep s\ On
Depends On  : chromaprint  faad2  gperftools  glu  libid3tag  libmad  
libmp4v2  libshout  lilv  opusfile  portaudio  portmidi  protobuf  qt5-script  
qt5-svg  qt5-x11extras  rubberband  taglib  upower  wavpack
[rocketmouse@archlinux ~]$ pacman -Si portaudio | grep s\ On
Depends On  : gcc-libs  jack

Regards,
Ralf

-- 
pacman -Q linux{,-rt{-cornflower,-pussytoes,,-securityink}}|cut -d\  -f2
5.2.4.arch1-1
5.2_rt1-0
5.0.21_rt16-1
5.0.19_rt11-1
4.19.59_rt24-0


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[Alsa-user] MIDI in for VirtualBox iOS guest

[Ralf Mardorf]

Hi,

the Synth One open-source project
[ https://github.com/AudioKit/AudioKitSynthOne ],
unfortunately not available for Linux, runs fine installed to an
iOS 12.2 VirtualBox guest [ https://i.imgur.com/4rHkCs4.jpg ]. Audio out
works and is available for usage by the Linux host.

[weremouse@moonstudio ~]$ lsb_release -a
LSB Version:core-9.20160110ubuntu0.2-amd64:core-9.20160110ubuntu0.2-
noarch:security-9.20160110ubuntu0.2-amd64:security-9.20160110ubuntu0.2-
noarch
Distributor ID: Ubuntu
Description:Ubuntu 16.04.6 LTS
Release:16.04
Codename:   xenial

MIDI in for the Linux host works without issues, whatever MIDI hardware
input I chose.

[weremouse@moonstudio ~]$ amidi -l
Dir DeviceName
IO  hw:0,0HDSPMx579bcc MIDI 1
IO  hw:1,0TerraTec EWX24/96 MIDI
IO  hw:4,0,0  Scarlett 18i20 USB MIDI 1

However, MIDI in isn't available for the VirtualBox guest.

Any ideas?

As a workaround I record MIDI tracks with Ardour running on the Linux
host, export the MIDI files and import them by Auria Pro or Cubasis
running on the iOS guest. IOW MIDI within the VirtualBox iOS guest works
without issues.

Regards,
Ralf









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Re: [Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Ralf Mardorf]

On Sun, 2018-12-23 at 11:57 -0600, Paul wrote:
> On 12/23/18 5:38 AM, Ralf Mardorf wrote:
> > On Sun, 23 Dec 2018 09:36:13 +0100, Ralf Mardorf wrote:
> > > HDMI
> > Oops, so the analog domain isn't on board :D. However, the culprit for
> > odd audio quality remains the analog domain.
> > 
> > 
> > ___
> > Alsa-user mailing list
> > Alsa-user@lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/alsa-user
> 
> Actually I can hear pretty clearly the difference between 16/44 CD 
> quality and 24/96 recordings, not to mention SACD music - this one blows 
> anything else out of water.
> 
> Obviously one needs decent hardware and speakers, and not every music 
> can reveal the difference. I bet if one mostly listens to rap in his 
> car, then it's not possible to even distinguish between low rate MP3 and 
> CD.
> 
> BTW, I think HD audio already supported by AMD Vega 11 GPU, I was able 
> to play 24/192 source today through HDMI and it wasn't down-converted, 
> at least my receiver showed 192Khz sampling. So please disregard my 
> question.
> 
> Thanks.

I never mentioned CD quality, I mentioned 48 KHz, not 44,1KHz! Let alone
that my main point is the analog domain.

Regarding computer related audio equipment, for example, the headphone
amp's audio quality of card 0 is way beyond of card 4. Sample rate and
bit depth of the digital domain are secondary, if the analog domain
isn't professional.

[rocketmouse@archlinux ~]$ aplay -l | grep -e card\ 0 -e card\ 4
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
card 4: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio]

Even DAT longplay would sound better played via the RME analog outputs,
than any better digital quality recording played by the Scarlett's
analog outputs. _But_ AFAIK even a pro-sumer device such as the Scarlett
beats any on-board audio device, in regards of converters as well as the
analog domain.

Sample rate and bit depth mean nothing at all, if the converters and the
analog domain are the weak parts of the audio chain.



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Re: [Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Ralf Mardorf]

On Sun, 23 Dec 2018 18:45:56 +0100, Ralf Mardorf wrote:
>On Sun, 23 Dec 2018 08:03:42 -0800, chris hermansen wrote:
>>Actually it appears that it is possible to distinguish high resolution
>>content  
>
>Some people that have done similar tests, claim that artificial
>strawberry flavour does taste more natural, than a strawberry.[snip] Do
>the strawberry yogurt test first, before you ask people to listen to
>music.

PS: I visited an elementary school's Christmas party, almost all
parents watched their acting children on the screen of a smartphone.
They moste likely did not watch their children directly, because it's
more natural to watch art on a smartphone screen nowadays, than taking a
look with the eyes directly to the stage. IOW a recording, is a
recording, is a recording, is a recording! If you reach the technical
and "spiritual" limit, you reach the limit. Most people nowadays are
just used to spiritual limits and their guess is that an overdose of
redundant technology somehow could rectify barbarism.


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Re: [Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Ralf Mardorf]

On Sun, 23 Dec 2018 08:03:42 -0800, chris hermansen wrote:
>Actually it appears that it is possible to distinguish high resolution
>content

Some people that have done similar tests, claim that artificial
strawberry flavour does taste more natural, than a strawberry. It was
my job to build professional studio gear, among others jobs I was
engineer for a famous German microphone company. I seriously wonder
what microphone was used to do any analog or digital recording in the
first place, that provides a signal, that could benefit from something
higher for playback than 48 KHz/16 bit. I'm not interested in reading
any paper, I like to get the recordings that prove to make a difference!
I doubt that some wobbling virtual synth beats the audio quality of
professional recorded music. Do the strawberry yogurt test first,
before you ask people to listen to music.


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Re: [Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Ralf Mardorf]

On Sun, 23 Dec 2018 09:36:13 +0100, Ralf Mardorf wrote:
>HDMI

Oops, so the analog domain isn't on board :D. However, the culprit for
odd audio quality remains the analog domain.


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Re: [Alsa-user] AMD RX Vega 11 HDMI - audio limited to 16bit/48KHz

[Ralf Mardorf]

On Sat, 22 Dec 2018 20:16:28 -0600, Paul via Alsa-user wrote:
>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?

Off-topic: For playback (not for production) of audio signals aimed for
the human ear and human brain, 16 bit 48 KHz is all you need. If you
aren't satisfied by the audio quality you get for playback at 16 bit
48 KHz, it's usually the analog domain that needs improvement first,
but indeed the digital to analog converters are available in different
qualities, too, so you also might need to replace them. The most weak
point in the analog chain usually are integrated (onboard) audio
devices. However, even if you own pro-sumer or professional audio
devices (PCIe/USB etc.) with very good digital to analog converters, the
analog domain of those devices still could make a big difference. Don't
expect that you gain anything regarding audio quality audible for
humans, as soon as 24/32 bit and 96/192KHz for playback are supported.
Btw. for audio productions it's better to have a higher bit resolution,
at least 24 bit and better 32 bit floating-point, but even than you
don't need a sample rate higher than 48 KHz.


