Re: [Alsa-user] snd_hdsp missing on raspberrypi pi 5

[Robert M. Riches Jr.]

Are you sure those are current?  "oss" usually refers to the
open-source-sound that predated ALSA around 25 years ago.

The modinfo description from kernel 5.10.0 for snd-pcm-oss
seems to indicate it's for OSS emulation: "PCM OSS emulation
for ALSA."  For kernel 5.10.0-29-amd64 from Devuan Chimaera,
snd-seq-oss is not even listed.

HTH

-- 
Robert Riches
spamtra...@jacob21819.net
(Yes, that is one of my email addresses.)



 Original Message 
> Message-ID: 
> Date: Sun, 19 May 2024 20:51:34 +0200
> To: alsa-user@lists.sourceforge.net
> From: David Kessler 
> Subject: Re: [Alsa-user] snd_hdsp missing on raspberrypi pi 5

> I suspect I am missing some modules. Respecively snd-pcm-oss and 
> snd-seq-oss. I am trying to find clues about where to enable it in 
> menuconfig. It was easier for snd-hdsp (:
> 
> Hope this will work! I can feel it is close but I am waiting to be 
> disappointed also...
> 
> Have a nice day!
> 
> Le 19.05.24 à 19:40, David Kessler a écrit :
> >
> > I am now running a kernel with PCI soundcard hammerfall hdsp enabled 
> > and snd-hdsp module loaded.
> >
> > But it's not happy because it has no available buffer...
> >
> > dsp@dsp:~ $sudo dmesg | grep snd
> > [    0.00] Kernel command line: reboot=w coherent_pool=1M 
> > 8250.nr_uarts=1 pci=pcie_bus_safe snd_bcm2835.enable_compat_alsa=0 
> > snd_bcm2835.
> > enable_hdmi=1  smsc95xx.macaddr=2C:CF:67:14:D6:4C 
> > vc_mem.mem_base=0x3fc0 vc_mem.mem_size=0x4000 
> >  console=ttyAMA10,115200 console=tty1
> > root=PARTUUID=4b23ff89-02 rootfstype=ext4 fsck.repair=yes rootwait 
> > quiet splash plymouth.ignore-serial-consoles
> > [    2.909467] snd_hdsp :02:01.0: enabling device ( -> 0002)
> > [    2.912961] snd_hdsp :02:01.0: RME Hammerfall DSP: no buffers 
> > available
> > [    2.912981] snd_hdsp: probe of :02:01.0 failed with error -12
> >
> > It took me a day to get here. Slow internet not helping as well 
> > neither does my rusty kernel compilling skills.
> >
> > Thank's for reaching me out to any clue what is going on here <3
> >
> > Le 19.05.24 à 11:55, David Kessler a écrit :
> >> Ok thanks I've came across this webpage also. I'am missing the 
> >> alsa-driver tarball to do make configure as explained here:
> >>
> >> https://www.alsa-project.org/wiki/Matrix:Module-hdsp
> >>
> >> I look further to compile the module into the kernel which make most 
> >> sense. So is itswapped essentially to save space?
> >>
> >> Le 19.05.24 à 11:44, David Kessler a écrit :
> >>> I am now running ubuntu server 24.0 lts
> >>>
> >>> jean@dsp:~$ modinfo soundcore
> >>> name:   soundcore
> >>> filename:   (builtin)
> >>> alias:  char-major-14-*
> >>> license:    GPL
> >>> file:   sound/soundcore
> >>> author: Alan Cox
> >>> description:    Core sound module
> >>> parm:   preclaim_oss:int
> >>> jean@dsp:~$ lspci
> >>> :00:00.0 PCI bridge: Broadcom Inc. and subsidiaries BCM2712 PCIe 
> >>> Bridge (rev 21)
> >>> :01:00.0 PCI bridge: ASMedia Technology Inc. ASM1083/1085 PCIe 
> >>> to PCI Bridge (rev 04)
> >>> :02:01.0 Multimedia audio controller: Xilinx Corporation RME 
> >>> Hammerfall DSP (rev 37)
> >>> 0001:00:00.0 PCI bridge: Broadcom Inc. and subsidiaries BCM2712 PCIe 
> >>> Bridge (rev 21)
> >>> 0001:01:00.0 Ethernet controller: Raspberry Pi Ltd RP1 PCIe 2.0 
> >>> South Bridge
> >>>
> >>> jean@dsp:~$ sudo dmesg | grep 0:02:01.0
> >>> [    2.311018] pci :02:01.0: [10ee:3fc5] type 00 class 0x040100 
> >>> conventional PCI endpoint
> >>> [    2.326482] pci :02:01.0: BAR 0 [mem 0x1b-0x1b]
> >>> [    2.365346] pci :02:01.0: BAR 0 [mem 
> >>> 0x1b-0x1b]: assigned
> >>>
> >>> jean@dsp:~$ modinfo snd_hdsp
> >>> modinfo: ERROR: Module snd_hdsp not found.
> >>>
> >>> jean@dsp:~$ hdsploader
> >>> hdsploader - firmware loader for RME Hammerfall DSP cards
> >>> Looking for HDSP + Multiface or Digiface cards :
> >>> Card 0 : vc4-hdmi-0
> >>> Card 1 : vc4-hdmi-1
> >>>
> >>> Looks like a trivial problem. I would be very happy if someone could 
> >>> learn me the next steps involved to make this card work on a 
> >>> raspberrypi 5.
> >>>
> >>> Much thanks!
> >>>
> >>> Le 19.05.24 à 10:16, David Kessler a écrit :
>  I am trying to bring to life a hammerfall soundcard on a RPI5. I'ts 
>  attached through a PCI interface to PCIe 1x interface to the PI. 
>  THe card is powered by eternal supply and is recognized with lspci 
>  like everything is normal but snd-hdsp module is missing on that 
>  kernel.
> 
>  I have alsa-firmware-loaders installed but still no snd-hdsp module
> 
>  That's where I get lost. I assume I have to compile the kernel with 
>  the right modules
> 
>  Is there someone which could help me out?
> 
>  Thank's!
> 
> 
> 
> 
> 
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Re: [Alsa-user] any hope for alsamixer controls for AMD RX 5500/5500M GPU?

[Robert M. Riches Jr.]

Alan, thank you for the quick and helpful reply.  After sending
the initial message to the list, I discovered that the GPU's
sound device has hideous latency when doing things like jumping
ahead or backward in mplayer.  Worse, when pausing and resuming
mplayer, a substantial fraction of a second of sound is lost.
For my use cases, those are intolerable problems with that sound
device.  I'll have to abandon the idea of using sound output
through the GPU.

On the off chance it might help someone else with a similar GPU
sound issue, here's a summary of what I found through "aplay -l"
and "amixer ...".  Comparing the below results vs. the
motherboard sound 'card' that does show a volume control in
alsamixer, it would appear the GPU's sound device does not
provide volume control at all.

First, "aplay -l" does show the devices numbered oddly:

card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Doing "amixer -c 0" gives this:

Simple mixer control 'IEC958',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'IEC958',1
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'IEC958',2
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'IEC958',3
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]
Simple mixer control 'IEC958',4
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [on]

Doing "amixer -c 0 info" gives this:

Card hw:0 'HDMI'/'HDA ATI HDMI at 0xfcea irq 107'
  Mixer name: 'ATI R6xx HDMI'
  Components: 'HDA:1002aa01,00aa0100,00100700'
  Controls  : 35
  Simple ctrls  : 5

Doing "amixer -c 0 scontrols" gives this:

Simple mixer control 'IEC958',0
Simple mixer control 'IEC958',1
Simple mixer control 'IEC958',2
Simple mixer control 'IEC958',3
Simple mixer control 'IEC958',4

Doing "amixer -c 0 controls" gives this:

numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack'
numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack'
numid=7,iface=CARD,name='HDMI/DP,pcm=7 Jack'
numid=13,iface=CARD,name='HDMI/DP,pcm=8 Jack'
numid=19,iface=CARD,name='HDMI/DP,pcm=9 Jack'
numid=2,iface=MIXER,name='IEC958 Playback Con Mask'
numid=8,iface=MIXER,name='IEC958 Playback Con Mask',index=1
numid=14,iface=MIXER,name='IEC958 Playback Con Mask',index=2
numid=20,iface=MIXER,name='IEC958 Playback Con Mask',index=3
numid=26,iface=MIXER,name='IEC958 Playback Con Mask',index=4
numid=3,iface=MIXER,name='IEC958 Playback Pro Mask'
numid=9,iface=MIXER,name='IEC958 Playback Pro Mask',index=1
numid=15,iface=MIXER,name='IEC958 Playback Pro Mask',index=2
numid=21,iface=MIXER,name='IEC958 Playback Pro Mask',index=3
numid=27,iface=MIXER,name='IEC958 Playback Pro Mask',index=4
numid=4,iface=MIXER,name='IEC958 Playback Default'
numid=10,iface=MIXER,name='IEC958 Playback Default',index=1
numid=16,iface=MIXER,name='IEC958 Playback Default',index=2
numid=22,iface=MIXER,name='IEC958 Playback Default',index=3
numid=28,iface=MIXER,name='IEC958 Playback Default',index=4
numid=5,iface=MIXER,name='IEC958 Playback Switch'
numid=11,iface=MIXER,name='IEC958 Playback Switch',index=1
numid=17,iface=MIXER,name='IEC958 Playback Switch',index=2
numid=23,iface=MIXER,name='IEC958 Playback Switch',index=3
numid=29,iface=MIXER,name='IEC958 Playback Switch',index=4
numid=6,iface=PCM,name='ELD',device=3
numid=31,iface=PCM,name='Playback Channel Map',device=3
numid=12,iface=PCM,name='ELD',device=7
numid=32,iface=PCM,name='Playback Channel Map',device=7
numid=18,iface=PCM,name='ELD',device=8
numid=33,iface=PCM,name='Playback Channel Map',device=8
numid=24,iface=PCM,name='ELD',device=9
numid=34,iface=PCM,name='Playback Channel Map',device=9
numid=30,iface=PCM,name='ELD',device=10
numid=35,iface=PCM,name='Playback Channel Map',device=10

Thanks,

Robert



 Original Message 
> From: Alan Corey 
> Date: Sun, 29 May 2022 02:30:50 -0400
> Message-ID: 
> 
> Subject: Re: [Alsa-user] any hope for alsamixer controls for AMD RX 
> 5500/5500M GPU?
> To: rm.ric...@jacob21819.net
> Cc: ALSA-User 


> Use amixer, not alsamixer at first.  It lets you see devices a different
> way.
> 
> I remember at least 1 device where something started numbering at 0 and
> something else at 1.  If it starts at 1, 0 will be invalid, but you can try
> going the other way.  Compare going of

[Alsa-user] any hope for alsamixer controls for AMD RX 5500/5500M GPU?

[Robert M. Riches Jr.]

Is there any hope of getting alsamixer controls for an AMD RX
5500/5500M GPU?

I'm running Devuan Chimaera with kernel 5.10.0-14-amd64
and the version numbers on most "*alsa*" packages between
1.2.4-1 and 1.2.4-2.  I am _NOT_ running PulseAudio.

My GPU shows up to "lspci -nn" as this:

0b:00.0 VGA compatible controller [0300]: Advanced Micro Devices, Inc. 
[AMD/ATI] Navi 14 [Radeon RX 5500/5500M / Pro 5500M] [1002:7340] (rev c5)

After switching from DisplayPort-to-DVI to DisplayPort-to-HDMI
to drive the visual side of the monitor, I had to switch to
sending sound output through the GPU rather than the motherboard
audio device.  Now, alsamixer does not show a volume control.