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Re: [Alsa-user] Bower's & Wilkins PX headphone - snd-usb-audio error

[Ralf Mardorf]

On Thu, 30 Aug 2018 21:36:42 +0200, Philipp Ludwig wrote:
>On 08/29/2018 07:20 AM, Takashi Iwai wrote:
>Searching on the net for viable test files was without success, but
>if you might have some files handy, I'm happy to test everything.
>At least 48kHz seem to work.
>>   aplay -v -Dplughw:1 some-44100hz-samples.wav

To get 44.1 KHz wavs e.g. use K3b to rip an audio CD. Use e.g. ffmpeg to
convert a wav to 48 KHz.

ffmpeg -i input.wav -ar 48000 output.wav

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Re: [Alsa-user] Bower's & Wilkins PX headphone - snd-usb-audio error

[Ralf Mardorf]

On Fri, 31 Aug 2018 03:48:55 +0200, Ralf Mardorf wrote:
>On Thu, 30 Aug 2018 21:36:42 +0200, Philipp Ludwig wrote:
>>On 08/29/2018 07:20 AM, Takashi Iwai wrote:
>>Searching on the net for viable test files was without success, but
>>if you might have some files handy, I'm happy to test everything.
>>At least 48kHz seem to work.  
>>>   aplay -v -Dplughw:1 some-44100hz-samples.wav  
>
>To get 44.1 KHz wavs e.g. use K3b to rip an audio CD. Use e.g. ffmpeg
>to convert a wav to 48 KHz.
>
>ffmpeg -i input.wav -ar 48000 output.wav

Oops:

On Thu, 30 Aug 2018 21:36:42 +0200, Philipp Ludwig wrote:
>Regarding playback, it seems that I got only 48kHz samples in
>/usr/share/sounds/alsa

So it's even more simple:

ffmpeg -i /usr/share/sounds/alsa/Noise.wav -ar 44100 /tmp/Noise_44100.wav

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

[Ralf Mardorf]

On Wed, 4 Jul 2018 22:37:36 +0200, Andreas Böhler wrote:
>The device does not work when the sample rate is changed 
>using Ardour.

Are you using Ardour with plain ALSA or with jack using the ALSA
backend? Some audio device only work when using jack.

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

[Ralf Mardorf]

On Wed, 4 Jul 2018 22:53:33 +0200, Ralf Mardorf wrote:
>On Wed, 4 Jul 2018 22:37:36 +0200, Andreas Böhler wrote:  
>>The device does not work when the sample rate is changed 
>>using Ardour.
>
>Are you using Ardour with plain ALSA or with jack using the ALSA
>backend? Some audio device only work when using jack.  
 ^ with Ardour

When using other software the same audio device could be used with
plain ALSA. This is a known issue.

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

[Ralf Mardorf]

On Sun, 22 Apr 2018 14:07:04 +0200, Marc Haber wrote:
>> 2. If you wish to continue using a DAT recorder in combination with
>> your computer, then consider to spend more than 23,85 € for your
>> computer's audio device.  
>
>I'm open for suggestions.

My apologies, it's a bad idea, now I notice that all cheaper
semi-professional USB devices around 300,- € only provide Coaxial
S/PDIF input and output, the only exception seems to be
Behringer, providing at least one device for around 200,- € with optical
ADAT IOs. The user manual mentions that this Behringer device's ADAT IOs
also could be used as optical S/PDIF IOs (not all devices with ADAT IOs
allow usage of those IOs as S/PDIF IOs). While I suspect (I don't know)
that the Behringer device works as good as devices from other vendors
with Linux, I've got concerns regarding durability, so I advice against
Behringer. There are second hand semi-professional PCI cards for less
than 50,-€ available, providing optical S/PDIF, but this doesn't help
you, if you need USB.

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

[Ralf Mardorf]

On Sun, 8 Apr 2018 16:41:08 +0200, Marc Haber wrote:
>Ping? noone?

All my DAT recorder drives are broken, the only reason to keep them is
to have DACs and ADCs I could use via S/PDIF, in addition to the analog
I/Os and ADAT I/Os of my audio devices, just in case I ever should need
additional I/Os, which actually never happened. You already could get a
semi-professional OOTB Linux compatible, especially class compliant
audio device, for half of the price even a second hand DAT recorder
does cost. I could use my broken DAT recorders converters via S/PDIF
with my audio devices.

1. One of my broken DAT recorders is a Sony consumer DAT. Better
discontinue to use a consumer DAT recorder, soon or later they all
suffer from drive issues.

2. If you wish to continue using a DAT recorder in combination with
your computer, then consider to spend more than 23,85 € for your
computer's audio device.

It isn't worth the hassle to save on the wrong things.

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Re: [Alsa-user] ALSA: Unknown device state '3'

[Ralf Mardorf]

PPS:

On Sun, 11 Mar 2018 09:46:21 +0100, Дмитрий wrote:
>RPi box

Oops, my apologies, I have not the slightest idea if Meltdown and
Spectre mitigation are affecting ARM cores at all. The Internet clams
that "Raspberry Pi isn’t susceptible to these vulnerabilities", but
perhaps the PTI related patch sets anyway have impact, dunno?!

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Re: [Alsa-user] ALSA: Unknown device state '3'

[Ralf Mardorf]

PS:
>On Sun, 11 Mar 2018 09:46:21 +0100, Дмитрий wrote:
>>I use a piece software called ecasound to push a stream from digital
>>input to the output. I've been using it on RPi box with Raspbian since
>>"Jessie" release. After updating it to "Stretch" I started to get such
>>error messages when I run ecasound like this "/usr/bin/ecasound
>>-B:rtlowlatency -b:256 -f:32,2,48000 -i:alsahw,1,0 -o:alsa,softvol":
>>  
>>> ALSA: Unknown device state '3'

"ALSA: Unknown device state" seems to be unrelated to a "performance
issue", but you never know ;).

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Re: [Alsa-user] ALSA: Unknown device state '3'

[Ralf Mardorf]

On Sun, 11 Mar 2018 09:46:21 +0100, Дмитрий wrote:
>I use a piece software called ecasound to push a stream from digital
>input to the output. I've been using it on RPi box with Raspbian since
>"Jessie" release. After updating it to "Stretch" I started to get such
>error messages when I run ecasound like this "/usr/bin/ecasound
>-B:rtlowlatency -b:256 -f:32,2,48000 -i:alsahw,1,0 -o:alsa,softvol":
>
>> ALSA: Unknown device state '3'
>> ALSA: playback xrun handling failed!  
>
>It happens when an input stream is stopped. I need to restart ecasound
>to get it working. I guess something is changed in ALSA in Debain
>"Stretch", because in "Jessie" I didn't have such issue. Is there a
>way to prevent such behavior?

Hi, just a shot into the dark,

maybe it's not ALSA, but Meltdown and Spectre mitigation. I don't know
if firmware/microcode is updated or KAISER or KPTI patch sets are
applied to the kernel used by your distro's release, neither do I know
if they affect DSP load. However, Meltdown and Spectre mitigation could
have impact on performance.

[rocketmouse@archlinux ~]$ uname -rm
4.14.24-rt19-1-rt x86_64
[rocketmouse@archlinux ~]$ cat /sys/devices/system/cpu/vulnerabilities/*
Mitigation: PTI
Mitigation: __user pointer sanitization
Mitigation: Full generic retpoline

Try booting with  nokaiser  or  nopti .

After doing so, the output should look like this:

[rocketmouse@archlinux ~]$ cat /sys/devices/system/cpu/vulnerabilities/*
Vulnerable
Mitigation: __user pointer sanitization
Mitigation: Full generic retpoline

Regards,
Ralf

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

[Ralf Mardorf]

On Sun, 4 Feb 2018 13:22:43 +0100, Clemens Ladisch via Alsa-user wrote:
>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/.

FWIW "some .conf file" means you could name it as you want, a common
used name is "alsa-base.conf".

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

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

[Ralf Mardorf]

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.

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

[Ralf Mardorf]

On Sun, 22 Oct 2017 15:09:15 -0400, John Z. wrote:
>Well, truth to be told, it *is* possible to build firefox with alsa
>support by flicking on a config switch. Its just unfortunate my distro
>(arch) dropped the alsa support with newest updates due to conflicts
>with some other features they'd like to support.

IIRC "ac_add_options --enable-alsa" was dropped for Arch's Firefox 52.0,
but IIRC then the repositories provided the builds with enable-alsa
again, but IIRC with Firefox 54 upstream dropped alsa completely.

There's a thread from March that seems to confirm this...
https://lists.archlinux.org/pipermail/arch-general/2017-March/043314.html
...yes, this time I build Firefox.

Palemoon is installed on my machine, too.

Just a few of the installed browsers:

[rocketmouse@archlinux ~]$ pacman -Q firefox apulse palemoon-bin icecat-bin 
qupzilla google-chrome 
firefox 56.0.1-1
apulse 0.1.10-1
palemoon-bin 27.5.1-1
icecat-bin 52.3.0-3
qupzilla 2.2.0-2
google-chrome 62.0.3202.62-1

In the past my favourite was qupzilla, but it became unstable and since
I don't have flash installed and HTML doesn't stream videos very good,
I tend to use chrome for this. Very often, when videos are interrupted
without flash, they run without interruption, when using google-chrome.

Btw. apulse firefox works without issues.

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

[Ralf Mardorf]

On Sun, 22 Oct 2017 12:18:43 -0400, John Z. wrote:
>Aside this switching issue, and few stubborn applications (hello there,
>firefox), I really fail to see the reason to use PA on top of alsa.

Off-topic:

For Firefox consider to use https://github.com/i-rinat/apulse .

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Re: [Alsa-user] Slot assignment not working

[Ralf Mardorf]

On Sun, 8 Oct 2017 11:44:06 -0700, Paul D. DeRocco wrote:
>> From: Clemens Ladisch via Alsa-user 
>>   
>> > 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?  
>
>The subsidiary ones are, but snd isn't. Can this parameter be supplied
>on the kernel command line as snd.slots=...? Or would it be better to
>recompile the kernel with snd as a module? If so, what's the CONFIG_
>symbol?

Probably it's

[rocketmouse@archlinux ~]$ zgrep -i snd /proc/config.gz | head -1
CONFIG_SND=m

https://cateee.net/lkddb/web-lkddb/SND.html

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[Alsa-user] I know...

[Ralf Mardorf]

...subscribers shouldn't send test mails to mailing lists. But since
sourceforge is buggy for other users, too, I risk to send this test
mail, since seemingly the issue described at
https://lists.linuxaudio.org/pipermail/linux-audio-user/2017-September/108596.html
disappeared for my new account. Does my old account work again?

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Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . . UPDATE

[Ralf Mardorf]

On Mon, 04 Sep 2017 15:30:46 +1000, Philip Rhoades wrote:
>apulse google-chrome

Hi,

google-chrome still has got alsa support. I run google-chrome without
apulse and firefox with apulse, because firefox dropped alsa support.
Regarding screen recording I don't have experiences, I just heard that
some users migrated from recordmydesctop to vokoscreen.

FWIW on my Linux pulseaudio isn't installed, but I don't use dmix or
anything else, since either just one app is running that requires an
audio interface or for music productions I launch jackd.

Regards,
Ralf

PS: OT: I noticed that HTML5 doesn't work without interruptions when
watching live streams, that's why I'm using google-chrome.

PPS: I can't reply to alsa-user anymore,
see
https://lists.linuxaudio.org/pipermail/linux-audio-user/2017-September/108596.html
 .


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Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . . UPDATE

[Ralf Mardorf]

On Mon, 04 Sep 2017 00:50:24 +1000, Philip Rhoades wrote:
>I now want to use "recordmydesktop" which is working fine with the mic 
>but not recording sound from videos that are playing eg from YouTube
>or local mpv etc - hopefully an alsa guru will have a solution for me?

Sorry, I can't help, but perhaps one of the following software is
helpful:

https://github.com/i-rinat/apulse
https://github.com/vkohaupt/vokoscreen

I don't know if

  vokoscreen

or

  apulse vokoscreen

does work, at least

  apulse firefox

works.

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Re: [Alsa-user] Native Instruments Komplete Audio 6 SPDIF-out issue

[Ralf Mardorf]

On Wed, 9 Aug 2017 20:42:30 +0200, Philip wrote:
>ALSA in Linux Mint 18.1 does not set up the SPDIF by default. Only an
>analog output with the respective profiles are shown in the end in the
>Pulse Audio controls.

Hi,

perhaps not an ALSA issue, but a pulseaudio issue?

Seemingly the device should work OOTB:

https://linuxmusicians.com/viewtopic.php?t=12161#p50901

Consider to subscribe to this forum and ask there.

FWIW I'm not using the same device as you, but among PCIe and PCI audio
interfaces, I'm using one USB class compliant device, too. It's a
Focusrite Scarlett 18i20 2nd gen and S/PDIF works OOTB.

I'm using ALSA and jackd 2 (not jackdbus) only. For packages with a hard
dependency on pulseaudio I build an empty dummy package
"pulseaudio". This works for at least Ubuntu and Arch Linux without
issues.

On Arch Linux I'm using https://github.com/i-rinat/apulse for Firefox,
on Ubuntu I don't have Firefox installed and nothing else I'm using
requires pulseaudio.

AFAIK you could disable pulseaudio temporarily, so for testing purpose
consider to at least disable pulseaudio, remove your asound.conf entry
and try again using plain alsa and/or jackd 1 or 2.

Btw. neither my consumer, pro-sumer, nor my professional audio devices
cause static noise when using balanced as well as unbalanced analog IOs.
For some unknown reason the headphone output of my RME PCIe card had
unbearable computer noise in the audio signal, some day it occurred,
stayed for months and then disappeared. I suspect a cable issue, but
can't say for sure. However, I'm using a new headphone cable, as well
as a new computer, since my old computer anyway was fishy and outdated.

Even if cheapest consumer gear does cause static noise, something is
fishy.

Regards,
Ralf

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

[Ralf Mardorf]

On Wed, 2017-08-09 at 08:42 +0200, Benjamin via Alsa-user wrote:
> Oh sorry that last email was a mistake! Could you please remove that!

1.

Most links to archives provided by
https://www.alsa-project.org/main/index.php/Mailing-lists#alsa-user_at_lists.sourceforge.net
are dead links, even googling for other archives, e.g. at nabble, leads
to nothing. However, if an admin would remove a mail from one mailing
list archive, there usually are copies provided by archives the admin
can't access.

2.

You sent this mail to the list. You reach other subscribers, but perhaps
not an admin. If an admin randomly should read your request and would be
willing to remove your mail, should the admin remove your request to
remove the original mail, too?

Résumé

Asking for a removal of something you sent to a mailing list makes no
sense, to the contrary, you only call attention on it.

I never heard of something that was sent to a mailing list and that was
removed from an archive. From time to time people sent requests to
remove something.

Note, there are likely thousands of copies on thousands of private
computers.


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Re: [Alsa-user] Route input to output with minimal latency

[Ralf Mardorf]

On Thu, 20 Jul 2017 18:14:07 +, Robert Bielik wrote:
>No, software input to output, i.e. capture data and send to playback
>directly with minimal latency.
>
>But I realize that mmapping is not possible since capture and playback
>device will have separate buffers, so I still have to copy from
>capture to playback buffers.

I don't know to what "zero-copy" in the description refers to, but even
if there should be the need to copy, I doubt that latency caused by
such copies would be an issue, at least not for a negligible amount of
IOs. I guess if latency isn't caused by audio hardware, than usually by
the signal processing of applications, but unlikely by the routing.
However, I never programmed using ALSA and my audio recording
experiences are from decades ago. I was just surprised about the
< 1 ms, since I would expect software side this shouldn't be an issue.