This is my current .asoundrc file:

v cut here v
pcm.Generic { type hw; card Generic; }
ctl.Generic { type hw; card Generic; }
pcm.HDMI { type hw; card HDMI device 8; }
ctl.HDMI { type hw; card HDMI; }
pcm.!default {
  type plug
  slave.pcm "HDMI"
}
ctl.!default ctl.HDMI
^ cut here ^

If I add the "device 8" part to the ctl.HDMI line, alsamixer
refuses to run:
cannot open mixer: Invalid argument

Is there any hope of getting alsamixer to show a volume control
for that audio device?  Or, is the hardware design such that the
volume control and so forth would need to be done in software
(as in PulseAudio)?  (If the latter, I'll probably have to plug
in a set of separate speakers and go back to the motherboard's
audio device.)

Thanks,

Robert


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

[Robert M. Riches Jr.]

For temporary purposes and/or a manual workaround, you could
create a symlink:

cd /dev/snd
ln -s ./-midiC0D0 ./-midiC1D0

The "./" prefix is to force the command parser in the 'ln'
executable to interpret the names as arguments rather than
options.

At the very least, that should tell you whether that is _ALL_
of the problem.

HTH

Robert



> From: Fernando Carello 
> Date: Sun, 31 Jan 2021 13:20:19 +0100
> To: Clemens Ladisch 
> Cc: alsa-user@lists.sourceforge.net
> 
> So, I've seen that a working MIDI keyboard "creates" this device:
> 
> /dev/snd/-midiC0D0
> 
> while my problematic MIDI keyboard has instead:
> 
> /dev/snd/-midiC1D0
> 
> see that "C1"?
> This has to do with "amidi --dump" requiring "--port hw:1" instead of
> the usual "--port hw:0"
> 
> Now:
> How can I configure ALSA (.asoundrc?) in order to correctly "map" that
> "port hw:1", so it can be used by a MIDI softsynth, which expects
> "hw:0"?
> 
> Thanks!
> 
> 
> Il giorno sab 30 gen 2021 alle ore 18:54 Fernando Carello
>  ha scritto:
> >
> > > > No events with asqedump:  :-(
> > >
> > > Try with "amidi --dump --port hw:1".
> >
> > This works!
> >
> > amidi --dump --port hw:1
> >
> > 90 42 17
> > 80 42 00
> > 90 3D 19
> > 90 3F 0C
> >
> > Thanks a lot!
> >
> > Now, how should I configure ALSA in order for my softsynth
> > (ZynAddSubFx) to "see" this keyboard?
> >
> > > Do other USB MIDI devices (or USB devices) work?
> >
> > Yes, other MIDI USB keyboards (for example Akai LPK25) work fine;
> > "aseqdump" shows MIDI messages and the softsynth plays without issues.
> 
> 
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Re: [Alsa-user] How to record audio from a live video (youtube via firefox) ?

[Robert M. Riches Jr.]

(Apologies if you want reply to list without also a direct reply.
My mail client has 'reply' and 'Reply' only.)

I am also using pure ALSA.  After a little web searching a few
days ago, I came up with this .asoundrc file that records to a
raw file:

v cut here v
pcm.Intel { type hw; card Intel; }
ctl.Intel { type hw; card Intel; }
pcm.NVidia { type hw; card NVidia; }
ctl.NVidia { type hw; card NVidia; }

pcm.rate48000Hz {
  type rate
  slave {
pcm writeFile # Direct to the plugin which will write to a file
format S16_LE
# channels 2
rate 48000
  }
#route_policy copy
}

pcm.writeFile {
  type file
  slave {
pcm "Intel" # Now write to the actual sound card
  }
  file "aplay-D_card0-t_raw-f_S16_LE-r48000-c_2.raw"
  format "raw"
}

pcm.!default {
  type plug
  slave.pcm "rate48000Hz"
}
ctl.!default ctl.Intel
^ cut here ^

Caution: Some web browsers take liberties with how they deal with
sound.  For example, some years ago, Firefox Hello (video chat)
would work only with very specific .asoundrc content.

Changing the sample rate may or may not work.  With Musescore, I
had to use exactly 48000, not 44100.

Here are example commands for converting to WAV and then to MP3:

Convert it to WAV:

sox -r 48k -e signed -b 16 -c 2 \
aplay-D_card0-t_raw-f_S16_LE-r48000-c_2.raw \
-r 44100 something.wav

Crop it as needed.

Optional: Convert it to MP3:

ffmpeg -i something.wav something.mp3

HTH

Robert



> Date: Sun, 27 Dec 2020 10:25:14 +0100
> From: tu...@posteo.de
> To: alsa-user@lists.sourceforge.net
> 
> Hi,
> 
>  I want to record the audio of a live stream video (youtube).
> 
>  Everything I tried resulted in audio files with constant or
>  intermitted sine wave like sounds.
> 
>  I am using pure alsa. I don't want pulseaudio and with jack
>  started firefoxs audio does not work.
> 
>  How can I accomplish this recording task successfully ?
> 
>  My setup:
> 
>  GENTOO Linux
>  External USB audio DAC (FIIO Olympus 2)
> 
>  No $HOME/.asoundrc
>  /etc/conf.d/alsasound:
> 
># RESTORE_ON_START:
># Do you want to restore your mixer settings?  If not, your cards will be
># muted.
># no - Do not restore state
># yes - Restore state
>
>RESTORE_ON_START="yes"
>
># SAVE_ON_STOP:
># Do you want to save changes made to your mixer volumes when alsasound
># stops? 
># no - Do not save state
># yes - Save state
>
>SAVE_ON_STOP="yes"
> 
>  arecord -l:
> 
>  List of CAPTURE Hardware Devices 
> card 0: Audio [DigiHug USB Audio], device 0: USB Audio [USB Audio]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
> 
> 
>  arecord -L:
> null
> Discard all samples (playback) or generate zero samples (capture)
> default
> Default Audio Device
> sysdefault
> Default Audio Device
> lavrate
> Rate Converter Plugin Using Libav/FFmpeg Library
> upmix
> Plugin for channel upmix (4,6,8)
> vdownmix
> Plugin for channel downmix (stereo) with a simple spacialization
> default:CARD=Audio
> DigiHug USB Audio, USB Audio
> Default Audio Device
> sysdefault:CARD=Audio
> DigiHug USB Audio, USB Audio
> Default Audio Device
> front:CARD=Audio,DEV=0
> DigiHug USB Audio, USB Audio
> Front output / input
> usbstream:CARD=Audio
> DigiHug USB Audio
> USB Stream Output
> 
> aplay -l:
>  List of PLAYBACK Hardware Devices 
> card 0: Audio [DigiHug USB Audio], device 0: USB Audio [USB Audio]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
> card 0: Audio [DigiHug USB Audio], device 1: USB Audio [USB Audio #1]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
> 
> lsusb -v (excerpt):
> 
> Bus 005 Device 002: ID 1852:7022 GYROCOM C Co., LTD Fiio E10
> Device Descriptor:
>   bLength18
>   bDescriptorType 1
>   bcdUSB   1.10
>   bDeviceClass0 
>   bDeviceSubClass 0 
>   bDeviceProtocol 0 
>   bMaxPacketSize0 8
>   idVendor   0x1852 GYROCOM C Co., LTD
>   idProduct  0x7022 Fiio E10
>   bcdDevice0.01
>   iManufacturer   1 FiiO
>   iProduct2 DigiHug USB Audio
>   iSerial 0 
>   bNumConfigurations  1
>   Configuration Descriptor:
> bLength 9
> bDescriptorType 2
> wTotalLength   0x0182
> bNumInterfaces  4
> bConfigurationValue 1
> iConfiguration  0 
> bmAttributes 0x80
>   (Bus Powered)
> MaxPower  500mA
> Interface Descriptor:
>   bLength 9
>   bDescriptorType 4
>   bInterfaceNumber0
>   bAlternateSetting   0
>   bNumEndpoints   1
>   bInterfaceClass 3 Human Interface Device
>   bInterfaceSubClass  0 
>   

Re: [Alsa-user] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?

[Robert M. Riches Jr.]

Have you experimented with different content in your .asoundrc
file?  I don't know about Google Meet and Chromium specifically,
but other setup (the old Firefox Hello, for example) required a
specific .asoundrc to make it work.  IIRC, this is it:

defaults.pcm.card 0
defaults.ctl.card 0

HTH

Robert



> Date: Sun, 26 Jul 2020 17:43:12 +1000
> From: Philip Rhoades 
> To: ALSA user 
> Reply-To: p...@pricom.com.au
> 
> People,
> 
> I am not sure what is going on - I seem to have had increased sound 
> problems on recent versions of Fedora (30-31).  I have been routinely 
> uninstalling PA for some years and haven't had more than the usual 
> number of problems with audio in that time.  Currently:
> 
> alsa-lib-1.2.1.2-4.fc31.i686
> alsa-lib-1.2.1.2-4.fc31.x86_64
> alsa-ucm-1.2.1.2-4.fc31.noarch
> alsa-utils-1.2.1-3.fc31.x86_64
> alsamixergui-0.9.0-0.29.rc2.fc31.x86_64
> qemu-audio-alsa-4.1.1-1.fc31.x86_64
> wine-alsa-5.0-1.fc31.i686
> wine-alsa-5.0-1.fc31.x86_64
> 
> http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7
> 
> If I use the analog rear mic instead of "Default" on GoogleMeet people 
> can hear me but I can't hear them - but the "Speakers" option can't be 
> changed from "Default".
> 
> Playing YT videos is fine and I can record from my mic here:
> 
>https://online-voice-recorder.com
> 
> Any suggestions about debugging would be greatly appreciated.
> 
> Regards,
> 
> Phil.
> -- 
> Philip Rhoades
> 
> PO Box 896
> Cowra  NSW  2794
> Australia
> E-mail:  p...@pricom.com.au
> 
> 
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Re: [Alsa-user] only 1.00 dB of playback volume range

[Robert M. Riches Jr.]

What is the path or URL to that list?  On sourceforge, I see
alsa-announce, alsa-cvslog, and alsa-user, but no alsa-dev*.