I might be mistaken.

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Re: [Alsa-user] Route input to output with minimal latency

[Ralf Mardorf]

On Thu, 20 Jul 2017 17:47:17 +, Robert Bielik wrote:
>I want to route input to output with minimal possible latency, this
>will run on a Raspberry Pi, and the latency should be < 1 ms.
>
>I was thinking... if the ALSA capture and playback device is mmapped
>to the same buffer area, this should be dealt with automatically. Is
>this possible ?

http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html

I wonder if you are mistaking apples for oranges. Do you want to
connect the audio device's hardware outputs with the audio device's
hardware inputs with cables or do you want to connect software outputs
with software inputs?

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Re: [Alsa-user] cards is moving around

[Ralf Mardorf]

On Mon, 3 Jul 2017 19:42:15 +0200, Kristoffer Gustafsson wrote:
>I want to use my hda-intel card.
>when i use aplay-l it is listed as card 1.
>I edit /usr/share/alsa.conf to 1.
>but when I edit it and reboot the intel card is card 2, so it uses the
>wrong card.
>What can I do about this?

Hi,

what is the output of

  aplay -l
  arecord -l
  amidi -l

resp. what driver/s do/es the other device/s use?

Add a file  /etc/modprobe.d/alsa-base.conf  with the following content:

  options snd slots=snd_hda_intel

Regards,
Ralf

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Re: [Alsa-user] Stub for libpulse

[Ralf Mardorf]

On Sun, 19 Mar 2017 17:42:56 +0100, Ralf Mardorf wrote:
>On Sun, 19 Mar 2017 17:36:20 +0100, Nicolas George wrote:
>>Thanks for the info. I tested with ogg123 because I did not want to
>>upgrade Firefox before being sure to have a solution. Strangely
>>enough, it works with Firefox indeed (tested with "Für Elise" from
>>Wikipedia rather than youtube, but that should not make any
>>difference). Good news.  
>
>I thought you already played the ogg with Firefox. Good to hear that it
>works for you with at least YouTube, too.
 
 Wiki :)

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Re: [Alsa-user] Stub for libpulse

[Ralf Mardorf]

On Sun, 19 Mar 2017 17:36:20 +0100, Nicolas George wrote:
>Thanks for the info. I tested with ogg123 because I did not want to
>upgrade Firefox before being sure to have a solution. Strangely enough,
>it works with Firefox indeed (tested with "Für Elise" from Wikipedia
>rather than youtube, but that should not make any difference). Good
>news.

I thought you already played the ogg with Firefox. Good to hear that it
works for you with at least YouTube, too.

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Re: [Alsa-user] Stub for libpulse

[Ralf Mardorf]

On Sun, 19 Mar 2017 12:36:46 +0100, Nicolas George wrote:
>I have found this project:
>https://github.com/i-rinat/apulse
>but it seems unable to get something as simple as ogg123 working.

Worked for me with Firefox 52.0 when at least making a test with
YouTube. Does it work for you with YouTube?

$ pacman -Q apulse-git 
apulse-git 0.1.7_13_gf445ae7-1

I can't test ogg at the moment, since I now have Firefox with alsa
enabled installed.


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Re: [Alsa-user] RME HDSPe AES32 linux debian 7

[Ralf Mardorf]

Hi Remy,

I can't help you with your issue, but I at least can report my
experiences with another HDSPe card.

The following output speaks for itself.

[rocketmouse@archlinux ~]$ hdspconf 

HDSPConf 1.4 - Copyright (C) 2003 Thomas Charbonnel 
This program comes WITH ABSOLUTELY NO WARRANTY
HDSPConf is free software, see the file copying for details

Looking for HDSP cards :
Card 0 : RME AIO S/N 0x579bcc at 0xf7d0, irq 16
Card 1 : TerraTec EWX24/96 at 0xd040, irq 16
Card 3 : HDA Intel HDMI at 0xf7e1 irq 30
No Hammerfall DSP card found.

However, hdspmixer works, but it seemingly is anything else, but close
to the current version of TotalMix.

I bought the card 6 years ago and never got access to more than 2 of
the 8 ADAT I/Os.

I'm using the RME card to mix music, because the audio quality of the
analog outputs is superb, but if I should need many I/Os and lowest
latency, I'm using a Focusrite Scarlett 18i20. In my experiences USB
allows lower latency, than PCIe does, let alone that in my case all
I/Os of the Focusride are available.

With the HDSPe AIO Ardour with ALSA instead of jackd doesn't work. This
seems to be an issue for some cards and at least the AIO is one of them.

I used the RME card with an AMD machine and now I'm using it with an
Intel machine.

Regards,
Ralf

[rocketmouse@archlinux ~]$ uname -a
Linux archlinux 4.9.13-rt12-1-rt-persianrug #1 SMP PREEMPT RT Wed Mar 8
02:21:10 CET 2017 x86_64 GNU/Linux

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Re: [Alsa-user] Create account on alsa-project wiki

[Ralf Mardorf]

On Sun, 18 Dec 2016 21:17:47 +0800, Adam Ward wrote:
>I am referring to account creation at http://www.alsa-project.org
>
>The URL I see is
>
>http://www.alsa-project.org/main/index.php?title=Special:UserLogin=signup=Help:Howto-Edit
> 

My apologies,

IIUC asking on this mailing list is the right way.

"To help protect against automated account creation, please answer the
question that appears below (more info):

Token to prevent spam? Please, ask on mailing lists..."

:)

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Re: [Alsa-user] Create account on alsa-project wiki

[Ralf Mardorf]

On Sun, 18 Dec 2016 16:02:25 +0800, Adam Ward wrote:
>I want to add details for the sun4i_codec.
>The wiki makes a point about captchas, but I do not see one.

Hi,

if it doesn't work, then try
https://en.wikipedia.org/wiki/Wikipedia:Request_an_account

FWIW I see a CAPTCHA at
https://en.wikipedia.org/w/index.php?title=Special:CreateAccount=ALSA

with Firefox 50.1.0, IceCat 45.5.1, Pale Moon 27.0.3,
QupZilla 2.0.2 (QtWebEngine 5.7.0), Chrome 55.0.2883.87,
Chromium 55.0.2883.87, Vivaldi 1.6.689.34, Opera 42.0.2393.85 and
Xombrero 1.6.4.

Tor exit nodes are blocked, tested with Tor browser 6.0.8 (based on
Mozilla Firefox 45.6.0).

Regards,
Ralf

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Re: [Alsa-user] ice1712 recording

[Ralf Mardorf]

On Thu, 8 Dec 2016 12:18:47 -0600, James Shatto wrote:
>As well as a few CLI options for the same.  Sox is good for that.

For SpaceFM I wrote a script to split and/or merge stereo wav. Written
on the fly, so most likely not good shell script writing, but it does
what it should do and IIRC ffmpeg was easier to use.