Thanks,

Robert



> Date: Mon, 31 Dec 2018 22:48:50 -0800
> From: frede...@ofb.net
> To: "Robert M. Riches Jr." 
> Cc: alsa-user@lists.sourceforge.net
> Reply-To: frede...@ofb.net
> 
> You probably want to send this to the ALSA developer list, this one is
> not very active.
> 
> On Mon, Dec 31, 2018 at 09:05:59PM -0800, Robert M. Riches Jr. wrote:
> >With a new OS installation on a pre-existing machine, some of the
> >USB audio devices now have only 1.00 dB of playback volume
> >adjustment, which of course is not enough to be terribly useful.
> >
> >The affected USB devices show up as "Audio Advantage MicroII" in the
> >output of "aplay -L".  From LSB, they show up as these:
> >
> >Bus 001 Device 004: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ 
> >Audio Controller
> >Bus 001 Device 003: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ 
> >Audio Controller
> >
> >I just installed Devuan Ascii on the machine.  It's running kernel
> >4.9.0-8-amd64 and alsa-utils version 1.1.3-1.  Doing "amixer -c 1
> >contents" or "amixer -c 2 contents" shows this for playback volume:
> >
> >numid=3,iface=MIXER,name='PCM Playback Volume'
> >  ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0
> >: values=0,0
> >  | dBminmax-min=-1.00dB,max=0.00dB
> >
> >and
> >
> >numid=3,iface=MIXER,name='PCM Playback Volume'
> >  ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0
> >: values=77,77
> >  | dBminmax-min=-1.00dB,max=0.00dB
> >
> >Using alsamixer to run the controls from end to end, I cannot
> >perceive any difference in volume level.
> >
> >In contrast, the other device I use (either HDA Intel or HDA NVidia)
> >has around a 60dB range according to amixer, and running the control
> >in alsamixer from end to end has a huge range of gain.
> >
> >Previously, the exact same hardware was running Debian 7/Wheezy and
> >then Slackware 14.2.  With both those operating systems, the USB
> >sound devices had a very adequate volume control range, certainly a
> >whole lot more than 1dB.
> >
> >Being as the USB sound devices are the ones my wife uses, I'd be
> >extremely grateful if somebody could point me a to way (if one
> >exists) to a usefully greater range of volume control.  Any
> >suggestions?
> >
> >Thanks,
> >
> >Robert
> >
> >
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> >https://lists.sourceforge.net/lists/listinfo/alsa-user


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[Alsa-user] only 1.00 dB of playback volume range

[Robert M. Riches Jr.]

With a new OS installation on a pre-existing machine, some of the
USB audio devices now have only 1.00 dB of playback volume
adjustment, which of course is not enough to be terribly useful.

The affected USB devices show up as "Audio Advantage MicroII" in the
output of "aplay -L".  From LSB, they show up as these:

Bus 001 Device 004: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio 
Controller
Bus 001 Device 003: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio 
Controller

I just installed Devuan Ascii on the machine.  It's running kernel
4.9.0-8-amd64 and alsa-utils version 1.1.3-1.  Doing "amixer -c 1
contents" or "amixer -c 2 contents" shows this for playback volume:

numid=3,iface=MIXER,name='PCM Playback Volume'
  ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0
: values=0,0
  | dBminmax-min=-1.00dB,max=0.00dB

and 

numid=3,iface=MIXER,name='PCM Playback Volume'
  ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0
: values=77,77
  | dBminmax-min=-1.00dB,max=0.00dB

Using alsamixer to run the controls from end to end, I cannot
perceive any difference in volume level.

In contrast, the other device I use (either HDA Intel or HDA NVidia)
has around a 60dB range according to amixer, and running the control
in alsamixer from end to end has a huge range of gain.

Previously, the exact same hardware was running Debian 7/Wheezy and
then Slackware 14.2.  With both those operating systems, the USB
sound devices had a very adequate volume control range, certainly a
whole lot more than 1dB.

Being as the USB sound devices are the ones my wife uses, I'd be
extremely grateful if somebody could point me a to way (if one
exists) to a usefully greater range of volume control.  Any
suggestions?

Thanks,

Robert


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Re: [Alsa-user] Stub for libpulse

[Robert M. Riches Jr.]

> Date: Sun, 19 Mar 2017 12:36:46 +0100
> From: Nicolas George 
> To: alsa-user@lists.sourceforge.net
> Subject: [Alsa-user] Stub for libpulse
>
>
>
> Hi.
>
> It seems that the Mozilla people have had the brilliant idea, starting
^
> with Firefox 52, to disable ALSA by default and only support PULSE.
> Apparently, they intend to remove the support for raw ALSA completely.
>
> Well, I do not want the PULSE server anywhere near my systems, and I
> believe many people here think the same.
>
> Does anyone know of a libpulse stub that provides the basic API but
> routes all the calls directly to libasound / ALSA?
>
> I have found this project:
> https://github.com/i-rinat/apulse
> but it seems unable to get something as simple as ogg123 working.
>
> Regards,
>

With due respect, I think you misspelled "terrible".  :-/

Thanks for the pointer to apulse.  I'll keep that around.

Have you considered using Firefox-ESR instead of plain Firefox?
Debian 7/Wheezy currently has Firefox-ESR 45.8.0.  One nice thing
about ESR is the UI doesn't get scrambled on a weekly basis, so
dialog buttons don't exchange positions on a whim.

Have you considered this might signal it's time to use a
different browser?


You're definitely _NOT_ alone in not wanting Pulse.  (Also,
please don't get me started on systemd.  :-(

In reality, Firefox support for plain ALSA has been buggy for
some time.  With the "Firefox Hello" thing they had a year or so
ago and apparently dropped, it simply did not work with a
.asoundrc file if the file's contents were not tremendously
simple and stated in precisely the syntax Firefox wanted.  Also,
I have a thin/zero-client setup at home, with .asoundrc tailored
to push sound to the appropriate audio device for each session.
Quite often, Firefox misdirects the sound from my wife's browser
session to my monitor.  That and a few other things are raising
my level of unhappiness toward Firefox, approaching the threshold
to look for a different browser.


Thanks.

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Re: [Alsa-user] no alsa-base.conf

[Robert M. Riches Jr.]

On Debian 7/Wheezy, I see this file:

/etc/modprobe.d/alsa-base.conf

Provided you have the locate or mlocate or slocate command
installed, you can do this (adding the leading 'm' or 's' if
needed) to see what path(s) might be on your system:

locate alsa-base.conf

If all else fails, inform us your distribution and release so
someone who knows that distribution can give a more precise
answer.

HTH

Robert


> From: Kristoffer Gustafsson 
> Date: Tue, 1 Nov 2016 02:39:04 +0100
> To: alsa-user@lists.sourceforge.net
>
> hi.
> I need to add things to the alsa-base.conf.
> but I can't find such a file.
> I want to add the option to ignore ctl error when using my xonar u5.
> what shall I edit instead?
> /Kristoffer
>
> -- 
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> Salängsgatan 7a
> tel:033-12 60 93
> mobil: 0730-500934

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Re: [Alsa-user] X insists on using card1 instead of card0 [solved]

[Robert M. Riches Jr.]

> To: alsa-user@lists.sourceforge.net
> From: Felix Miata 
> Date: Sun, 22 May 2016 15:26:05 -0400
>
> Clemens Ladisch composed on 2016-05-22 15:23 (UTC+0200):
>
> Thanks for your helpful reply!
>
> > Felix Miata wrote:
>
> >> I was able to get Xorg sounds working
>
> > X.org is a graphics system; it does not handle sound.
>
> Xorg was used as shorthand for apps that only run within an Xorg environment, 
> e.g. DE system sounds and device notifications, HTML5 videos in web browsers, 
> SMplayer and so forth.
>
> > Are you talking about PulseAudio?  Did you try running pavucontrol?
>
> Since posting here I was able to get most sounds to work without pulseaudio 
> installed. The only holdout remained HTML5 in Firefox, even after going 
> through Mozilla.org's and Mozillazine.org's troubleshooting steps WRT this 
> exact problem. I also commented in an existing bug:
>
> https://bugzilla.mozilla.org/show_bug.cgi?id=803042
>
> The key to getting sound to work turned out to be /etc/asound.conf, which 
> hadn't previously existed, containing 'pcm.!default   "hdmi:0,0"'.

Felix, thank you for referencing that Firefox bug report.  I saw
I had not voted for it--just now fixed that.  If you'd like some
attention to be paid to fixing the Firefox bug, you might consider
voting for it.  (However, don't expect it.  IME, platform-specific
bugs don't get fixed--ever.)

One alternative that IIUC _should_ work is to use ~/.asoundrc for
a per-user workaround rather than /etc/asound.conf for an all-user
workaround.

Sorry, but I don't know anything that would likely help system
sounds for a "desktop environment".

Thanks,

Robert Riches

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Re: [Alsa-user] snd-aloop with multiple identical sound cards

[Robert M. Riches Jr.]

> From: José Luis Artuch <art...@speedy.com.ar>
> To: "Robert M. Riches Jr." <rm.ric...@jacob21819.net>
> Cc: alsa-user@lists.sourceforge.net
> Date: Fri, 29 Apr 2016 23:55:56 -0300
>
> El vie, 29-04-2016 a las 17:26 -0700, Robert M. Riches Jr. escribió:
> > The module's options are likely at least a start toward the
> > solution.  Here's an anonymized version of what I had in
> > /etc/modprobe.conf when I used the loopback soundcard:
> > 
> > options snd_aloop enable=1,1,1,1,1,1,1 index=4,5,6,7,8,9,10 
> > id=lbuser1,lbuser2,lbuser3,lbuser4,lbuser5,lbuser6,lbuser7
> > 
> > Depending on your distribution, you might need to put the options
> > in a different path/file.
> > 
> > Fwiw, I found a problem with Debian 7/Wheezy's snd_aloop in
> > connection with [Net]Jack.  The virtual soundcard would accept
> > playback samples much faster than a real soundcard would have,
> > which resulted in xruns.
> > 
> > HTH
> > 
> > Robert
> > 
> Thank you very much Robert. The OS is Debian GNU/Linux 8.
> I loaded the data from modprobe:
> modprobe snd-aloop enable=1,1,1,1 index=10,11,12,13
> id=lbuser1,lbuser2,lbuser3,lbuser4
> With cat /proc/asound/cards I see that lbuser1 replaces Loopback,
> lbuser2 replaces Loopback_1, lbuser3 replaces Loopback_2, lbuser4
> replaces Lopback_3.
> Now the question is if at every startup can be assigned always lbuser1
> to card_0, lbuser2 to card_1, lbuser3 to card_2, lbuser4 to card_3, for
> example editing the asound.conf file. I want to capture the sound played
> in each real card, always with the same virtual card.
> I do not experienced enough with snd-aloop on Debian 7, also I'm always
> doing these tests without graphical environment.
> Greetings.
> José Luis
> > 
> > > From: José Luis Artuch <art...@speedy.com.ar>
> > > To: alsa-user@lists.sourceforge.net
> > > Date: Fri, 29 Apr 2016 16:49:07 -0300
> > >
> > > Hi,
> > > I fixed the names of each real sound card (card_0, card_1, card_2, ...).
> > > Now, loading the snd-aloop module for all real sound cards, virtual
> > > sound cards Loopback (Loopback, Loopback_1, Loopback_2, ...) are
> > > created. How I can fix always the same virtual sound card Loopback for
> > > each real sound card ?.
> > > Thanks.
> > > José Luis
> > >
> > >
> > > --
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> > > Alsa-user@lists.sourceforge.net
> > > https://lists.sourceforge.net/lists/listinfo/alsa-user

(Sorry the quoting got scrambled.  With newsgroups, I understand
top-posting is always frowned upon.  With email, it's not quite
as clear.)

That's the intent and purpose of those options, to let you force
them to be consistent from one boot to the next.  At least it's
supposed to work that way if you put the options in
/etc/modprobe.conf (or equivalent).  The names can be whatever
you want; there is nothing special about the string "lbuser${n}".
When I used snd_aloop, I used the names of real users so the
users could refer to them in their .asoundrc files.  In my
experience, it was always consistent.  Sound should work the same
with or without a graphical environment.

HTH

Robert

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Re: [Alsa-user] snd-aloop with multiple identical sound cards

[Robert M. Riches Jr.]

The module's options are likely at least a start toward the
solution.  Here's an anonymized version of what I had in
/etc/modprobe.conf when I used the loopback soundcard:

options snd_aloop enable=1,1,1,1,1,1,1 index=4,5,6,7,8,9,10 
id=lbuser1,lbuser2,lbuser3,lbuser4,lbuser5,lbuser6,lbuser7

Depending on your distribution, you might need to put the options
in a different path/file.

Fwiw, I found a problem with Debian 7/Wheezy's snd_aloop in
connection with [Net]Jack.  The virtual soundcard would accept
playback samples much faster than a real soundcard would have,
which resulted in xruns.

HTH

Robert


> From: José Luis Artuch 
> To: alsa-user@lists.sourceforge.net
> Date: Fri, 29 Apr 2016 16:49:07 -0300
>
> Hi,
> I fixed the names of each real sound card (card_0, card_1, card_2, ...).
> Now, loading the snd-aloop module for all real sound cards, virtual
> sound cards Loopback (Loopback, Loopback_1, Loopback_2, ...) are
> created. How I can fix always the same virtual sound card Loopback for
> each real sound card ?.
> Thanks.
> José Luis
>
>
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> Applications Manager provides deep performance insights into multiple tiers of
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Re: [Alsa-user] A long shot I know: recording from a POTS phone for voicemail

[Robert M. Riches Jr.]

> Date: Tue, 23 Feb 2016 19:06:44 -0500
> From: doug 
> To: p...@pricom.com.au, alsa-user@lists.sourceforge.net
>
>
> On 02/23/2016 03:10 PM, Philip Rhoades wrote:
> > People,
> >
> > I know this is a bit of a long shot but does anyone here have any
> > experience setting up a voicemail recording system for a POTS phone?  I
> > thought it should be simpler than using say using Asterisk to set up a
> > whole PABX but maybe it isn't . .
> >
> > Yhanks,
> >
> > Phil.
> It depends on what you are trying to do. If you just want to record the 
> audio on a telephone line, the easiest way is
> a small audio transformer and a capacitor. Say a 1 Ohm 1:1 audio 
> transformer and about a 0.1 microfarad capacitor
> rated at 100 volts DC or better in series with the transformer on the 
> phone line side. Connect the secondary to the recorder
> or computer line input.
>
> If you're trying to run voice in both directions from, say, a computer 
> to the phone line, you need a telephone modem, which
> is explicitly designed to do that job.

If you use the transformer and capacitor solution, you'll also
need something to provide a DC path across the phone line or the
phone company switch will hang up the call.  Here's a schematic I
posted to another forum:

https://tech.lds.org/forum/download/file.php?id=2636

If maximum economy is required, especially for a one-time event,
you can use a 1K resister instead of the choke and drop the MOV,
Zener diodes, and R1.

HTH

Robert

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[Alsa-user] fwiw: solution to silent Vimeo in Firefox

[Robert M. Riches Jr.]

In case it might be helpful to anyone else now or in the future,
here's a solution to a problem:

On my Debian 7/Wheezy system, one user could play a Vimeo piece
with sound, but another user got silence for the same piece.  The
second user also had silence for some Google logos that allegedly
had sound for other people.  The key difference was the use of
the 'plug' feature in the users' .asoundrc files.  The following
diff/patch got sound working for the second user.  My guess is
the Vimeo player produces a specific sample rate (or perhaps
other format attribute) and adding the 'plug' feature causes ALSA
to do the necessary conversion.

--- ../../.asoundrc.~1~ 2011-08-19 21:23:33.894407686 -0700
+++ ../../.asoundrc 2016-01-03 15:15:12.271767959 -0800
@@ -1,6 +1,9 @@
 pcm.Intel { type hw; card Intel; }
 ctl.Intel { type hw; card Intel; }
 pcm.NVidia { type hw; card NVidia; }
 ctl.NVidia { type hw; card NVidia; }
-pcm.!default pcm.Intel
+pcm.!default {
+  type plug
+  slave.pcm "Intel"
+}
 ctl.!default ctl.Intel

HTH

Robert

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

[Robert M. Riches Jr.]

 Date: Sun, 23 Aug 2015 19:11:17 +0200
 From: Gunnar Arndt madenhac...@gmail.com
 To: alsa-user@lists.sourceforge.net

 Hi Alsa users,


 I have the following idea, which may be of interest for other users, 
 too: I would like to abuse the 'normal' unbalanced stereo input of my 
 on-board sound as a balanced mono input.
 I have a measurement microphone with an integrated amplifier and a 
 balanced mono output which should be connected to consumer sound devices 
 with an `ordinary' unbalanced input (I have added a very short 
 explanation on the term `balanced' to the post scriptum).

 My equipment:

 * Ubuntu 14.04.3 with latest updates, including Kernel 3.19 and ALSA 1.0.25.
 * HD-Audio-based Realtek AL1150 codec on an Asrock X99 Extreme4 
 mainboard - with unbalanced stereo input, of course.
 * Earthworks M30BX microphone with a balanced mono output (an XLR plug).
 * A DMX out cable (XLR to 3.5mm stereo) connects the mic correctly to 
 the line input of the sound card: Ground to ground, balanced 
 non-inverting to unbalanced left, balanced inverting to unbalanced right.
 ...

In the analog domain, one issue will be the signal levels.  Line
level is usually a few hundred millivolts.  Mike level is usually
on the order of a few millivolts.  When going through a stereo
line input, you might not have enough gain.  There is also some
likelihood that the line input's front-end's noise floor might be
too high for your mike signals.

Another analog domain issue is that noise and other common mode
excursions might be of greater amplitude than signal and might be
larger than your input channel can handle.  A true balanced input
channel, perhaps with a transformer front end, would be more
likely to handle larger common mode noise than desired
differential signal.

The cheapest analog hardware method to convert from balanced to
unbalanced requires two conditions: 1) the balanced output must
come from a transformer (coil of wire on a ferrous core); _AND_
2) you are willing to sacrifice a little noise floor in exchange
for economy.  That solution is to just ground one of the balanced
wires and use the other as signal.

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.

For the digital domain solution, if your use case is
non-real-time, you could record a stereo signal into a 2-channel
WAV file and then post-process it into a mono WAV file by taking
the difference.  Sox might have a filter for that.  Otherwise, a
C program wouldn't take long to write to do that.  An alternative
to saving the (noisy) stereo file would be to output RAW samples
and do the conversion using Linux pipes--but that would probably
introduce considerable latency, depending on buffer sizes.

HTH

Robert

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Re: [Alsa-user] starters help (help!)

[Robert M. Riches Jr.]

There are two levels to make a module available:

First, the module needs to be compiled for your kernel and
available in (usually, IME) /lib/modules...  The distributions I
have worked with put all modules they compile in the kernel
packages.  Well, Tiny Core has some modules in separate packages.
I don't know how Mint packages modules.  (I'm assuming I remember
correctly that you're using Mint.)  It might be possible your
kernel does not have that module compiled.  If the module isn't
compiled for your kernel, that's beyond my ability to instruct
for a distribution I have not used.

Second, the module needs to be loaded into the running kernel.
That is done by the modprobe command.

HTH

Robert


 Date: Mon, 17 Aug 2015 20:41:08 +0200
 From: F. Dols f.j.h.d...@gmail.com
 To: Robert M. Riches Jr. rm.ric...@jacob21819.net, 
  alsa-user@lists.sourceforge.net

 Hi,

 sudo modinfo indigodjx

 gives:

 modinfo: ERROR: Module indigodjx not found.

 So, how do I proceed to install that module?

 thanks,

 F


 On 08/16/2015 11:30 PM, Robert M. Riches Jr. wrote:
  In another reply, you posted that sudo aplay -L did not show
  the Indigodjx card.
 
  I wonder if the udev rule might be missing, which IIUC could
  cause the module to not load, which IIUC could cause the card to
  not be visible.
 
  Is the INDIGODJX kernel module loaded?
 
   sudo lsmod | grep indigodjx
 
  You might unplug the card from USB and then do one of these
 
   sudo tail -f /var/log/messages
 
   sudo tail -f /var/log/syslog
 
  (whichever works on your system) while plugging the card back in
  to USB.  That should tell what udev is doing when the card is
  plugged in.  You might also do something similar to this to see
  what the kernel says is going on:
 
   sudo dmesg | tail -44
 
  This should tell you if you have any udev rules for your card:
 
   grep -ri indigodjx /etc/udev /lib/udev
 
  That recursive grep would have shown any file in or under those
  paths that mentioned indigodjx.  On my Debian 7 system, it does
  now return anything.  Debian 7 is pretty old, so perhaps too old
  to have a udev rule for the card.
 
  To see if the kernel module can see your card, you might try
  this:
 
   sudo modprobe indigodjx
 
  Also, this indicates there's an enable parameter for the module:
 
   sudo modinfo indigodjx
 
  IME, sound cards normally default to enabled, but perhaps this
  driver is different.  You might need to manually force it to be
  enabled.
 
  HTH
 
  Robert
 
 
  Date: Sun, 16 Aug 2015 20:14:10 +0200
  From: F. Dols f.j.h.d...@gmail.com
  To: alsa-user@lists.sourceforge.net
 
  Hi,
 
  I need some starting pointers for the following.
 
  A.
  I use Mint 17.1 and bought the Indigodjx soundcard.
 
  On
  http://www.alsa-project.org/main/index.php/Matrix:Main
 
  I do not see the name ECHO nor INDIGO  in the Sound Card List.
 
  However, I do see the name INDIGODJX in the DRIVERS list.
 
  What does this mean? Is ALSA able to see this soundcard under Mint 17.1
  or not?
 
 
  B.
  I ALSA can see my soundcard, I would like to have some steps for making
  it possible.
 
  I have ALSA running: My Mixxx shows ALSA and uses ALSA for the build-in
  soundcard that I want to replace with the Indigodjx.
 
  I do understand what ALSA is, but I have as yet no intention to play
  with soundcard settings.
 
  My only thing is to have the Indigodjx recognised and used by Mixxx and
  Banshee.
 
 
  HOW DO I PROCEED? ( I am willing to make a intro doc based on my
  experiences, but as for now, I have to even know how to start..
 
 
 
 
 
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Re: [Alsa-user] starters help (help!)

[Robert M. Riches Jr.]

In another reply, you posted that sudo aplay -L did not show
the Indigodjx card.

I wonder if the udev rule might be missing, which IIUC could
cause the module to not load, which IIUC could cause the card to
not be visible.

Is the INDIGODJX kernel module loaded?

sudo lsmod | grep indigodjx

You might unplug the card from USB and then do one of these

sudo tail -f /var/log/messages

sudo tail -f /var/log/syslog

(whichever works on your system) while plugging the card back in
to USB.  That should tell what udev is doing when the card is
plugged in.  You might also do something similar to this to see
what the kernel says is going on:

sudo dmesg | tail -44

This should tell you if you have any udev rules for your card:

grep -ri indigodjx /etc/udev /lib/udev

That recursive grep would have shown any file in or under those
paths that mentioned indigodjx.  On my Debian 7 system, it does
now return anything.  Debian 7 is pretty old, so perhaps too old
to have a udev rule for the card.

To see if the kernel module can see your card, you might try
this:

sudo modprobe indigodjx

Also, this indicates there's an enable parameter for the module:

sudo modinfo indigodjx

IME, sound cards normally default to enabled, but perhaps this
driver is different.  You might need to manually force it to be
enabled.

HTH

Robert


 Date: Sun, 16 Aug 2015 20:14:10 +0200
 From: F. Dols f.j.h.d...@gmail.com
 To: alsa-user@lists.sourceforge.net

 Hi,

 I need some starting pointers for the following.

 A.
 I use Mint 17.1 and bought the Indigodjx soundcard.

 On
 http://www.alsa-project.org/main/index.php/Matrix:Main

 I do not see the name ECHO nor INDIGO  in the Sound Card List.