[rocketmouse@archlinux bin]$ ls -Ggl s2*
lrwxrwxrwx 13 Apr 27  2016 s2a -> s2m
lrwxrwxrwx 13 Apr 27  2016 s2d -> s2m
lrwxrwxrwx 13 Apr 27  2016 s2l -> s2m
-rwxr-xr-x 1 1791 Apr 28  2016 s2m
lrwxrwxrwx 13 Apr 27  2016 s2r -> s2m
[rocketmouse@archlinux bin]$ cat s2m
#!/bin/dash

version="2016-04-28"

usage()
{
  cat< mono.wav:
s2m input_file.wav
  stereo.wav > left.wav:  
s2l input_file.wav
  stereo.wav > right.wav:
s2r input_file.wav
  stereo.wav > dual {left,right}.wav:
s2d input_file.wav
  stereo.wav > all {mono,left,right}.wav:
s2a input_file.wav

EOF
  exit $1
}

outfile()
{
  outfile="$(echo "$infile" | sed 's/\(.*\).wav/\1_'$1.$suffix/I)"
  if [ -f "$outfile" ]; then
echo "$outfile already exists"
usage 1
  fi
}

ex2file()
{
  outfile $1
  echo "Export to $outfile"
  case $1 in
mono)
  ffmpeg -i "$infile" -ac 1 "$outfile"
;;
left)
  ffmpeg -i "$infile" -map_channel 0.0.0 "$outfile"
;;
right)
  ffmpeg -i "$infile" -map_channel 0.0.1 "$outfile"
;;
  esac
}

case $1 in
  -h|--help)
usage 0
  ;;
esac

infile="$1"
if [ ! -f "$infile" ]; then
  echo "No file $infile"
  usage 1
fi

num_ch=$(exiftool "$infile" | grep -v "File Name" | grep -v "Directory" | grep 
"Num Channels" | cut -d: -f2 | sed s/\ //)
plural=""
if [ "$num_ch" = "" ]; then
  num_ch="No"
else 
  if [ "$num_ch" -ge "2" ]; then
plural="s"
  fi
fi
if [ "$num_ch" != "2" ]; then
  echo "$num_ch channel$plural, not a stereo file"
  usage 1
fi

suffix=$(echo "$infile" | rev | cut -d. -f1 | rev)
if [ "$(echo "$suffix" | tr [:upper:] [:lower:])" != "wav" ]; then
  echo "Suffix is \"$suffix\", must be \"wav\", case sensitivity is not 
required"
  usage 1
fi

case $(basename $0) in
  s2m)
ex2file mono
  ;;
  s2l)
ex2file left
  ;;
  s2r)
ex2file right
  ;;
  s2d)
outfile right
ex2file left
ex2file right
  ;;
  s2a)
outfile left
outfile right
ex2file mono
ex2file left
ex2file right
  ;;
esac

exit



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Re: [Alsa-user] ice1712 recording

[Ralf Mardorf]

On Mon, 5 Dec 2016 14:27:22 +, zcx wrote:
>On 04/12/16 21:09, Ralf Mardorf wrote:
>> Only one app can grab the device, if you run two instances of the
>> same app, only one instance can grab the device.  
>
>Yes, this appears to be the case.  But do you know why?

A car has got four wheels, why can't you drive each wheel to a
completely different place, at the same time? Why are the four wheels
in sync? It doesn't matter what car you own, by design it's the same for
all cars. A car with four wheels and four seats provides only one seat
for a driver and only one steering wheel.

As already pointed out, one sound server is able to access the device
and several apps could access the sound server. Problem solved.

Regards,
Ralf

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Re: [Alsa-user] ice1712 recording

[Ralf Mardorf]

On Sun, 4 Dec 2016 20:18:24 +, zcx wrote:
>I have a Delta 44 sound card here that uses the ice1712 chipset.
>
>Am I right in thinking that although the card has 4 mono inputs, it
>can only capture one stream at a time?  arecord seems to think so...

Only one app can grab the device, if you run two instances of the same
app, only one instance can grab the device.

If several apps should be able to use the device at the same time, you
need a workaround, e.g. dmix or e.g. a sound server, such as e.g. jackd.

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Re: [Alsa-user] sans-pulseaudio Firefox? was: a strange thing

[Ralf Mardorf]

On Mon, 28 Nov 2016 18:00:49 +0100, Miroslav Rovis wrote:
>You are a dev, and I respect that.

JFTR I'm not an ALSA dev.

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Re: [Alsa-user] sans-pulseaudio Firefox? was: a strange thing

[Ralf Mardorf]

Miroslav, you don't need to know the Arch package manager commands, just
the output is important. I'm using Firefox without pulseaudio. You don't
need to switch to Arch Linux, this works on other distros, too. Firefox
doesn't require pulseaudio.

Arch Linux does use systemd and it's more or less impossible to replace
it by another init system. However, this is off-topic, as well as
discussing spyware in browsers and similar issues or talking about
Poettering, Sievers and friends.

FWIW 
http://lists.linuxaudio.org/pipermail/linux-audio-user/2016-June/105188.html
I didn't check if it's already available.

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Re: [Alsa-user] sans-pulseaudio Firefox? was: a strange thing

[Ralf Mardorf]

On Wed, 16 Nov 2016 14:48:23 +0100, Miroslav Rovis wrote:
>"build your own Firefox with ALSA enabled"

libpulse is just a make dependency, but even if it should be installed,
it doesn't matter.

https://www.archlinux.org/packages/extra/x86_64/firefox/
https://git.archlinux.org/svntogit/packages.git/tree/trunk?h=packages/firefox

I'm using Firefox on Arch Linux without pulseaudio.

[weremouse@moonstudio archlinux]$ sudo systemd-nspawn -q pacman -Qi firefox 
pulseaudio libpulse | grep Na -A2
Failed to create directory /mnt/archlinux/sys/fs/selinux: Read-only file system
Failed to create directory /mnt/archlinux/sys/fs/selinux: Read-only file system
Name: firefox
Version : 50.0-1
Description : Standalone web browser from mozilla.org
--
Name: pulseaudio
Version : 2013.08.18-1
Description : Dummy package
--
Name: libpulse
Version : 9.0-1
Description : A featureful, general-purpose sound server (client library)

To use Firefox without pulseaudio, I never needed to build it myself.
Even not on other distros, but Arch is my everyday Linux and I anyway
try to stay away from Firefox, so often it's not installed.

[weremouse@moonstudio archlinux]$ ls -hAl /usr/local/bin/firefox; apt list 
qupzilla; lsb_release -d
lrwxrwxrwx 1 root root 6 Jan 12  2016 /usr/local/bin/firefox -> icecat
Listing... Done
qupzilla/xenial,now 1.8.9~dfsg1-3 amd64 [installed]
Description:Ubuntu 16.04.1 LTS

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Re: [Alsa-user] a strange thing

[Ralf Mardorf]

On Mon, 14 Nov 2016 16:37:28 +0100, John P. Hartmann wrote:
>Other applications may need pulse audio to produce sound altogether.

AFAIK skype requires pulseaudio. I'm using plain ALSA or jackd. The OP
might need pulseaudio, if getting sound from several desktop
applications at the same time is wanted, without taking care about
another sound server, such as jackd, or using another alternative such
as dmix.

I assume it was not intended to send it off-list:

Begin forwarded message:

Date: Mon, 14 Nov 2016 16:37:28 +0100
From: "John P. Hartmann"
To: Ralf Mardorf
Subject: Re: [Alsa-user] a strange thing


pulseaudio -k

And to start it, pulseaudio --start

On my Ubuntu 14.4, the control panel applet for sound contains nothing 
unless pulseaudio is running.  I also discovered that firefox needs the 
proper card configured in the applet, but once that is done, pulse
audio is not needed to run.

Other applications may need pulse audio to produce sound altogether.

On 11/14/2016 04:24 PM, Ralf Mardorf wrote:
> I agree, for troubleshooting purpose disabling pulseaudio is a good
> idea. I don't know how to do this,

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Re: [Alsa-user] a strange thing

[Ralf Mardorf]

My apologies,

>> On Mon, 14 Nov 2016 06:33:28 -0800, chris hermansen wrote:  
>http://kodi.wiki/view/PulseAudio/HOW-TO:_Disable_PulseAudio_and_use_ALSA_(without_removing_PulseAudio)_for_Ubuntu

I agree, for troubleshooting purpose disabling pulseaudio is a good
idea. I don't know how to do this, but likely the link provided by
Chris explains how to do this. AFAIK how to disable pulseaudio could
depend on the used version of pulseaudio.