 However, I do see the name INDIGODJX in the DRIVERS list.

 What does this mean? Is ALSA able to see this soundcard under Mint 17.1 
 or not?


 B.
 I ALSA can see my soundcard, I would like to have some steps for making 
 it possible.

 I have ALSA running: My Mixxx shows ALSA and uses ALSA for the build-in 
 soundcard that I want to replace with the Indigodjx.

 I do understand what ALSA is, but I have as yet no intention to play 
 with soundcard settings.

 My only thing is to have the Indigodjx recognised and used by Mixxx and 
 Banshee.


 HOW DO I PROCEED? ( I am willing to make a intro doc based on my 
 experiences, but as for now, I have to even know how to start..





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Re: [Alsa-user] over the net sound stream

[Robert M. Riches Jr.]

That would be a job for the arrangement I mentioned with the ALSA
loopback soundcard and NetJACK.  I used to do that with some
zero/thin diskless client machines, so I know it can be done.

Let's label your machines: Let's say you are running ALSA
playback clients on machine A and want the sound to be beamed to
machine B.  On machine A, you point .asoundrc to a loopback
soundcard so that all ALSA clients will play sound to the
loopback soundcard.  You run NetJACK to take the samples from the
loopback soundcard and route to NetJACK.  You configure NetJACK
to send the sound from machine A to machine B.  On machine B, you
configure NetJACK to send the sound to the desired hardware
soundcard.

In theory, you could use netcat aka nc and a couple of homegrown
ALSA clients as a substitute for NetJACK.  I don't know whether
it would take less effort to write the clients or set up NetJACK.

HTH

Robert


 From: daggs da...@gmx.com
 To: li...@lazygranch.com
 Date: Thu, 26 Mar 2015 08:01:40 +0100
 Cc: alsa-user@lists.sourceforge.net

 thanks for the info guys, but it isn't quite what I need. I need to stream 
 any sound from one computer to another.

 I wonder, I can define in ~/.asoundrc which card to stream the sound to, so 
 assuming that I have a listener on another computer, it isn't that hard to 
 define a remote ip in a similar manner and just dump the data we stream to 
 the card into the network instead.

 is there anything bad with it?

  Sent: Thursday, March 26, 2015 at 3:15 AM
  From: li...@lazygranch.com
  To: alsa-user@lists.sourceforge.net
  Subject: Re: [Alsa-user] over the net sound stream
 
  ???http://unix.stackexchange.com/questions/40058/pipe-system-sound-to-another-computer
  
  Skip down to the netcat post. I haven't tried this, but netcat is amazing 
  at piping stuff around. The post uses nc instead of netcat.
  
  
    Original Message  
  From: Robert M. Riches Jr.
  Sent: Wednesday, March 25, 2015 5:47 PM
  To: alsa-user@lists.sourceforge.net; da...@gmx.com
  Subject: Re: [Alsa-user] over the net sound stream
  
  There is a way to do it using the ALSA loopback soundcard and
  [Net]JACK. It's not easy. There are some tutorials on the web,
  though it's easy to get confused about whether the tutorials are
  talking about JACK1 vs. JACK2. IME, the ALSA loopback soundcard
  has a disadvantage that it lets the playback client get too far
  ahead, which can cause difficulties in some situations.
  
  HTH
  
  Robert
  
  
   From: daggs da...@gmx.com
   To: alsa-user@lists.sourceforge.net
   Date: Wed, 25 Mar 2015 20:25:54 +0100
  
   Greetings,
  
   is there a way to stream sound from one machine to another via a network 
   using alsa?
   in addition, what is the purpose of the aserver bin which is part of alsa 
   libs pkg?
  
   Thanks.

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Re: [Alsa-user] over the net sound stream

[Robert M. Riches Jr.]

There is a way to do it using the ALSA loopback soundcard and
[Net]JACK.  It's not easy.  There are some tutorials on the web,
though it's easy to get confused about whether the tutorials are
talking about JACK1 vs. JACK2.  IME, the ALSA loopback soundcard
has a disadvantage that it lets the playback client get too far
ahead, which can cause difficulties in some situations.

HTH

Robert


 From: daggs da...@gmx.com
 To: alsa-user@lists.sourceforge.net
 Date: Wed, 25 Mar 2015 20:25:54 +0100

 Greetings,

 is there a way to stream sound from one machine to another via a network 
 using alsa?
 in addition, what is the purpose of the aserver bin which is part of alsa 
 libs pkg?

 Thanks.

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[Alsa-user] second kernel panic involving snd_aloop on Debian Wheezy

[Robert M. Riches Jr.]

For the second time in a little over 90 days, my main system has
had a kernel panic in which snd_aloop is implicated.  The system
is using this kernel (latest stable for Wheezy):

Linux one 3.2.0-4-amd64 #1 SMP Debian 3.2.63-2+deb7u1 x86_64 GNU/Linux

ii  linux-image-3.2.0-4-amd64 3.2.63-2+deb7u1   
 amd64Linux 3.2 for 64-bit PCs

This time, one user was on a VNC X session using the ALSA
loopback soundcard and JACK to send sound to the thin/zero client
machine.  When I logged in to the console and started X on it,
the other user's sound playback in progress changed to sound like
a motorboat.  I saw that the alsa_in process's output file was
growing rapidly.  This time, I did not delete the file but killed
the alsa_in process using the default signal.  Immediately, the
console showed the kernel panic dump I captured with enough
pixels to easily read the text.  The snd_aloop module name is
visible in the middle of the image.

After two kernel panics in a little over 90 days, this one
causing /home's mdadm RAID to be resync'ed, I'm going to need to
abandon use of snd_aloop and JACK.

Has anyone else seen this type of thing?  If so, is there a any
less drastic measures to prevent the kernel panics?

Thanks,

Robert

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Re: [Alsa-user] Newbie to compile opl3 driver

[Robert M. Riches Jr.]

(second attempt, had forgotten to copy the list)

The driver is a kernel module, not usually a package.  Try

modprobe snd-opl3sa2

and/or

modprobe snd-opl3-lib

and/or

modprobe snd-opl3-synth

I don't have a kernel as recent as 3.7.10.  Debian Wheezy's
kernel 3.2.0-4 has snd-opl3-{lib,synth} kernel modules.  An old
remaster of TinyCore with kernel 3.0.3 has snd-opl3sa2,
snd-opl3-lib and snd-opl3-synth.

To see if kernel modules might differ with your kernel, do 'find
lib/modules -name '*opl3*.ko*'.  If no luck there, check your
kernel's config to see whether the OPL3 drivers were enabled.

Regarding compiling from source, is there a file with a name
similar to 'bootstrap'?  Sometimes that is a precursor to
./configure.  If there's an INSTALL or README* file, that often
contains compilation instructions.

HTH

Robert


 Date: Mon, 03 Nov 2014 03:19:19 +0100
 To: Advanced Linux Sound Architecture - User
   alsa-user@lists.sourceforge.net
 From: John Smith allesblo...@web.de

 Hello list,

 this Toshiba Tecra 8000 runs Antix Linux (Kernel 3.7.10-antix.5-486-smp  
 i686 (32 bit)) for a few days now. Neither does it produce any sound nor  
 does inxi -F mention any sound devices.
 The laptop is equipped with a Yamaha Opl3-sa2 chip. It works nicely under  
 the Windows98SE that is now to be replaced by Antix.

 apt-get install opl3-sa2 cannot find the requested package. So I followed
 http://www.alsa-project.org/main/index.php/Matrix:Module-opl3-sa2 and  
 downloaded the sources of alsa-driver, alsa-lib, alsa-firmware and  
 alsa-utils. But I got stuck when calling ./configure  
 --with-cards=opl3-sa2 --with-sequencer=yes ; make ; make install because  
 there is no executable configure.
 I got the sources again via git, checked out version 1.0.25 and still no  
 ./configure script.

 What am I doing wrong?

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[Alsa-user] anyone else see kernel panic when killing alsa_in?

[Robert M. Riches Jr.]

Has anyone else ever seen a kernel panic immediately after
killing an alsa_in process?

I'm running Debian Wheezy, x86_64.  A user's alsa_in process
had produced about 700MB of log.  I made the mistake of
removing the log file.  A couple of minutes later, I did
killall alsa_in as root.  (There was only the one alsa_in
process running.)  Immediately, the kernel panic message
and hex dumps filled the screen.

The alsa_in process was part of a lash-up using netjack
for a thin/zero-client setup.  I often kill alsa_in
processes that have gone runaway and produced a few GB of
log file, but never before have I made the mistake of
removing the log file before killing the process.

I plan to file a bug report on Debian because a user could
have done everything here, which makes this a potential
DOS vulnerability.

(Fortunately, the system came back up okay, though RAIDs
might need resyncing.)

Robert

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Re: [Alsa-user] Tinycore and Haswell produce only chopped noise

[Robert M. Riches Jr.]

(Oops.  Forgot to copy the list.)

Chris and list,

Thank you for the advice to check /proc/asound/Intel.  There's
undoubtedly good stuff in there--if my magic decoder ring wasn't
in the shop.  :-)

To be more clear, the scratchy, chopped white noise sound is all
I had ever heard from aplay.  It's not that it plays some modes
but not others; it doesn't play properly with any mode.  I reran
some tests, based on your advice.  Info interspersed below.

Thanks,

Robert


 Date: Fri, 29 Aug 2014 22:46:27 -0700
 From: chris hermansen clherman...@gmail.com
 To: Robert M. Riches Jr. rm.ric...@jacob21819.net
 Cc: Alsa-user@lists.sourceforge.net


 Robert and list,

 On Aug 29, 2014 10:08 PM, Robert M. Riches Jr. rm.ric...@jacob21819.net
 wrote:
 
  Trying to get sound working on TinyCore 5, 64-bit kernel, 32-bit
  userspace with a Haswell i5-4690 CPU and an Asus H97M-Plus
  motherboard.  My primary need is at least one channel of audio
  output from VLC.
 
  Kernel modules appear loaded.  aplay -l and aplay -L show one
  card, ALC887-VD.  Alsamixer appears to show an HDMI device with
  no controls and another device (ALC-887, I think) with controls.
  I unmuted and turned up the gain on master on that second device
  in order to get any sound output.  Oh, and with aplay and a WAV
  file, I have to use plughw:1, because hw:1 refused to do
  single channel.

 What kind of WAV file is it? What happens if you try a stereo 16 bit 44.1
 kHz WAV file?

File A, the file I had originally tried, is Signed 16 bit Little
Endian, Rate 48000 Hz, Mono.

The first time I play a file with CD rates and such (file B), I
hear silence, and toward the end of the file, I see this:

aplay: pcm_write:1939: write error: Input/output error

If I then play file A, I hear chopped pieces of file B.  Then, if
I play file B again, I hear chopped pieces of file A.  It seems
something in aplay or the driver is seriously confused.

 What happens if you use the default output device?

That results in a segmentation fault, every time.

 Why can't you use a newer kernel?

Kernel 3.8.13 is the latest available in the TinyCore
repositories.  To compile my own kernel for TinyCore would
require setting up a build environment, configuring the newer
kernel, etc.  That would take much too much time--probably at
least a few evenings.  A Sept. 14 hard deadline looms to have a
full VLC plus monitoring appliance.  A newer kernel would
guarantee a missed deadline.  A couple of soundcards are cheaper.
After sending this, I'm heading to Frys.

 If it works with stereo and you need a mono signal out, can you fabricate a
 different cable?

 
  With aplay on a WAV file, I get scratchy sounds (similar to
  pulsed white noise or someone scratching fingernails on a
  microphone screen) out of the headphone jack.

 It would be interesting to look at /proc/asound/Intel files while playing
 something that sounds scratchy. I have had scratchy sounds in the past when
 the driver didn't properly deal with the bit rate and depth.

/proc/asound/Intel is a symlink to /proc/asound/card1.  I'll snip
the original details and append the output of something similar
to this:

head -9 `find /proc/asound/card1 -type f`

 Also any related messages in the /var/log/syslog file?

Starting syslogd created /var/log/messages but not
/var/log/syslog.  Nothing showed up in /var/log/messages for
attempts with 'aplay -D plughw:1 ...'.  The only things that ever
showed up there were segmentation faults from trying to use the
default device.

This is the tail of dmesg output.  The line at 16.931012 is
probably irrelevant.  Nothing earlier in dmesg output appeared to
be relevant to ALSA.

[8.675200] input: HDA Intel Line as
/devices/pci:00/:00:1b.0/sound/card1/input6
[8.675334] input: HDA Intel Rear Mic as
/devices/pci:00/:00:1b.0/sound/card1/input7
[8.675450] input: HDA Intel Front Mic as
/devices/pci:00/:00:1b.0/sound/card1/input8
[8.675557] input: HDA Intel Front Headphone as
/devices/pci:00/:00:1b.0/sound/card1/input9
[8.675667] input: HDA Intel Line Out Side as
/devices/pci:00/:00:1b.0/sound/card1/input10
[8.675803] input: HDA Intel Line Out CLFE as
/devices/pci:00/:00:1b.0/sound/card1/input11
[8.675901] input: HDA Intel Line Out Surround as
/devices/pci:00/:00:1b.0/sound/card1/input12
[8.676067] input: HDA Intel Line Out Front as
/devices/pci:00/:00:1b.0/sound/card1/input13
[   16.931012] mtrr: no MTRR for e000,3ff found
[  248.663732] aplay[1073]: segfault at 8 ip f777302d sp
ffe8f914 error 4 in libasound.so.2.0.0[f7704000+a1000]
[  361.240347] aplay[1258]: segfault at 8 ip f76d002d sp
fff91f54 error 4 in libasound.so.2.0.0[f7661000+a1000]
[  363.685898] aplay[1262]: segfault at 8 ip f778d02d sp
ffd27364 error 4 in libasound.so.2.0.0[f771e000+a1000]

  Googling didn't find anything that appeared relevant.
 
  Where could I find

[Alsa-user] Tinycore and Haswell produce only chopped noise

[Robert M. Riches Jr.]

Trying to get sound working on TinyCore 5, 64-bit kernel, 32-bit
userspace with a Haswell i5-4690 CPU and an Asus H97M-Plus
motherboard.  My primary need is at least one channel of audio
output from VLC.

Kernel modules appear loaded.  aplay -l and aplay -L show one
card, ALC887-VD.  Alsamixer appears to show an HDMI device with
no controls and another device (ALC-887, I think) with controls.
I unmuted and turned up the gain on master on that second device
in order to get any sound output.  Oh, and with aplay and a WAV
file, I have to use plughw:1, because hw:1 refused to do
single channel.

With aplay on a WAV file, I get scratchy sounds (similar to
pulsed white noise or someone scratching fingernails on a
microphone screen) out of the headphone jack.

Googling didn't find anything that appeared relevant.

Where could I find whether that kernel (3.8.13) supports Haswell
HDMI audio and/or the H97M's audio?

Any pointers to tricks to maybe get the existing sound devices
working?

Buying another sound card is acceptable if necessary.  Any
suggestions for something  US$50 that is old enough that is sure
to work with the 3.8.13 kernel?  It will be feeding a church PA
system but studio-quality sound is not necessary.

Thanks in advance,

Robert


Details:


Output of uname -a:

Linux box 3.8.13-tinycore64 #777 SMP Fri Oct 18 15:13:45 UTC 2013 x86_64 
x86_64 x86_64 GNU/Linux

Output of aplay -l:

 List of PLAYBACK Hardware Devices 
card 1: Intel [HDA Intel], device 0: ALC887-VD Analog [ALC887-VD Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: Intel [HDA Intel], device 1: ALC887-VD Digital [ALC887-VD Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Output of aplay -L:

null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default:CARD=Intel
HDA Intel, ALC887-VD Analog
Default Audio Device
sysdefault:CARD=Intel
HDA Intel, ALC887-VD Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Intel,DEV=0
HDA Intel, ALC887-VD Digital
IEC958 (S/PDIF) Digital Audio Output

Output of lsmod:

Module  Size  Used byNot tainted
snd_seq_dummy  12288  0 
snd_seq_oss24576  0 
snd_seq_midi_event 12288  1 snd_seq_oss
snd_seq36864  5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 12288  3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss36864  0 
snd_mixer_oss  16384  1 snd_pcm_oss
snd_hda_codec_realtek49152  1 
snd_hda_codec_hdmi 28672  1 
snd_hda_intel  28672  0 
snd_hda_codec  65536  3 
snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel
snd_hwdep  12288  1 snd_hda_codec
snd_pcm57344  4 
snd_pcm_oss,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec
snd_timer  20480  2 snd_seq,snd_pcm
snd45056 12 
snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore  12288  1 snd
snd_page_alloc 12288  2 snd_hda_intel,snd_pcm
cpufreq_conservative12288  0 
cpufreq_userspace  12288  0 
cpufreq_stats  12288  0 
cpufreq_powersave  12288  0 
squashfs   28672  0 
loop   20480  0 
ppdev  12288  0 
parport_pc 24576  0 
parport28672  2 ppdev,parport_pc
eeepc_wmi  12288  0 
asus_wmi   16384  1 eeepc_wmi
sparse_keymap  12288  1 asus_wmi
video  16384  1 asus_wmi
backlight  12288  2 asus_wmi,video
mxm_wmi12288  0 
serio_raw  12288  0 
acpi_cpufreq   12288  0 
wmi12288  2 asus_wmi,mxm_wmi
microcode  16384  0 
pcspkr 12288  0 
xhci_hcd   69632  0 
mperf  12288  1 acpi_cpufreq

(The End)

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Re: [Alsa-user] e-mu 0204 onset/offset distortions

[Robert M. Riches Jr.]

 From: Samuele Carcagno sam.carca...@gmail.com
 To: alsa-user@lists.sourceforge.net
 Date: Sun, 16 Feb 2014 13:36:02 +

 Hi,

 I'm trying to set up the e-mu 0204 for psychoacoustics research purposes on 
 Debian Wheezy.
 When using aplay to play a short (200 ms, or 900 ms) wav file, at the onset 
 and at the offset of the sound
 there are audible pops and clicks. I'm actually using a GUI program I wrote 
 in pyqt4 that writes the sound
 to a wav file, and then calls aplay to play it. This program can also output 
 sound through PyAudio (portaudio) and
 pyalsaaudio. If I play a sound with PyAudio, then there are no pops and 
 clicks at the onset/offset of the
 sound, however, with PyAudio there are occasional distortions in the middle 
 of the sound.
 The strange thing is that if I first play some sounds with PyAudio and then 
 switch to aplay, aplay
 works well without pops and clicks at the onset/offset of the sound. I have 
 the impression that PyAudio
 is setting some soundcard parameters that make it work well with aplay. If I 
 close the pyqt4 GUI and
 open it again, the problem with aplay returns. Do you have any idea of what 
 the problem could be,
 and how it could be fixed?

 Some additional info. The problem with aplay is present also if I use it from 
 the command line rather
 than the pyqt4 app. pyalsaaudio has the same problem as aplay, again if I 
 play some sounds with PyAudio
 and then switch to aplay the problem is fixed, as long as I keep the pyqt4 
 app open. I have been using
 the same program with the e-mu 0202 and it has been working without any 
 issues, so I would discard the pyqt4
 app as the source of the program. I have tried plugging the 0204 on different 
 USB ports but the problem persists.

Coincidentally, I just finished doing some coding for PyAudio and
(py)alsaaudio on Debian 7/Wheezy, though I don't use PulseAudio.
I also found that PyAudio has pops and clicks characteristic of
Xruns (overruns and/or underruns), while AlsaAudio does not.
(FWIW, I decided to use AlsaAudio rather than PyAudio, though I
coded my program with a constant that makes it trivial to switch
between the two.)

In another reply, Bill Unruh suggested routing the output through
the line input of another card.  That is a very good idea and may
give you an idea of what's happening with the pops and clicks in
the middle of the file with PyAudio.

Another idea is to look at the original file to see if it has a
DC offset.  In my experience, you can get a click at the start or
end of a file if the first or last frame has a value very far from
zero.  If that is the cause of the beginning and ending transients,
you might want to consider editing your file to shave off a few
frames so it starts and ends very close to zero.  Another idea
would be to make a short (a few ms) fade-in or fade-out.

Another possible issue if if PyAudio vs. AlsaAudio set up the
card to different sample rates, one of them using software to
resample from the input WAV file to match what they set the card
to.

HTH

Robert

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Re: [Alsa-user] e-mu 0204 onset/offset distortions

[Robert M. Riches Jr.]

 Date: Sun, 16 Feb 2014 14:56:08 -0800 (PST)
 From: Bill Unruh un...@physics.ubc.ca
 To: Samuele Carcagno sam.carca...@gmail.com
 Cc: alsa-user@lists.sourceforge.net

 On Sun, 16 Feb 2014, Samuele Carcagno wrote:

  On Sunday 16 Feb 2014 20:47:44 you wrote:
  I have never seen that in any of the files I have played. That hints that 
  that
  is your input stream, rather than some problem with the soundcard itself,
  although I have also never used your sound card.
 
  So to be clear, you recorded onto a .wav file that sound. That .wav file 
  did
  not have those transients. You then played that file with aplay, and 
  recorded
  the output, starting the input before the file started playing and ending
  after it stopped.
 
  yes, that's correct. Because aplay does play the wav file well after it is 
  primed
  by playing some sounds with PyAudio, I wonder what PyAudio is doing and 
  whether
  I could instruct aplay to do the same through some command-line option.
 
  Just for completeness I should add that I observed this behaviour with the 
  soundcard
  plugged on different computers, as well as with the Debian testing branch 
  and the current
  Ubuntu development branch. I removed pulseaudio before running any of these 
  tests as it has other
  problems for me:
 
  http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-October/014756.html
 
  Probably I can circumvent the onset/offset transient issue with a function 
  that automatically
  plays some sounds with pulseaudio each time my pyqt4 program starts. I will 
  also have to check
  that the soundcard plays sounds with 24-bit depth.

 It definitely sounds like a problem with your soundcard. 
 What happens if you put in say 1 sec of silence before the sound starts
 playing. 
 Mind you that trailing DC really looks suspicious, as if your file has a DC
 component, rather than the soundcard, but is belied by the fact that if you
 play it later, it is fine. The DC looks like it is dying away at the
 beginning, but then it sure should not be there at the end.

I agree there's something very odd about the DC segments before
and after the tone burst.  I think I can just barely see the
10ms fade-in/out.  It's a little more apparent when zoomed in.
The DC segments look like maybe switching transients while the
card is being initialized.  Perhaps playing a short piece of
silence via PyAudio before playing your real content via AlsaAudio
would be the most practical solution for that situation.