Regards,
Ralf

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Re: [Alsa-user] a strange thing

[Ralf Mardorf]

On Mon, 14 Nov 2016 06:33:28 -0800, chris hermansen wrote:
>On Nov 14, 2016 06:18, "Kristoffer Gustafsson" wrote:
>> Is there a way to remove pulse audio then and just use alsa?
>> If so I can use my soundcard without having to buy Another one. 
>
>Please read the following
>
>http://kodi.wiki/view/PulseAudio/HOW-TO:_Disable_PulseAudio_and_use_ALSA_(without_removing_PulseAudio)_for_Ubuntu
>
>It explains why and how to disable pulse rather than uninstall it.

1.

We don not know if pulseaudio is the culprit.

2.

Even if we should know that it is the culprit, depending on the OP's
needs, it might be wiser to fix the issue and to use pulseaudio.

3.

Assuming regarding the OP needs it should be better to get rid of
pulseaudio, then it would be idiotic to disable it, instead of removing
it completely. Why should somebody disable it, if it's completely
unwanted? If I don't need a package, I don't install it. If something I
should need, works without it, but has got a hard dependency to it, I
simply build an empty dumm package.

Any evidence for the below claim?

"Removing or purging PulseAudio from a Ubuntu system can break other
installed software and potentially cause errors and broken
dependencies."

This is just a claim without explanation. As already pointed out, just
replace it by a dummy package, you even don't need to recompile
software.

If needed I could explain how to build a DEB dummy package and how to
build an Arch Linux dummy package. I'm using them since pulseaudio does
exist without ever experiencing an issue.

But again, removing pulseaudio might not what the OP wants? It's just a
_possibilities_ that pulseaudio is the culprit for the issues, nobody
mentions that it's the cause for the issues.

Regards,
Ralf

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Re: [Alsa-user] a strange thing

[Ralf Mardorf]

On Mon, 14 Nov 2016 15:15:18 +0100, Kristoffer Gustafsson wrote:
>Is there a way to remove pulse audio then and just use alsa?
>If so I can use my soundcard without having to buy Another one.

Actually we need to know what you exactly want to do, with audio. Sure,
assuming your GNOME desktop should have got a hard dependency to
pulseaudio, it most likely would be enough to replace it, with an empty
dummy package, to get rid of pulseaudio. Getting rid of pulseaudio
might not solve your issue and depending on your needs, you might want
to use pulseaudio.

Did you already sent a request to a support channel of the distro you
are using? IOW did you ask on a distro related forum or mailing list?

Regards,
Ralf

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Re: [Alsa-user] a strange thing

[Ralf Mardorf]

On Mon, 14 Nov 2016 09:37:18 +0100, Kristoffer Gustafsson wrote:
>Today I tried installing debian without gnome just to have fun.
>then the card worked!
>how can this be?

Two of several possibilities

1. No pulseaudio

2. Randomly it becomes the default hw:0 at startup, without an
   alsa-base.conf to ensure this.

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

Regards,
Ralf

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Re: [Alsa-user] all soundcards you use

[Ralf Mardorf]

PCI

TerraTec EWX 24/96

works without limitation
very cheap at ebay
usable audio quality
Linux mixer for this card is available
did not test Ardour ALSA without jackd


PCIe

RME HDSPe AIO

Only 2 of 8 ADAT IOs are accessible (this issue doesn't happen on
FreeBSD or Windows)
Linux mixer for this card available, but it's not the latest Total Mix
Ardour ALSA without jackd doesn't work
xrun issues even with long latencies, but even with xruns amazing good
audio quality
relatively expensive audio card

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Re: [Alsa-user] why does alsa want jack

[Ralf Mardorf]

Hi Kristoffer,

On Tue, 25 Oct 2016 02:01:07 +0200, Kristoffer Gustafsson wrote:
>I get that error all the time on my computer abot jack server is not
>running or cannot be started.

it for sure isn't ALSA that wants jack, most likely an app requires
jack. You need to describe more precisely when and how you get the
message. Pops up a window, if you log in a user session? Do you start
an app and the app opens an error window?

>No sound at all in linux.

Ok, here 'linux' seemingly is for the install as a whole, the kernel
and applications.

>I solved this by reinstalling linux, but I don't want to reinstall all
>the time. this is strange.

Do you reinstall 'linux' the kernel, or do you reinstall everything?

However, after reinstalling sound works for a while and after a while
it stops working?

For what purpose do you need sound? Do you want to hear sound made by a
desktop environment? If you open a browser, to listen to the sound of a
video? When running a virtual synth?

What window manager/desktop environment do you run? Unity, GNOME, KDE,
Xfce ...? What distro? Ubuntu, Suse, Debian ...?

Is jack installed? Yes? No? You don't know?

Regards,
Ralf

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Re: [Alsa-user] Logitech Webcam C270; sound not working

[Ralf Mardorf]

On Mon, 19 Sep 2016 15:01:51 +0200, Vladimir Savic wrote:
>On Mon, Sep 19, 2016 at 2:14 PM, Ralf Mardorf
>> your old syntax was _incomplete_.  
>
>Funnily enough, I never touched that file. I haven't manually
>configured anything. Strange...

Hi,

you mentioned "Antergos linux (Arch based)". At least a default
Arch Linux install doesn't add anything to /etc/modprobe.d/. No package
I'm aware off generates /etc/modprobe.d/alsa-base.conf with some
defaults, let alone that something detects available audio devices and
ensures that they have a fixed order. I'm mainly running Arch Linux,
with much real-time audio software.

To see if a package owns /etc/modprobe.d/alsa-base.conf run

  [rocketmouse@archlinux ~]$ pacman -Qo /etc/modprobe.d/alsa-base.conf
  error: No package owns /etc/modprobe.d/alsa-base.conf

Very unlikely, but still possible a package generated
/etc/modprobe.d/alsa-base.conf with some default settings, but unlikely
that it also detects available devices and decides in what order those
devices should be made available during startup. Assuming that a
package provides some defaults, than more likely a package
owns /etc/modprobe.d/alsa-base.conf instead of generating it, but even
then it unlikely index modules of available devices, to sort them that
way.

Regards,
Ralf

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Re: [Alsa-user] Logitech Webcam C270; sound not working

[Ralf Mardorf]

On Mon, 19 Sep 2016 10:14:13 +0200, Vladimir Savic wrote:
>Thanks, peaope!
>
>Worked like charm. I still don't get why though. It is the same thing,
>but written using different syntax, or? Who knows...
>
>Thanx again!

You are welcome!

What does your alsa-base.conf look like now?

The "slot" method is more comfortable, as already pointed out

  options snd slots=,snd_usb_audio

would work, too. Also

  options snd slots=snd_hda_intel

would do the job.

My does look like this

  [rocketmouse@archlinux ~]$ cat /etc/modprobe.d/*alsa*.conf 
  # ALSA module ordering
  options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

So even while the two ICE1712 cards are removed...

  [rocketmouse@archlinux ~]$ aplay -l
   List of PLAYBACK Hardware Devices 
  card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
Subdevices: 1/1
Subdevice #0: subdevice #0
  [rocketmouse@archlinux ~]$ arecord -l
   List of CAPTURE Hardware Devices 
  card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
Subdevices: 1/1
Subdevice #0: subdevice #0
  [rocketmouse@archlinux ~]$ amidi -l
  Dir DeviceName
  IO  hw:0,0HDSPMx579bcc MIDI 1
  IO  hw:3,0,0  USB Device 0x170b:0x11 MIDI 1
  IO  hw:4,0,0  nanoKONTROL MIDI 1

...hw:1 and hw:2 are reserved for them and two attached USB
devices became hw:3 and hw:4.