One other thing I remembered about PyAudio with my program is I
see a lot of whining about 'Unable to find definition ...',
'No such file or directory', 'Unknown PCM rear', 'Failed to create
secure directory: Permission denied', and so forth with PyAudio
during program initialization.  With AlsaAudio, there is no such
complaining.

HTH

Robert

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Re: [Alsa-user] OSS emulation doesn't allow mixing.

[Robert M. Riches Jr.]

ChaosEsque Team,

Congratulations on being first person from a mailing list that I
have ever added to an email deny list.  If you can't accept
reasonable advice without foul-mouthed reviling and threats of
violence against a benefactor, you aren't allowed in my inbox.

Robert


 Date: Thu, 30 Jan 2014 13:53:14 -0800 (PST)
 From: ChaosEsque Team chaosesquet...@yahoo.com
 To: Bill Unruh un...@physics.ubc.ca
 Cc: alsa-user@lists.sourceforge.net

 When a program uses OSS emulation (what? writing to /dev/dsp, the file, the 
 unix way?) what I, and every other person wants is for sound to simply be 
 sent to the main Alsa sound card. That is all. Without blocking. With mixing. 
 We don't want to deal with the failings of the old opensource OSS drivers, we 
 want our sound from the old apps to go to Alsa. And you ... know this.

 Oss never allowed mixing. This is an
  emulation of oss. It does not allow
  mixing. If you want mixing uses alsa with a frontend. Or get
  the newer oss
  implimentations.
 
  Alsa is now the standard Oss emulation is NOT a separate
  sound system, it is
  an emulation of an old system under alsa. emulation,
  including all its
  shortcomings.

 If you spoke this way to me in real life, God willing: I would beat you 
 untill you were on the ground, then I would (God willing) continue beating 
 you while kneeling over you untill you had to spend the next four years in 
 the hospital, and then the rest of your life dealing with your injuries. I 
 kinda hope we meet. I definitly hope that bad things happen to you and you are
 physically disabled you piece of shit. Go away and use something else if you 
 want old programs to work. 

 Code the feature. Linus should have never allowed ALSA in without it.
 If I were Linus I would take away maintainership from you guys and give it to 
 someone who is willing to allow OSS emulation to mix.

 Too bad this doesn't work, like I'd want it to work.
 options snd-pcm-oss nonblock_open=1

 Daniel Mack zon...@gmail.com
 Go troll some other list please.

 Asking for sound to mix when old apps send sound data to the system is 
 trolling? GO AND ... YOURSELF.
 I hope bad things happen to you people. You, like everyone else now, try to 
 shut down and ignore all other views by claiming trooolling.
 ... YOU.

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Re: [Alsa-user] Akai EIE

[Robert M. Riches Jr.]

 Date: Thu, 9 Jan 2014 08:09:17 -0500
 From: Nathan Jackson nate.ds.jack...@gmail.com
 To: Clemens Ladisch cladi...@googlemail.com
 Cc: alsa-user@lists.sourceforge.net

  Clemens Ladisch wrote:
  This device cannot change any of its sample format parameters.  So
  either it sends wrong data, or some data gets dropped by the computer.
  The Raspberry Pi has a horrible USB implementation, and not very much
  bandwidth, so I'd guess it's the latter.

 See the problem is that it works with an older version of Raspbian, so
 I know it did work at one time.

  Please check if recording with this device works on a 'real' PC.

 It works on both the Pi in an old version of the software and also my
 PC just fine.

 I'd like to debug to try and fix the problem, but I'm not sure where
 to start on this.  Whether the problem is a bug or a configuration
 issue on my end.

 -Nathan

fwiw: The Pi's USB implementation may well be horrible relative
to more costly machines, but it should have enough bandwidth to
handle most audio jobs.  As I type this, I'm watching HD video
that was captured from a USB DTV tuner via a Raspberry Pi's USB
port.  (During video recording, the stream was sent back out the
network port while it was coming in via USB.)  If the Raspberry
Pi can handle HD video, it should be able to handle the vast
majority of audio tasks.

HTH

Robert Riches
rm.ric...@jacob21819.net

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Re: [Alsa-user] M-Audio keystudio usb midi keyboard works on Ubuntu Studio 12.04, not on 13.10, now what ?

[Robert M. Riches Jr.]

 Date: Sun, 3 Nov 2013 20:47:23 +0200
 From: Jyrki Saarela jyrki.saar...@gmail.com
 To: Daniel Mack zon...@gmail.com
 Cc: alsa-user@lists.sourceforge.net

 On Sun, Nov 3, 2013 at 12:31 PM, Daniel Mack zon...@gmail.com wrote:
  Ok, then we have a good test setup to investigate on.

 Seems so, except  (see below)

  Please provide the output of 'lsusb -v' as well as a full dmesg output
  of that machine then, with the keyboard connected of course, for both
  Ubuntu versions.

 on 12.04 keyboard was listed, as expected

  Also, what kernel does 12.04 (and 13.10, respectively) ship with again?
  Check with 'uname -a'.

 and after reboot to 13.10, it is listed there also. Nothing was
 changed, now it just works. I really don't know what to say...

 Well, now there is another problem :) Neither qjackctl or Patchage can
 connect the keyboard anywhere. Not to midi trough nor to qsynth.

 dmesg outputs not attached, seems that message gets too large...

 - Jyrki

Very long shot: Does it still work under 13.10 if everything is
powered off and then booted directly to 13.10?  This is based on
a wild guess that there might be firmware that gets downloaded to
the device that enables 13.10 to see it.  (I have a USB-attached
TV tuner that enumerates as one USB device, installs firmware,
disconnects as a USB device, then re-enumerates as a different
USB device once it has its firmware.)

Another long shot: Might it be flaky physical connectors?

HTH

Robert Riches (just another lurker)

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Re: [Alsa-user] Using compressor + limiter with 6 channel sounds

[Robert M. Riches Jr.]

I probably don't know any answers, but would like to make sure I
at least understand the question.  Are you trying do AGC on a
pair-wise basis?  Or, is it something else you're trying to do?

Thanks,

Robert Riches


 Date: Sun, 13 Oct 2013 15:02:42 +0200
 From: Paolo Bolzoni paolo.bolzoni.br...@gmail.com
 To: Uwe upu...@googlemail.com
 Cc: alsa-user@lists.sourceforge.net

 I guess I can split the six channels in three limiters using the multi
 plugin, but how I can join them back?
 In the ladspa are other limiters, but they do not have the gain
 control so they do not actually increase the
 volume. Just adding plain 20db of amplification distorts the sound...
 Maybe there is an limited amplifier
 or something?

 I am trying to set-up a chain of plugins to always get the max volume...


 On Sun, Oct 13, 2013 at 12:58 AM, Uwe upu...@googlemail.com wrote:
  the problem with the configuration, I think, is: the plugins can handle a
  maximum of 2 channels.
 
  unfortunately, I cannot tell you exactly how, but it *should* be possible to
  route 3 pairs of channels to separate instances of the plugins.
 
  such a setup might also make sense musically. it might be a good idea to
  setup separate plugins for center and bass because the signal in the
  channels differs significantly from the left/right channels in frequency
  range and level.
 
  have fun, Uwe
 
 
  2013/10/12 Paolo Bolzoni paolo.bolzoni.br...@gmail.com
 
  Dear list,
  This is my .asoundrc, and the pair compressor + limiter works fine for
  stereo input:
 
  --- 8
  pcm.ladcomp_compressor {
type ladspa
slave.pcm ladcomp_limiter;
path /usr/lib/ladspa;
plugins [{
label dysonCompress
input {
#peak limit, release time, fast ratio, ratio
controls [0 1 0.5 0.99]  }
}]
  }
 
  pcm.ladcomp_limiter {
type ladspa
slave.pcm plughw:Audigy2;
path /usr/lib/ladspa;
plugins [{
label fastLookaheadLimiter
input {
 #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
  time (s) 0.01 - 2
 controls [ 20 -1 0.8  ]  }
}]
  }
  8 ---
 
  Unfortunately, it does not work for 6 channels.
 
  I.e., this one works fine (I setup my system to use a certain PCM
  via environment variable):
  $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 2 -t wav -l 5
 
  This one does not and you hear only the left and right channels.
  $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 6 -t wav -l 5
 
  Of course the hardware is wired correctly and this one works as expected:
  $ ALSAPCM='pcm.surround51' speaker-test -c 6 -t wav -l 5
 
 
  Since normally you have to select surround51 manually I thought that the
  problem could be in the output pcm, and I rewrote like this. But it is the
  same:
 
  --- 8
  #[...] analogous 51 compressor omitted
 
  pcm.plug51 {
type plug
slave.pcm surround51
slave.channels 6
  }
 
  pcm.ladcomp_limiter51 {
type ladspa
slave.pcm plug51
path /usr/lib/ladspa
plugins [{
label fastLookaheadLimiter
input {
 #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
  time (s) 0.01 - 2
 controls [ 20 -1 0.8  ]  }
}]
  }
  8 ---
 
  Is there a way? Any insight?
 
  Yours faithfully,
  Paolo
 
 
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Re: [Alsa-user] seeking low latency usb or PCIE sound card/interface

[Robert M. Riches Jr.]

 Date: Fri, 20 Sep 2013 09:19:48 +0200
 From: Clemens Ladisch cladi...@googlemail.com
 To: Robert M. Riches Jr. rm.ric...@jacob21819.net, 
  alsa-user@lists.sourceforge.net

 Robert M. Riches Jr. wrote:
  I'm seeking suggestions for a low latency usb (or PCIE) sound
  interface (or card).  Or, maybe some good news that USB sound
  devices are generally reasonably low in latency.

 All USB sound devices use the same driver and have almost the
 same latency.

Ahhh, thank you for that info.  I think I figured out two ways to
measure the latency of a round trip through two devices and
associated software:

  Manual observation:

1) Attach two USB sound adapters.
2) Attach microphone to card A's input.
3) Using 'unbuffer', run arecord on card A directed to aplay
   on card B.
4) Attach headphones to card B.
5) Listen for gross latency.

  More precise measurements:

1) Attach two USB sound adapters.
2) Connect source of analog sound to card A's L input.
3) Connect card B's L output to card A's R input.
4) Using 'unbuffer', run arecord on card A, tee to a file,
   into aplay to card B.
5) View the saved samples in an oscilloscope program.

Now to find a device at a decent price that has stereo line input
rather than just mono microphone input...

  The above setup initially used Mageia 1 on the main machine, and
  life was good.  Sound was essentially perfect.  However, when I
  switched the main machine to Mageia 2, kernel
  3.4.52-server-1.mga2, sound became hideously chopped.  Somewhere
  around 25% or more of the sound samples are being lost due to
  xruns.  Research into the problem showed snd_aloop was _NOT_
  keeping time but would accept samples as rapidly as the playback
  client would send them.

 This sounds like a bug in snd-aloop.  Please try updating to
 a modern kernel.

That would be the ideal.  I'd love to have the free time it would
require to compile the latest kernel and then test it to make
sure the distribution's userspace doesn't depend on something
unique to their older patched kernel.

Thanks again.

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[Alsa-user] seeking low latency usb or PCIE sound card/interface

[Robert M. Riches Jr.]

I'm seeking suggestions for a low latency usb (or PCIE) sound
interface (or card).  Or, maybe some good news that USB sound
devices are generally reasonably low in latency.

My target latency would be around 25-50msec per component.  This
is a thin-client setup, so playback sound would go through three
components.  Two-channel playback is the only essential
requirement.

Anyway, what I'm thinking is to attach a real PCIE or USB sound
card or interface to the main machine (well, two of them); point
the .asoundrc files at one of them, feed analog signal into
another USB device's input, into a Raspberry PI, over ethernet
via NetJACK or maybe just netcat, and to the thin-client machine.
That way, the real hardware sound card would throttle the samples
at the front end of the chain, which should be massively better
than the current situation.

The detail, lest I not provide enough background:

I have one main machine (because I insist on ECC RAM) and a
couple of thin-client machines.  The main machine currently runs
Mageia 2 on a Xeon W3680 (so CPU horsepower is ample).  I have
several loopback sound cards (one per user) using snd_aloop.  The
users' .asoundrc files play into a loopback/snd_aloop card,
NetJACK sends sound from there to the thin-client, and it goes
out over the thin-client machine's hardware (HDMI and green
jack).  The thin-client runs a remastered TinyCore Linux, on an
i3 540 CPU, booting via PXE.  There's a private gigabit LAN
between the main machine and the thin-client machines.  Only one
thin-client is generally powered on at a time.  All machines run
straight ALSA (and NetJACK); Pulse is _NOT_ being used.  (I'd
uninstall pulse, but I don't like RPM complaining about massive
numbers of missing dependencies.)  Neither CPU nor network are
significant limiting factors.

The above setup initially used Mageia 1 on the main machine, and
life was good.  Sound was essentially perfect.  However, when I
switched the main machine to Mageia 2, kernel
3.4.52-server-1.mga2, sound became hideously chopped.  Somewhere
around 25% or more of the sound samples are being lost due to
xruns.  Research into the problem showed snd_aloop was _NOT_
keeping time but would accept samples as rapidly as the playback
client would send them.  That resulted in massive xruns.  The
problem can be demonstrated with nothing more than aplay,
snd_aloop, and arecord in Mageia 2 on the main machine.  I
believe it's a bug in snd_aloop, but I don't have the expertise
to write a convincing bug report for something in kernelspace.

Anyway, are there USB sound interfaces with latencies no more
than 25-50msec?  (Or, are they around 500msec latencies like I
hear at work on Doze-based laptops over Lync?)

Thanks,

Robert Riches

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Re: [Alsa-user] recording audio from speakers

[Robert M. Riches Jr.]

 From:  ?? akhil...@gmail.com
 Date: Mon, 29 Jul 2013 13:50:01 +0700
 To: Ralf Mardorf ralf.mard...@alice-dsl.net
 X-Headers-End: 1V3hHm-0005Gs-GX
 Cc: 72-alsa alsa-user@lists.sourceforge.net

 you may also try alsa's loopback module
 https://bbs.archlinux.org/viewtopic.php?pid=765075#p765075

At least with some distribution releases, the ALSA loopback
module doesn't work well.  For example, with Mageia 1 it worked
perfectly.  Then, with Mageia 2, it has severe xruns, both with
net-JACK and with only aplay and arecord.  Evidence suggests the
loopback module accepts samples too rapidly from the sending
client, which overruns the buffer of the receiving client.

I asked here a while ago if there were solutions.  Yes, I ought
to file a bug report if I could scrape together a few minutes to
do so.

Robert

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Re: [Alsa-user] help with first ALSA program, please?

[Robert M. Riches Jr.]

 Date: Sat, 18 May 2013 12:36:57 -0500
 From: Perry Kivolowitz pe...@kivolowitz.com
 To: alsa-user@lists.sourceforge.net

 Hi All,

 I have written an ALSA output program which is producing garbled results.

 The results are the same when I wrote an equivalent program using 
 RtAudio / ALSA.

 The platform is an RK3066 based stick computer running Picuntu.

 The sound file is processed by sndfile. Playing a sin wav at 375 Hz also 
 produces unexpected results.

 Playing the sound file with aplay works fine.

 ...

On the off chance you haven't already tried this, if it were
my program I would instrument the program to print out buffer
pointers for each packet of samples sent to Alsa (or wherever
you're sending the data).  If it were my program, it would
likely turn out that I had made some errors in pointer
arithmetic or something like that.

HTH

Robert

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Re: [Alsa-user] install problem

[Robert M. Riches Jr.]

 Date: Sat, 4 May 2013 06:14:12 -0700 (PDT)
 From: Bill Unruh un...@physics.ubc.ca
 To: Joe Armstrong joea...@gmail.com
 Cc: alsa-user@lists.sourceforge.net

 On Sat, 4 May 2013, Joe Armstrong wrote:

  On Fri, May 3, 2013 at 9:24 PM, Bill Unruh un...@physics.ubc.ca wrote:
  On Fri, 3 May 2013, Joe Armstrong wrote:
 
  On Fri, May 3, 2013 at 7:39 PM, Bill Unruh un...@physics.ubc.ca wrote:
 
  On Fri, 3 May 2013, Joe Armstrong wrote:
 
 
  Try running aplay, etc as root, and see if it works, to see if it is some
  permission problem. If that does not work either,then the problem lies
  elsewhere.
 
 
 
  Well running aplay as non root works (ie does not crash) (but no sound
  and really strange behaviour as I explained earlier)
 
  As root I see this:
 
  $ sudo aplay -vv sound.wav
 
 
  Try logging on as root
  su -
 
  rather than doing sudo.
 
  Same thing happens
 
  root@nuc:~# aplay -vv  sound.wav
  Home directory not accessible: Permission denied
  ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
  aplay: main:682: audio open error: No such file or directory
 
  I suppose the next thing to do is recompile asla from the sources ...

 Or look at the source of aplay and put in some debugging commands so that you
 know which file it is referring to when it says no such file or directory or
 which Home directory is not accessible. 
 Is your home directory nfs mounted or something? 
 The audio open error seems to be from snd_pcm_open 
 Also try 
 -Dhw:0 
 or 
 -Dplughw:0
 as an argument for aplay so it does not just use the default.
 (assuming that the card you want to use is card 0)

You could put strace -o somefile  at the front of the command
to see what system call is triggering the error message.  If the
correlation isn't clear, omit the -o and its argument and tee
both stdout and stderr from the aplay command to a file.  Or,
run the strace aplay ... command from inside 'script'.

HTH

Robert Riches

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Re: [Alsa-user] Slightly OT: how to pipe output to look like a soundcard

[Robert M. Riches Jr.]

(Oops.  Forgot to reply to the list as well...)

I'm definitely not an ALSA expert, but I believe there's supposed
to be a way to do what you're asking.  There's a kernel module
called snd_aloop that creates one or more artificial sound cards.
You can play sound samples to it from one client and capture
sound samples from it with another client.  You can put options
for the loopback sound cards in /etc/modprobe.conf or similar and
then make sure to load the module when you boot (or later).

In this case, you could pipe the output of your SDR program into
aplay with appropriate options to set type (probably -t raw),
set format, and direct the samples to the loopback card with
something similar to -D hw:2,0,0.  Then, connect one (or IIUC,
multiple) capture clients to something similar to -D hw:2,1,0.

Unfortunately, there appears to be a problem with at least some
recent kernels.  On about February 23, I posted a problem and
question about behavior I notice on Mageia 2 where the loopback
sound card allows its input (playback) stream to run
significantly faster than the specified/accepted rate, and the
output (capture) stream gets overruns.  I have yet to hear an
answer or other reply to that problem/question.  I'm about to
file a bug report about the problem.

Fortunately for you, in your case, if your SDR sends at exactly
the right rate, you might be okay once the buffers all find their
pace and stride.

HTH

Robert


 To: alsa-user@lists.sourceforge.net
 From: li...@lazygranch.com
 Date: Sat, 2 Mar 2013 01:52:50 +

 Not exactly ALSA related, but I have software defined radio
 program that feeds an audio stream to stdout. Can I make it
 look like a /dev/dsp device so other programs can use it.  That
 is programs designed to read from soundcards?

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Re: [Alsa-user] 2 sound outputs?

[Robert M. Riches Jr.]

(Ooops.  Forgot to reply to the list as well...)

Hi Doug,

I'm definitely not an ALSA expert, but I am aware there are ways
that should do what you want.  The following answer assumes you
are _NOT_ running PulseAudio.  If you are running PulseAudio,
you'll probably need answers from someone who knows PulseAudio.

If you do aplay -l, you will see a listing of your sound units
by device names.  If you do aplay -L, you will see a listing of
your sound units by PCM names.  Make a note of both name forms
for each of your units.

Most ALSA clients have a -d or -D option that takes one of the
name forms.  Apparently, some require one name form and some
require the other.  Try both forms and use whichever works.
Once you find which name form works with the client in question,
simply use the name of the sound unit to direct sound there.

If you have a client with absolutely no way to specify a sound
unit to use, you could use separate user accounts, each with a
.soundrc file that directs ALSA to use a given device as the
default audio device.  There is documentation on the details to
do that, probably better than I could remember to tell you.

HTH

Robert


 Date: Fri, 01 Mar 2013 18:13:38 -0500
 From: Doug dmcgarr...@optonline.net
 To: alsa-user@lists.sourceforge.net

 I have N10/ICH 7 Audio Controller from Intel on the mobo, and I have 
 GF116 Audio Controller as part of a GeForce video card from NVIDIA, 
 which feeds
 an HDMI port.  I would like to be able to get sound out of both units at 
 the same time, so as to be able to work with the sound at the computer, and
 also see Internet movies with sound on the TV set, which is in another 
 room. Using Alsa on PCLinuxOS32KDE-12.2 on a Foxconn G41MXE mobo.
 Some time ago, someone on the pclos forum tried to help me get both 
 sources working, without success.  Is this possible, and if so, how?

 --doug

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[Alsa-user] loopback device (snd_aloop) appears running too fast, causing overruns

[Robert M. Riches Jr.]

(first-time poster to list)

My snd_aloop loopback device appears to be running too fast
and is causing overruns.  OS is Mageia 2.  Kernel is
3.4.32-server-2.mga2.  Hardware is Asus P6T6 WS Revolution
in case that matters.

First symptom: Was using snd_aloop with JACK for a zero-
client setup with Mageia 1 on the machine running snd_aloop,
and sound quality was good.  Switching to Mageia 2, sound
quality is terrible.  It sounds like there's a short silent
period several times a second--badly chopped-sounding.
Tried all the JACK tuning I could find to no avail.  This
might not be directly caused by the loopback device.

Second symptom: I wrote two programs.  The first captures
from the loopback device and sends the data to a network
port.  The second receives the data from a network port and
plays to a real soundcard.  With settings for CD rate,
44100 frames per second, 176000 bytes per second, the sender
would get overruns every couple of seconds starting about
15 seconds after starting.  Instrumenting the sender for
data capture rate showed about 55k frames per second.

Third symptom: Swithed to standard components: aplay, arecord,
pv, and nc in this manner:

   sender:arecord -D hw:2,1,0 -t raw -f cd - | pv | nc localhost 51230
   receiver:  nc -l 51230 | pv | aplay -t raw -f cd

This shows the same pattern of overruns in the sender that
my program saw.  The receiver shows about 172kB/s, but the
sender shows 215kB/s until the overruns start being reported.

Adding a '-a' option to the pv commands, and doing pv -a
on the stream that feeds aplay to hw:2,0,0 of a CD-rate
track and letting everything run for a while, the aplay
into hw:2,0,0 shows 217kB/s, the sender shows 181kB/s, and
the receiver shows 172kB/s.

Shouldn't the loopback device control the frame rate to
be what the applications opened it for?

Is there a solution to keep the above sender and receiver
command chain from having overruns?

Thanks,

Robert

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