If you are using the corrected "index" method, then your old syntax was
_incomplete_.

Please, reply to the list and bottom post...

Regards,
Ralf

...and remove signatures, respl. everything unneeded, like the
"---"...

>On Mon, Sep 19, 2016 at 9:40 AM, Ralf Mardorf wrote:
>
>> On Mon, 19 Sep 2016 09:07:28 +0200, Vladimir Savic wrote:  
>> >$ cat alsa-base.conf
>> >options snd_hda_intel index=0
>> >options snd_usb_audio index=1  
>>
>> Delete this and edit alsa-base.conf to
>>
>>options snd slots=snd_hda_intel,snd_usb_audio
>>
>> or if you should insist in using the old index method edit it too
>>
>>   options snd cards_limit=2
>>   alias snd-card-0 snd_hda_intel
>>   alias snd-card-1 snd_usb_audio
>>   options snd_hda_intel index=0
>>   options snd_usb_audio index=1
>>
>> Regards,
>> Ralf
>>
>> 
>> --

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Re: [Alsa-user] Logitech Webcam C270; sound not working

[Ralf Mardorf]

On Mon, 19 Sep 2016 09:07:28 +0200, Vladimir Savic wrote:
>$ cat alsa-base.conf
>options snd_hda_intel index=0
>options snd_usb_audio index=1  

Delete this and edit alsa-base.conf to

   options snd slots=snd_hda_intel,snd_usb_audio

or if you should insist in using the old index method edit it too

  options snd cards_limit=2
  alias snd-card-0 snd_hda_intel
  alias snd-card-1 snd_usb_audio
  options snd_hda_intel index=0
  options snd_usb_audio index=1

Regards,
Ralf

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Re: [Alsa-user] Logitech Webcam C270; sound not working

[Ralf Mardorf]

On Mon, 19 Sep 2016 08:48:24 +0200, Clemens Ladisch wrote:
>Vladimir Savic wrote:
>> [  395.494123] snd-usb-audio 1-12:1.2: cannot find the slot for
>> index 1 (range 0-1), error: -16  
>
>Some configuration settings in some .conf file in /etc/modprobe.d/ are
>wrong.

Try running

  cat /etc/modprobe.d/*alsa*.conf

Did you try to get a steady module ordering?

Assuming you did and you wanted to get snd_usb_audio as hw:0, then
remove what ever you added and try adding a file
named /etc/modprobe.d/alsa-base.conf

  options snd slots=snd_usb_audio

or if it should become hw:1

  options snd slots=,snd_usb_audio

Regards,
Ralf

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Re: [Alsa-user] Microphone volume on USB soundcard very low, are there any snd_usb_audio options?

[Ralf Mardorf]

On Tue, 2 Aug 2016 11:38:59 +0200, Ralf Mardorf wrote:
>are you using pulseaudio?

My apologies, I missed "The tools I'm using are ALSA only".

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Re: [Alsa-user] Microphone volume on USB soundcard very low, are there any snd_usb_audio options?

[Ralf Mardorf]

On Tue, 2 Aug 2016 10:05:36 +0200, Schmidt, Christian Thorge wrote:
>Hello all,
>
>I have a problem with the microphones volume being way too low
>(compared to what the same connector does under macOS) on a USB
>"soundcard". I have described the whole ordeal here (with pictures and
>system info):
>http://unix.stackexchange.com/questions/299322/microphone-volume-on-usb-soundcard-very-low-snd-usb-audio-options
>
>tl:dr: A "dedicated microphone adapter", eg. a C-Media USB soundcard
>device with an XLR mic connector (Bus 001 Device 004: ID 0d8c:0008
>C-Media Electronics, Inc.) that uses snd_usb_audio yields a very low
>mic input even when fully turned "up" all the way under alsamixer.
>There's no "boost" control for the input device. The same adapter
>yields a clean and strong signal under a different OS.
>
>As always, thanks for any pointers, any help or encouraging words ;)

Hi,

are you using pulseaudio?

I don't have pulseaudio installed, perhaps pulseaudio has an additional
volume control and you need to increase this, too, by using pavucontrol
as suggested by the link you posted? Did you already tested it?

Is there a setting regarding +4 dBu / -10 dbV or something related
to a possible included microphone amp for MacOS? Does your MacOS use
a proprietary driver or like Linux a class compliant default thingy?

What is the product name? And what does
http://www.cmedia.com.tw/support say to this issue?

Regards,
Ralf

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Re: [Alsa-user] usb audio. should this not work?

[Ralf Mardorf]

PS:

Are you using a desktop environment with a sound server other than jack,
but most likely pulseaudio? You might need a tool like   pavucontrol  to
make the USB device available.


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Re: [Alsa-user] usb audio. should this not work?

[Ralf Mardorf]

On Fri, 1 Jul 2016 12:56:55 +0200, Kristoffer Gustafsson wrote:
>The things you said on this list on using mu ysb soundcard didn't work
>at all. to use my intel card i created a file called default.conf
>and inside that file I type
>options snd-hda-intel index=1
>so I tried to replase it with
>options snd-usb-audio index=1
>I thought that should work, but it didn't

Usually the file is named /etc/modprobe.d/alsa-base.conf
or /etc/modprobe.d/alsa.conf .

Content of the file, old school, but nobody is doing it anymore:

  options snd cards_limit=2
  alias snd-card-0 snd-usb-audio
  alias snd-card-1 snd-hda-intel
  options snd-usb-audio index=0
  options snd-hda-intel index=1

New method:

  options snd slots=snd-usb-audio,snd-hda-intel

Respectively just ...

  options snd slots=snd-usb-audio

or just ...

  options snd slots=,snd-hda-intel

I guess it doesn't matter if you use a "-" or "_", I used the "_":

  [rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
  # ALSA module ordering
  options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

Regards,
Ralf

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Re: [Alsa-user] how to make sure internal card is card 0

[Ralf Mardorf]

On Fri, 03 Jun 2016 02:30:16 -0400, cov...@ccs.covici.com wrote:
>Ralf Mardorf <ralf.mard...@alice-dsl.net> wrote:
>
>> On Fri, 27 May 2016 04:58:44 -0400, cov...@ccs.covici.com wrote:  
>> >I have an alsa.conf, but not alsa-base.conf.  
>> 
>> Assumed the alsa.conf should be located in /etc/modprobe.d/, then add
>> 
>>   options snd slots=snd_emu10k1
>> 
>> to the bottom of this file.  
>
>Thanks, it worked perfectly!

You're welcome!

Regards,
Ralf

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Re: [Alsa-user] Mutiple soundcards

[Ralf Mardorf]

On Tue, 31 May 2016 16:20:27 -0300, José Luis Artuch wrote:
>I do not understand what has been the problem ...

Perhaps a bad connection.

Issues for my motherboard also were caused by a weak CMOS battery,
without getting informed about the weak battery. One time I even thought
my mobo is completely broken. After resetting the CMOS RAM by jumper,
not by just removing the battery, everything was ok again.

Assumed issues should happen from time to time and after
a while everything is ok again, I disconnect and connect everything
again, replace the battery and clear the CMOS RAM by jumper.

Assumed the capacity of the PSU is merely adequate, it might cause such
an issue, too, I don't know.

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Re: [Alsa-user] how to make sure internal card is card 0

[Ralf Mardorf]

On Fri, 27 May 2016 04:58:44 -0400, cov...@ccs.covici.com wrote:
>I have an alsa.conf, but not alsa-base.conf.

Assumed the alsa.conf should be located in /etc/modprobe.d/, then add

  options snd slots=snd_emu10k1

to the bottom of this file.

Regards,
Ralf

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Re: [Alsa-user] how to make sure internal card is card 0

[Ralf Mardorf]

Regarding https://sourceforge.net/p/alsa/mailman/alsa-user/?style=flat
neither somebodies reply, nor my reply came through the list, that's why
I resend my reply and Cc now.

Begin forwarded message:

Date: Fri, 27 May 2016 10:16:21 +0200
To: alsa-user@lists.sourceforge.net
Subject: Re: [Alsa-user] how to make sure internal card is card 0


Hi,

write /etc/modprobe.d/alsa-base.conf [1] with the content

  options snd slots=snd_emu10k1

assumed snd_emu10k1 (snd-emu10k1) should be the module for your internal
sound card. The Emu card then always will be hw:0 and the USB device
always hw:1.

Regards,
Ralf

[1]
For my machine USB devices always are >= hw:3.

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

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Re: [Alsa-user] how to make sure internal card is card 0

[Ralf Mardorf]

Hi,

write /etc/modprobe.d/alsa-base.conf [1] with the content

  options snd slots=snd_emu10k1

assumed snd_emu10k1 (snd-emu10k1) should be the module for your internal
sound card. The Emu card then always will be hw:0 and the USB device
always hw:1.

Regards,
Ralf

[1]
For my machine USB devices always are >= hw:3.

[rocketmouse@archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

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Re: [Alsa-user] combine usb microphones

[Ralf Mardorf]

On Fri, 15 Apr 2016 13:52:28 + (UTC), joop wrote:
>Hi all,
>
>i have 4 usb microphones connected to my system that i would like to
>combine into one virtual 'sound card' with 4 seperate mic channels.
>
>i've researched this but could not find an answer...
>
>what would i put in my .asoundrc?

I don't know, but using 4 mics as one virtual sound card, without
sync, leads to nothing. Maybe this is useful:
http://kokkinizita.linuxaudio.org/linuxaudio/zita-ajbridge-doc/quickguide.html

I don't know, but you at least need some kind of hardware sync or
software resampling.


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Re: [Alsa-user] Alsa Multitrack Line In

[Ralf Mardorf]

Hi,

why not using jack with the ALSA backend, together with pulseaudio or
without pulseaudio at all, instead of using ALSA directly?

http://www.jackaudio.org/

There might be reasons to not use jack, but at least for multi-track
recording this IMO is the best and easiest approach. FWIW Ardour now
allows to use ALSA directly, seemingly without the need to set up ALSA
on your own. I couldn't test it, since it doesn't work with all RME
cards, but most likely there are no issues with Envy24 (ice1712) cards
when using Ardour.

IOW depending on what you want to do, consider to use jack and/or
Ardour.

Regards,
Ralf

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Re: [Alsa-user] intel hda vs asus xonar STX

[Ralf Mardorf]

Perhaps a sound server defaults to use the integrated sound device?
Pulseaudio nowadays seems to be installed as the default sound server by
most Linux distros that provide an out of the box working
desktop environment and even for other distros it's an annoying
dependency. For my Linux installs empty dummy packages fake to fulfil
the pulseaudio dependency, while I'm using jackd to use several audio
streams.

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Re: [Alsa-user] Unbalanced stereo input as balanced mono input

[Ralf Mardorf]

On Sun, 23 Aug 2015 18:07:32 -0700 (PDT), Bill Unruh wrote:
Now, that may mean that there is common mode noise, in which case
running the balanced into an unbalanced would be very noisy, or the
noise was all in the ground wire, in which case bal-unbal might work.

It's the nature of unbalanced connections, that they are noisier than
balanced connections.

However, there are several ways to implement balanced IOs. Using two
unbalanced mono input channels as a single balanced input channel is
not one of that ways.

Most important is that the output and input fit together, than to care
about balanced or unbalanced. Cable length and impedance need more
attention.

The microphone might provide line output (unusual, but not impossible).

The OP should consider to use microphone output instead and to use a
pre-amp for the input.

I suspect to care about cable length and impedance is more important
than to fake a balanced input. I doubt that a faked balanced input using
two unbalanced mono inputs of an elcheapo audio device is good for any
kind of measurement or audio recording.

Btw. there's always a discussion if the cold output should be
open-circuit for balanced output to unbalanced input. So you could
consider the unbalanced to unbalanced connections I posted as wrong.
OTOH a connection of cold and ground is wanted for unbalanced output to
balanced input. We could discuss the amount of cores of the cable, to
provide the latter, but since this thread is about balanced to faked
balanced, we could add a discussion about balanced to balanced
grounding. The grounding between balanced IOs could also be done in
different ways.

Note that he does not say what measurements he wants to do. If it is
just levels then the crappy sound card might do, but even freq
response is liable to be dominated by his the response of his sound
card. And noise is almost certainly dominated by the sound card.

IIRC the OP even didn't mention if the microphone should be used for
measurements or underwater vocal recording.

We aren't talking about a professional solution, we are talking about
the advantage to fake a balanced input when using a crappy user input
device in combination with the balanced output of a measuring
microphone.

There's no advantage! We do not need to care about technical reasons
pro and con a faked balanced input by two unbalanced inputs. We simply
should notice that 1. relatively good balanced pre-amps are very cheap
nowadays and 2. they were more expensive years ago, when nobody
considered to make two unbalanced inputs of home recording gear a faked
balanced input. Why wasn't this done? Nobody was brilliant enough to
think about this superp idea?

Unbalanced stereo input as balanced mono input is an utterly wrong
idea.

I recommend to get a cheap, but anyway relatively good microphone
pre-amp or directly pay for a better audio card with an integrated
microphone pre-amp and FWIW to use defaults regarding the grounding,
IOW not to modify gear and to use averaged, common cables.

If getting new equipment shouldn't be an option, I would use the
provided IOs as they are and not fake a balanced input.

2 Cents,
Ralf

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Re: [Alsa-user] Unbalanced stereo input as balanced mono input

[Ralf Mardorf]

On Mon, 24 Aug 2015, Gunnar Arndt wrote:
Btw, that noise is actually from the on-board wiring - it's there even 
if nothing is connected to line input.

If nothing is connected even the best discrete circuits could be noisy.
For e.g. phono pre-amps it's common to insert short circuit connectors,
when they are unused.

 Another analog hardware solution would be to use an audio
 isolation transformer in front of your digitizer.  Radio Shack
 used to sell a fairly cheap audio isolation transformer that
 worked surprisingly well.  A metal box for shielding is probably
 a good idea.  
I'm not sure if I understand correctly what its purpose would be - 
unbalance the signal, like what I know as an opposite DI box?
If you put under consideration that the noise comes from the mainboard 
for the most part, you'll agree that it would not make sense to 
unbalance the signal before reaching the mainboard, as some box would 
require.

It makes sense!

Mic balanced -long-wire- balanced in, circuit with transformer, unbalanced out 
-short-wire- unbalanced in, mobo audio device

If a transformer is the best solution for your usage is questionable, but
it for sure is a better solution, than what you're doing.

I know that a better sound card is required - especially for the 
frequency range: the mic goes down to 9Hz, whereas the current sound 
chip goes only to 20Hz.

Audio engineering isn't rocket science.

A lot of quite good prosumer sound cards can't go lower than 20 Hz.
Don't care that much about key data, such as signals that might have half of
the level, than other signals or a wide frequency response range.
The quality of circuits and signals depends on balance of a lot of
specifications, that have to fit to the usage.

Consider your faked balanced input as very imbalanced. If you get rid of
noise, you not necessarily win sound quality, let alone a linear or what
ever kind of response that might be wanted for measurements.

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